- sr-users - kamailio.org

[Serusers] recording REGISTER messages and errors
by Iqbal 27 Aug '05

27 Aug '05

Hi Is there a way, to do this, in acc table Iqbal

2 1

[Serusers] Radiusclient problem
by Ashutosh Kumar 27 Aug '05

27 Aug '05

Hi, I have installed radiusclient-ng-5.1 (from tarball) with ser-0.9.0. But still SER shuts down saying "ERROR: load_module: could not open module </usr/local/lib/ser/modules/auth_radius.so>: libradiusclient.so.2: cannot open shared object file: No such file or directory". Why is SER still trying to find radiusclient library instead of the radiusclient-ng library which is already installed. Regards, Ashutosh Kumar

2 1

[Serusers] Save call details to mySQL
by Andrew Madrigallos 27 Aug '05

27 Aug '05

Hi Everyone, I'm new in using SER, I was using different server before, my question is how can I save call details on mySQL for future accounting? I browse the archive and everybody answer same thing which is not for newbie please post sample configuration here. Thanks, Andrew _____________________________________________________ Powered by Crown Multimedia (http://www.crownmultimedia.net)

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Re: [Serusers] Checking URI
by Ryan Pagquil 26 Aug '05

26 Aug '05

Yes, it really make sense. I'll find the way how to check those From and To header of a SIP request. Maybe I will use textops.so to do this or any other module that can handle these kind of application. In your experience what did you use to make work this kind of scenario? --Ryan Steve Blair wrote: > > Sorry I missed the route[3] stuff because my mail client wrapped > the message text. Even though I got the route block number wrong I > think my original comment is still valid. If these inter-domain calls hit > route[3] then there you need to either check if the "from" user is > permitted to send calls or you need to check if the "to" address is a > valid > local address. Which approach you take depends on your implementation. > > If your iptel.org users are on soft clients where they can easily > enter the > authentication credentials then I think that would be a good approach. > If you > are trying handle multiple doamins (such as other business partners) > then the > problem is a little more difficult. We simply trust certain remote IP > addresses > to be the address of a "friendly" peer. We cannot prompt them for > credentials > because their users don't have accounts on our system. > > Does this make sense? > > _Steve > > Ryan Pagquil wrote: > >> Hi Steve, >> Here is the route[1] configuration. The calls from outside >> domains are falling to route[3] which make the call to be blocked. >> >> ##Default message handler## >> route[1] { >> >> t_on_reply("1"); >> >> if (!t_relay()) { >> >> if (method=="INVITE" || method=="ACK") { >> end_media_session(); >> }; >> >> sl_reply_error(); >> }; >> } >> >> Thanks, >> Ryan >> >> Steve Blair wrote: >> >>> >>> It looks like the type of call you describe should fall through to >>> route[1] is that >>> the behavior you are seeing? Anyway in route[1] (or somewhere ) you >>> need to >>> accept that the from address might not be a local user and allow for >>> the >>> case where the from address is a "trusted alternate domain like >>> iptel.org". I'm >>> guessing, but I cannot confirm from this code, that route[1] is >>> checking the >>> from user credentials and providing you with the authentication >>> required >>> message. >>> >>> At that point you could elect, and I think a soft phone will, prompt >>> the user for these >>> credentials. >>> >>> -Steve >>> >>> Ryan Pagquil wrote: >>> >>>> Hi Steve, >>>> >>>> Here is what happens with my current configuration. When I >>>> call from my iptel.org account going to my local users it prompts >>>> me "407, Proxy Authenctication Required". So what I like to do is >>>> to allow those calls going to my local users from other domain. >>>> BTW, here is my ser.cfg's INVITE message handler. >>>> >>>> ##INVITE message Handler## >>>> route[3] { if (client_nat_test("3")) { >>>> setflag(7); >>>> force_rport(); >>>> fix_nated_contact(); >>>> }; # if (!proxy_authorize("", "subscriber")) { >>>> # proxy_challenge("", "0"); >>>> # break; >>>> # } else >>>> >>>> if (!check_from() && method=="INVITE") >>>> { sl_send_reply("403", "User From=ID"); >>>> break; }; >>>> >>>> lookup("aliases"); >>>> if (uri!=myself) { >>>> route(1); >>>> break; >>>> }; >>>> >>>> if (!lookup("location")) { >>>> sl_send_reply("404", "User not found"); >>>> break; >>>> }; >>>> >>>> if (method=="CANCEL") { route(1); >>>> break; >>>> }; >>>> >>>> >>>> consume_credentials(); >>>> >>>> if (isflagset(6) || isflagset(7)) { >>>> use_media_proxy(); >>>> }; >>>> >>>> route(1); >>>> } >>>> >>>> I misse call with your SIP phone, just testing it. =) >>>> >>>> Thanks, >>>> Ryan >>>> >>>> Steve Blair wrote: >>>> >>>>> >>>>> Set the alias=<your sip domain> parameter then use if >>>>> (uri==myself) construct. >>>>> >>>>> -Steve >>>>> >>>>> Ryan Pagquil wrote: >>>>> >>>>>> Hello, >>>>>> How can I check if the call is for my domain? What >>>>>> specific module should I use? >>>>>> >>>>>> Thanks, >>>>>> >> > -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Users] clienst behind same NAT
by jayesh nambiar 26 Aug '05

26 Aug '05

hi all, Thanx Klaus. Sorry to ask a question again related to the same problem. I had already tried writing the avp_write after lookup but even that causes a problem. When avp_write is done it writes the private IP address of the callee while the source_ip is the IP address of the NAT. I beleive this IP address is written because of the fix_nated_contact. Again, when avp_check is done, it returns false and again the RTP session gets started. IS there any way out to this problem. Thanx again.. Jayesh --------------------------------- Start your day with Yahoo! - make it your home page

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[Serusers] exec_dset & rtp stream
by Mik Cheez 26 Aug '05

26 Aug '05

Does anybody have a sample of a script that uses exec_dset to route to a PSTN, as well as handle the RTP stream for NAT'ed and non NAT'ed clients? Everything I've done either ends with a 483 or a large amount of the calls get a 481 "transation does not exist". the exec_dset could return many different values, so I cannot just use 'rewritehost'. :)

1 0

[Serusers] DB Mode in Permissions module
by Ricardo Martinez 26 Aug '05

26 Aug '05

Hello List. I'm using the permission modules (in particular de allow_trusted command) to authorize the trusted hosts in my ser box. I'm also using the db_mode "1" (cache mode), to avoid that every INVITE became a request to my local database. Once in a while i need to add new "trsuted hosts" to my platform, but the changes in the database are not reflected in the allow_trusted query (since is cached). My question is : Is there a way to force the SER to read the changes in the DB without restarting SER.? or i have to use db_mode 0 to acomplish this?. I really appreciate any help here. Thanks in advance. Regards, Ricardo Martinez.-

2 1

[Users] help with uac_redirect
by Kevin M. Murphy 26 Aug '05

26 Aug '05

Greeting all, I am currently working at migrating a Vovida deployment to OpenSer, and need to keep some of the Vovida architecture. I have been messing with the uac_redirect module and have found it to be almost exactly what i am looking for with one exception. I would like to fork calls serially based on contacts in a 302, however i have only been able to do this in parallel. Is there a way to make this happen. Here is some example config stuff, Thanks in advance to all. ---------------------------------------------------------------------- #test redirect config (to service two vovida Proxyies) : # the idea here is that the two ip addresses will query the redirect server first and then route apropriately # the first host here is invalid, while the second should work, if (method=="INVITE") { if(src_ip=="xxx.xxx.xxx.15") { rewritehostport("xxx.xxx.xxx.20:5066"); #prefix("23801"); append_branch("sip:4439121133@xxx.xxx.xxx.20:5060"); sl_send_reply("302", "Moved Temporarily"); } if(src_ip=="xxx.xxx.xxx..20") { rewritehostport("xxx.xxx.xxx.15:5066"); prefix("23801"); append_branch("sip:238014105716405@xxx.xxx.xxx.20:5060"); sl_send_reply("302", "Moved Temporarily"); } #currently the Proxies have a failed route statment like this: failure_route[1] { get_redirects("3:2"); append_branch(); t_relay(); } ---------------------------------------------------------- The behavior I would like to see is that the first contact is tried, and, after failure, the second contact is tried. Thanks again, Kevin

2 1

[Serusers] how do I route to ranges?
by Dana Olson 26 Aug '05

26 Aug '05

I am not sure how to approach this, or even if it's possible. I want to have this kind of thing (pseudocode): if sipuri >0 & sipuri <96 { do stuff } if sipuri >95 & sipuri < 120 { do stuff } And so on. Assuming that the sipuri is a number, and anything outside of the ranges, or letters is ignored (and handled elsewhere). I could build individual routes for each number, but I am hoping there is an easier way to do it. If anyone could help me out, that'd be really awesome. -- Dana

2 2

[Serusers] FW: Radius packets not being sent to radius server
by Ashutosh Kumar 26 Aug '05

26 Aug '05

Hi, Finally I got the radius working. But now I am getting the error message rc_avpair_gen: received unknown VSA attribute 103, vendor 9 of length 18: 0x683332332D72657475726E2D636F64653D31 I am using cardsaver radius (quantum). Can someone explain the error message. Regards, Ashutosh Kumar ________________________________________ From: Ashutosh Kumar [mailto:ashutoshk@chetu.com] Sent: Tuesday, August 23, 2005 12:13 AM To: 'serusers(a)lists.iptel.org' Subject: Radius packets not being sent to radius server Hi, I m using radius with SER-0.9.0 and also using all the reqd functions viz. www_raidus_authorize etc.. , but when I ngrep for any connection with my radius server, I dont get any radius-packets in the result. Also the log file contains rc_send_server: no reply from RADIUS server a.b.c.d.:1812 (a.b.c.d is the radius servers IP). This is despite the fact that the radius server is running and hosting another running service, also radisuclient is configured fine. Kindly guide. 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

[Users] authentication in multi-domain
by jayesh nambiar 26 Aug '05

26 Aug '05

hi all, Thanx Klaus and Bogdan, for rectifying my idea about using Openser in multi-domain environment. It did help me a lot. But sorry to ask one more question regarding the same issue. How do we authenticate the clients in multi-domain environment. I beleive openser allows to add users with same username in different domain!! If so then is it required that, when a call is made to such user the caller needs to specify the complete username with domain.. I did not actually check how it works but i thought just to clear the doubt. And how do we authenticate the users registered in different domains. pls help me understand this.. Thanx a lot for previous responses. Jayesh ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

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[Serusers] Total NewBie to Linux
by Herman Janse van Vuuren 26 Aug '05

26 Aug '05

Hi There, I have looked at the manuals and stuff, got everything installed, but how to I get it to work? where to I go and enter my IP of my billing server? How do I configure calling cards on this server? How do I register a user with a permanent account. Best Regards, Herman Janse van Vuuren Q-Kon (PTY) LTD Tel:+27 12 665 0052 ext 216 Mobile: +27 82 920 6959 Email: hjvvuuren(a)qkon.com

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Re: [Serusers] Checking URI
by Ryan Pagquil 26 Aug '05

26 Aug '05

Hi Steve, Here is the route[1] configuration. The calls from outside domains are falling to route[3] which make the call to be blocked. ##Default message handler## route[1] { t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } Thanks, Ryan Steve Blair wrote: > > It looks like the type of call you describe should fall through to > route[1] is that > the behavior you are seeing? Anyway in route[1] (or somewhere ) you > need to > accept that the from address might not be a local user and allow for the > case where the from address is a "trusted alternate domain like > iptel.org". I'm > guessing, but I cannot confirm from this code, that route[1] is > checking the > from user credentials and providing you with the authentication required > message. > > At that point you could elect, and I think a soft phone will, prompt > the user for these > credentials. > > -Steve > > Ryan Pagquil wrote: > >> Hi Steve, >> >> Here is what happens with my current configuration. When I call >> from my iptel.org account going to my local users it prompts me "407, >> Proxy Authenctication Required". So what I like to do is to allow >> those calls going to my local users from other domain. BTW, here is >> my ser.cfg's INVITE message handler. >> >> ##INVITE message Handler## >> route[3] { if (client_nat_test("3")) { >> setflag(7); >> force_rport(); >> fix_nated_contact(); >> }; >> # if (!proxy_authorize("", "subscriber")) { >> # proxy_challenge("", "0"); >> # break; >> # } else >> >> if (!check_from() && method=="INVITE") { >> sl_send_reply("403", "User From=ID"); >> break; }; >> >> lookup("aliases"); >> if (uri!=myself) { >> route(1); >> break; >> }; >> >> if (!lookup("location")) { >> sl_send_reply("404", "User not found"); >> break; >> }; >> >> if (method=="CANCEL") { route(1); >> break; >> }; >> >> >> consume_credentials(); >> >> if (isflagset(6) || isflagset(7)) { >> use_media_proxy(); >> }; >> >> route(1); >> } >> >> I misse call with your SIP phone, just testing it. =) >> >> Thanks, >> Ryan >> >> Steve Blair wrote: >> >>> >>> Set the alias=<your sip domain> parameter then use if (uri==myself) >>> construct. >>> >>> -Steve >>> >>> Ryan Pagquil wrote: >>> >>>> Hello, >>>> How can I check if the call is for my domain? What >>>> specific module should I use? >>>> >>>> Thanks, >>>> -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Serusers] Check if OUI is registered
by Dmitry Isakbayev 26 Aug '05

26 Aug '05

Hi, When I process an Invite, is there a way to check if the originating UI is registered with the proxy? Thanks, Dmitry

2 1

[Serusers] dbtext module
by Carlos 26 Aug '05

26 Aug '05

Hi, I'm trying to use dbtetx module, i've written the ser.cfg with the necesaries changes, also i do a directory /tmp/serdb with those files(location, aliases, version & subscriber), but when i run my ser i have this problem: /* 0(0) fixing /usr/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/lib/ser/modules/rr.so record_route 0(0) fixing /usr/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/lib/ser/modules/registrar.so save 0(0) DBT:dbt_load_file: request for table [location] 0(0) DBT:dbt_load_file: db is [/tmp/serdb] 0(0) DBT:dbt_load_file: loading file [/tmp/serdb/location] 0(0) DBT: column[0] is STR! 0(0) DBT: column[1] is STR! 0(0) DBT: column[2] is STR! 0(0) DTB:dbt_db_get_table: --------------- 0(0) DBT:db_query: table not loaded! 0(0) DBT:db_query: error while quering table! 0(0) preload_udomain(): Error while doing db_query 0(0) register_udomain(): Error while preloading domain 'location' 0(0) domain_fixup(): Error while registering domain ERROR: error ffffffff while trying to fix configuration 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: tm_shutdown : empting DELETE list 0(0) DEBUG: tm_shutdown : empting hash table 0(0) DEBUG: tm_shutdown: releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : done 0(0) DBT:destroy ... 0(1410) shm_mem_destroy 0(1410) destroying the shared memory lock */ I don`t know if i have to create any table, or something like that, i supose location file is something like this: /* username(str) domain(str,null) password(str) garcia:iptel.org:1234 */ Thanks, Carlos

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RE: [Serusers] Routing multiple numbers to 1 account
by Igor Balk 26 Aug '05

26 Aug '05

The idea is not to query any tables as they are at different server but use radius to provision aliases and then use lookup("aliases") I believe there was a patch a way back for 8.12 but seems it did not found it way to CVS -----Original Message----- From: Chris St Denis [mailto:chris@aebc.com] Sent: Wednesday, August 24, 2005 3:44 PM To: Igor Balk; serusers(a)lists.iptel.org Subject: RE: [Serusers] Routing multiple numbers to 1 account This is basically the same as aliases. Query the registration table. If its not found there push the AVP alias into the ruri/username and lookup based on that. All aliases are is a static registration table anyway I believe. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Igor Balk Sent: Wednesday, August 24, 2005 11:59 AM To: serusers(a)lists.iptel.org Subject: RE: [Serusers] Routing multiple numbers to 1 account Guys, do any of you know if there is a way in SER to dynamically set up aliases using radius? I'm trying to achieve similar results but with aliases. Thank you. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Chris St Denis Sent: Wednesday, August 24, 2005 1:26 PM To: 'Erik Versaevel'; 'SER Users' Subject: RE: [Serusers] Routing multiple numbers to 1 account I am doing this by pulling the translation out of a database using radius AVP (could be done direct to database too) and sticking it as the destination. It looks like this # Try to lookup the number directly registered. if(!lookup("ser_location")) { #Number isn't found, so try to do an avp translation on it avp_pushto("$ruri/username", "s:callee_vdid"); if(!lookup("ser_location")) { #Look up the new value #Still can't find the user. Give up. sl_send_reply("404", "I can't find 'em"); break; } } -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Erik Versaevel Sent: Wednesday, August 24, 2005 6:49 AM To: SER Users Subject: [Serusers] Routing multiple numbers to 1 account Hello all, At this moment i'm using the full e164 numbers as account's in ser, that is for every number someone wants delivered to them they would have to register it using the account details. This however poses a problem with a 100 bock of numbers for example while a device can only register a few of them so what i would like to do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way the devices only has to register once and recieves the number dialed in the incomming INVITE. If i use aliases to alias numbers to 1 account the account allways recieves it's own registered contact instead of the number dialed, is there any way to support number blocks in this way? For example: Device registeres as device_1@sipserver with contact device_1@device_ip:port SER gets invite for number in the block of device_1 and sends the invite as dialed_number@device_ip:port instead of device_1@device_ip:port Kind reagards, Erik Versaevel _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] Checking URI
by Ryan Pagquil 26 Aug '05

26 Aug '05

Hi Steve, Here is what happens with my current configuration. When I call from my iptel.org account going to my local users it prompts me "407, Proxy Authenctication Required". So what I like to do is to allow those calls going to my local users from other domain. BTW, here is my ser.cfg's INVITE message handler. ##INVITE message Handler## route[3] { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; # if (!proxy_authorize("", "subscriber")) { # proxy_challenge("", "0"); # break; # } else if (!check_from() && method=="INVITE") { sl_send_reply("403", "User From=ID"); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; if (method=="CANCEL") { route(1); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } I misse call with your SIP phone, just testing it. =) Thanks, Ryan Steve Blair wrote: > > Set the alias=<your sip domain> parameter then use if (uri==myself) > construct. > > -Steve > > Ryan Pagquil wrote: > >> Hello, >> How can I check if the call is for my domain? What specific >> module should I use? >> >> Thanks, >> > -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Serusers] failure_route question
by Java Rockx 25 Aug '05

25 Aug '05

Hi All. I'm using t_check_status() to do call forwarding on 486 Busy responses. I have an odd thing happening where SER receives the 486 Busy message (for which SER properly sends an ACK response) and then the failure_route calls append_branch() and t_relay() to try another destination. After the new leg of the call is setup and then torn down, the original 486 Busy seems to be hung in the tm module, because it magically appears in ngrep, which shows it being sent back to the caller as if the append_branch() never took place. Is there anything special that I should be doing to "swallow" the 486 Busy response to prevent it from being sent to the caller UA? Regards, Paul

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[Serusers] is that compliant?
by Olivier Taylor 25 Aug '05

25 Aug '05

Hi all, I have a problem with ringing message coming from our provider. Associate with the message, i find a SDP. Some UACs doesn't like that and doesn't give a ringback (Snom and Grandstream ata 486) SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.2.101;rport=1039;received=81.247.182.42;branch=z9hG4bKfeb085ba0caf55 99..To: <sip:0495283361@finalcut.be>;tag=507b8a49..From: " Family Taylor" <sip:7321073@finalcut.be>;tag=cf8413e3df61a5a5..Call-ID: ae6f02682bb43efd@192.168.2.101..CSeq: 4667 INVITE..Record-Route: <sip:82.146.123.252>..Content-Type: application/sdp..Content-Length: 217....v=0..o=VoipSwitch 7772 7772 IN IP4 213.246.41.194..s=VoipSIP..i=Audio Session..c=IN IP4 213.246.41.194..t=0 0..m=audio 6772 RTP/AVP 18 101..a=rtpmap:18 G729/8000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. Is that correct and compliant? Thanks, Olivier

1 0

[Serusers] Forward to busy and no answer
by Hernando Maisano 25 Aug '05

25 Aug '05

I need its aid, like setting the SER, I have still the problem with call forwarding on busy or no answer? ....I am to use a module AVPOP, but it does not walk. Please help Me..... Thanks

1 0

[Users] printer friendly documentation
by Gary Mart 25 Aug '05

25 Aug '05

First, I like what you are doing, but is their a better way to get printed copies of the documentation than to individually display and print the documentation for each module? Gary

2 1

[Users] Faster to do avp_printf();avp_check OR search()?
by Kanakatti M. Subramanya 25 Aug '05

25 Aug '05

e.g., if I want to do something if there is a Refer-To header in the message, am I better off doing avp_printf("$someAvp", "$rt"); avp_check("$rt", "re/.../i"); or search("/.../"); Just want ot know whether to get started rewriting all the "search()" strings i have in my ser.cfg.... cheers

5 7

[Users] LCR module
by Ozan Blotter 25 Aug '05

25 Aug '05

Dear List, I cannot find enough documentation about LCR Module, anyone using and successful SER.CFG for it ? Thanks, Ozan

2 2

[Users] mediaproxy logging
by Klaus Darilion 25 Aug '05

25 Aug '05

Hi! Is it possible to redirect the syslog output of mediaproxy and proxydispatcher to a certain file? e.g. as openser can: log_facility=LOG_LOCAL4 regards klaus

1 0

[Serusers] Checking URI
by Ryan Pagquil 25 Aug '05

25 Aug '05

Hello, How can I check if the call is for my domain? What specific module should I use? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

2 1

[Serusers] SERWEB
by Colin Jordan 25 Aug '05

25 Aug '05

I am having trouble to write to mysql from /htdocs/serweb/html/styles/user_interface/index.php. Where am I going wrong DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

1 0

[Serusers] RPM serweb
by Marc Khayat 25 Aug '05

25 Aug '05

Anybody knows where I can get the rpm package of the serweb for ser-0.9.3? Thanks,

1 0

[Serusers] Active and standby gateway
by Sam Lee 25 Aug '05

25 Aug '05

I have 2 voice gateway which apparently both routes the same path. I'm trying to use 1 as an active gateway where my SER routes the voice traffic to, the other used as a hot standby. Are there anyway i can specify this implementation in my configuration? Regards, Sam

2 1

[Serusers] Access Denied Error in SER setup
by Bhaskar Bhumkar 25 Aug '05

25 Aug '05

Hi, In the process of setting up the SER proxy server, I always get the following error : # /usr/sfw/bin/mysqladmin -h <my_machine_name> -u mysql password 'viperdodge' /usr/sfw/bin/mysqladmin: unable to change password; error: 'Access denied for user: '@<my_machine_name>' to database 'mysql'' I am referring to the following document for setting up the proxy server : http://developers.sun.com/solaris/articles/solaris_as_sip/solaris_as_sip.ht… I could finish upto step 5.3 i.e. I finished the test run successfully, however I am having a problem in setting up the root password. If I skip this step and go ahead, I get the following error whenever I try to use serweb using http://<my_machine_name>/admin : Database error: pconnect(<my_machine_name>, ser, $Password) failed. MySQL Error: () Session halted. I tried all the options as per the MYSQL manual which is available at : http://dev.mysql.com/doc/mysql/en/access-denied.html But it's not of much help to me. I will appreciate if somebody has any pointers for this. Thanks, Bhaskar...

2 1

[Serusers] If else using call source
by Sam Lee 25 Aug '05

25 Aug '05

Dear All, Can i do my call routing based on the calling party ? So far from what i see in the examples in iptel.org, it is using the request URI , which is the call destination. Any help ? eg. If i want to route all calls coming from 1234 to a number, how do i do it ? Please let me know if you don't understand my question. Regards, Sam

2 1

[Serusers] Radius Module acc_radius.so Loading Problem
by Fuchs Gernot 25 Aug '05

25 Aug '05

Hello Serusers, After deploying the binary distribution of SER 0.9.3 for Debian from the repository "deb http://apt.sip-router.org/debian/ stable main contrib non-free" I was able to install the RADIUS based AUTH against a FreeRADIUS server. However after loading the acc_radius.so module I encounter module loading errors as in: # WARNING: no fork mode and more than one listen address found(will use only the the first one) stateless - initializing textops - initializing acc - initializing ERROR: error while initializing modules # No other errors are encountered for "ser -c". When I remove the module acc_radius.so, everything is OK again and AUTH works fine against the RADIUS server. My ser.cfg for the modules looks like: # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_radius.so" loadmodule "/usr/lib/ser/modules/acc_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- RADIUS params -- modparam("auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # # Your help is very much appreciated! Kind regards, --Gernot

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[Serusers] PSTN gateway - Sipura 3000
by feedbak 25 Aug '05

25 Aug '05

I am using a Sipura 3000 with Ser 8.13 When I forward a PSTN call to the gatway. It disregards the fact that I have stripped the initial 9 using strip(1) and dials the 9 on the PSTN side. I have set the device for one stage Dialing. and have the following in my my ser.cfg # Check for PSTN1 if (uri~"^sip:9") { strip(1); t_relay_to_udp ("192.168.100.15", "5061"); break; }; Any ideas ? Thanks, Dinesh

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Re: [Serusers] Routing multiple numbers to 1 account
by Erik Versaevel 25 Aug '05

25 Aug '05

Disticintive ring is not an option as the device won't handle that, replacing the from would break Caller ID :) However, i might have an idea now. * Create aliases in alias table Invite comes in if(lookup(alias)) {#alias excists Append-hf("Called number: ruri\username") if(lookup(location)) { #user found, replace username with the called number from the Called number header field } That might work, i'll see if i can set this up in my test machine. Thanks again for your help, Erik Versaevel Chris St Denis wrote: >Ah, I see your problem. > >One option there is to either: >1. use distinctive ring to differentiate. Something like: > append_hf("Alert Info: Bellcore-r1\r\n"); >Or >2. Set the caller ID to be the called id with uac_replace_from or something. > >Other than that I can't suggest anything to help you. > >-----Original Message----- >From: Erik Versaevel [mailto:erik@infopact.nl] >Sent: Wednesday, August 24, 2005 2:01 PM >To: Chris St Denis >Cc: 'SER Users' >Subject: Re: [Serusers] Routing multiple numbers to 1 account > >Chris, > >So the invite still gets send to the contact uri sip_user registered in >ser ( in this case sip_user@1.2.3.4:5060), that's not what i want since >the end device won't know which number has been dialed in order to reach >it (it has no way to see that 5551234 has been dialed to get to it). >This looks like alias done by radius instead of the standard ser mysql >aliases. > >Kind regards and thanks for your help, > >Erik > > >Chris St Denis wrote: > > > >>Using the numbers from your example >> >>Call comes in to 5551234 >>5551234 is looked up in registration table. >>5551234 is not found in registration table. >>5551234 is looked up via radius which returns sip_user >>Invite is rewritten to be for sip_user >>sip_user is looked up in the registration table and the usual lookup action >>is taken. >> >>-----Original Message----- >>From: Erik Versaevel [mailto:erik@infopact.nl] >>Sent: Wednesday, August 24, 2005 1:52 PM >>To: Chris St Denis >>Cc: 'SER Users' >>Subject: Re: [Serusers] Routing multiple numbers to 1 account >> >>Chris, >> >>What exactly do you replace in this case? what i need to do is the >>following: >> >>user sip_user registers his contact in ser (ie sip_user@1.2.3.4:5060) >> >>SER recieves invite for 5551234(a)mysipserver.com, makes a lookup in a db >>table, finds that 5551234(a)mysipserver.com belongs to sip_user and >>forwards the message to 5551234@1.2.3.4:5060 (not sip_user@1.2.3.4:5060) >> >>This: >> avp_pushto("$ruri/username", "s:callee_vdid"); >>looks like you replace the username part of the request uri (in this >>case 5551234(a)mysipserver.com) with something else (which is what i kinda >>want to do), >> >>Kind regards, >> >>Erik >> >> >>Chris St Denis wrote: >> >> >> >> >> >>>I am doing this by pulling the translation out of a database using radius >>>AVP (could be done direct to database too) and sticking it as the >>>destination. >>> >>>It looks like this >>> >>> # Try to lookup the number directly registered. >>> if(!lookup("ser_location")) { >>> #Number isn't found, so try to do an avp translation on it >>> avp_pushto("$ruri/username", "s:callee_vdid"); >>> if(!lookup("ser_location")) { #Look up the new value >>> #Still can't find the user. Give up. >>> sl_send_reply("404", "I can't find 'em"); >>> break; >>> } >>> } >>> >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On >>>Behalf Of Erik Versaevel >>>Sent: Wednesday, August 24, 2005 6:49 AM >>>To: SER Users >>>Subject: [Serusers] Routing multiple numbers to 1 account >>> >>>Hello all, >>> >>>At this moment i'm using the full e164 numbers as account's in ser, that >>>is for every number someone wants delivered to them they would have to >>>register it using the account details. >>>This however poses a problem with a 100 bock of numbers for example >>>while a device can only register a few of them so what i would like to >>>do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way >>>the devices only has to register once and recieves the number dialed in >>>the incomming INVITE. If i use aliases to alias numbers to 1 account >>>the account allways recieves it's own registered contact instead of the >>>number dialed, is there any way to support number blocks in this way? >>> >>>For example: >>>Device registeres as device_1@sipserver with contact >>> >>> >>> >>> >>device_1@device_ip:port >> >> >> >> >>>SER gets invite for number in the block of device_1 and sends the invite >>>as dialed_number@device_ip:port instead of device_1@device_ip:port >>> >>>Kind reagards, >>> >>>Erik Versaevel >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> >>> >>> >>> >>> >> >> >> >> > > > > -- InfoPact Netwerkdiensten B.V. http://www.infopact.nl/ Emmastraat 11-13 3255 BD Oude Tonge tel. +31 (0)187 64 77 11 mob. +31 (0)645 18 69 67 fax. +31 (0)187 64 77 99

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[Serusers] Routing multiple numbers to 1 account
by Erik Versaevel 24 Aug '05

24 Aug '05

Hello all, At this moment i'm using the full e164 numbers as account's in ser, that is for every number someone wants delivered to them they would have to register it using the account details. This however poses a problem with a 100 bock of numbers for example while a device can only register a few of them so what i would like to do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way the devices only has to register once and recieves the number dialed in the incomming INVITE. If i use aliases to alias numbers to 1 account the account allways recieves it's own registered contact instead of the number dialed, is there any way to support number blocks in this way? For example: Device registeres as device_1@sipserver with contact device_1@device_ip:port SER gets invite for number in the block of device_1 and sends the invite as dialed_number@device_ip:port instead of device_1@device_ip:port Kind reagards, Erik Versaevel

2 4

[Serusers] as it is the logica of scripts?
by Walter Willis 24 Aug '05

24 Aug '05

I was testing with scripts of serr one receives calls of asteriskk and the other can make calls to asterisk? I see that metodo INVITES in one of scripts is repeated twice, in any case podrian to give a serious guide to me of as script in form generates . my idea is to cause that being can receive and send calls to asterisk. thanks

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RE: [Serusers] Routing multiple numbers to 1 account
by Igor Balk 24 Aug '05

24 Aug '05

Guys, do any of you know if there is a way in SER to dynamically set up aliases using radius? I'm trying to achieve similar results but with aliases. Thank you. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Chris St Denis Sent: Wednesday, August 24, 2005 1:26 PM To: 'Erik Versaevel'; 'SER Users' Subject: RE: [Serusers] Routing multiple numbers to 1 account I am doing this by pulling the translation out of a database using radius AVP (could be done direct to database too) and sticking it as the destination. It looks like this # Try to lookup the number directly registered. if(!lookup("ser_location")) { #Number isn't found, so try to do an avp translation on it avp_pushto("$ruri/username", "s:callee_vdid"); if(!lookup("ser_location")) { #Look up the new value #Still can't find the user. Give up. sl_send_reply("404", "I can't find 'em"); break; } } -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Erik Versaevel Sent: Wednesday, August 24, 2005 6:49 AM To: SER Users Subject: [Serusers] Routing multiple numbers to 1 account Hello all, At this moment i'm using the full e164 numbers as account's in ser, that is for every number someone wants delivered to them they would have to register it using the account details. This however poses a problem with a 100 bock of numbers for example while a device can only register a few of them so what i would like to do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way the devices only has to register once and recieves the number dialed in the incomming INVITE. If i use aliases to alias numbers to 1 account the account allways recieves it's own registered contact instead of the number dialed, is there any way to support number blocks in this way? For example: Device registeres as device_1@sipserver with contact device_1@device_ip:port SER gets invite for number in the block of device_1 and sends the invite as dialed_number@device_ip:port instead of device_1@device_ip:port Kind reagards, Erik Versaevel _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] VoIp and Java
by javabuddy 24 Aug '05

24 Aug '05

Is there any available api for java on VoIp. - javabuddy. --------------------------------------------------------------------------- Broadband/rich interface + mail box saftey = Roomity.com. ~~1124911739820~~ SIP_Router_Users_List.roomity.com no sign up to read, ad supported. ---------------------------------------------------------------------------

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[Serusers] call transfer process
by Iqbal 24 Aug '05

24 Aug '05

Hi If I have a inbound call frompstn which hits SER, and then the call is passed to asterisk, the receiver then decides to transfer the call, can this be controlled within asterisk/ser, without the PSTN gateway being involved, I was thiking that although REFER messages are sent, couldnt SER take care of this and just transfer the call to a new sip contact. The problem I have got is that I have a upstream gateway which doesnt support REFER :-) Iqbal

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[Serusers] SER default p-time 20 ms
by Frank Kostin 24 Aug '05

24 Aug '05

Hello, I try debugging and get this error "found match with different intervals (20 != 30)" Is it possible to change the p-time into SER from 20 to something else like 30 or 40 ? Thanks, frank --------------------------------- Start your day with Yahoo! - make it your home page

2 1

[Serusers] ACKs not getting routed?
by Chris St Denis 24 Aug '05

24 Aug '05

I am having a problem with incoming calls where ACK messages dont seem to be getting (loose) routed. Here is the relevant part of my config ====================== route { xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> - %tu\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops -- probebly a loop or something"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy for accounting purposes record_route(); #Accounting if a call is started or ended if (method=="INVITE" || method=="BYE" || method=="CANCEL") { acc_rad_request("200"); } # grant Route routing if route headers present # loose-route processing. Decreases security slightly regarding # forged messages, but decreases load. if (loose_route()) { xlog("L_NOTICE", "Loose Routing\n"); t_relay(); break; }; ====================== My understand is this should cause the ACK messages that are part of the transaction to get routed. They should not pass beyond this block of code right? However, the SIP conversation goes as follows. Gateway invites proxy Proxy says 100 trying Proxy says 100 trying Proxy invites user User says 100 trying User says 180 Ringing User says 180 Ringing Thus far all is well When the user picks up the phone User tells proxy 200 ok Proxy tells gateway 200 ok Gateway tells proxy ACK At this point the proxy should be relaying the ACK to the user, but it does not. Instead due to timeout the user sends another 200 ok and that last block of conversation repeats. Any ideas why this would not be working? (I tried to send this with a full SIP dialog but it got rejected as being too big so this email is the chopped down version)

1 1

[Users] configuring multi-domain support
by jayesh nambiar 24 Aug '05

24 Aug '05

Hi all, I was trying to configure my script for hosting multiple domains. I went through the admin's guide and understood the procedure of creating new subscriber table for each domain and serve them. But I would be very thankful if someone let me know how does automated multi-domain support works and how do we maintain multiple domains using automated support. I think using domain table will help, but any information regarding serving multiple domains in an efficient manner will be very helpful. Thanx a lot. jayesh.. --------------------------------- Start your day with Yahoo! - make it your home page

3 2

[Users] BRANCH route support
by Bogdan-Andrei Iancu 24 Aug '05

24 Aug '05

Hi everybody, I just finished adding support for BRANCH routes. The idea and frame were partial ported from SER (thanks to Juha for extracting and adapting patches). For those who are not familiar with this concept: branch route is a route to be execute separately for each branch before being sent out - changes in that route should reflect only on that branch. I say partial because the features, implementation were improved. Shortly here is a list: - remove from grammar default BRANCH ROUTE definition - since this is a 100% TM route, make no sense - RURI may be changed from a branch route (the routing will accordingly change); This capability is essential, since there are core function which alter the RURI, functions which cannot be blocked in specific route types - not handling this, may easyly lead to memory faults or leaks. - flags per branch - if flags are changed in branch_route, the change will not be visible for other branches, but it will be visible for all replies of that branch (it might be useful for per branch NAT traversal) - full AVP support in branch route - more efficient handling of lumps (changes in request) - instead of cloning and removing the entire lump tree for each branch, using lump flags, after each branch, the original lump tree will be just cleaned only what was added during branch route will be removed). still pending: how to handle append_branch() core function.....I will prefer to handle properly all core functions, since their execution cannot be banned from specific routes. For the moment only textops, avpops and xlog modules were enabled for branch route. Other will come in the future. regards, Bogdan PS: testing done only superficial (is quite late now).....any help/reports are welcomed :D

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[Serusers] SER IP PBX for multiple clients
by Waldo Rubinstein 24 Aug '05

24 Aug '05

Hello, I'm still trying to learn more about SER. I've been using Asterisk to manage virtual PBX services for different companies by using multiple contexts within Asterisk. However, since I only use Asterisk with SIP UAs and to communicate with ITSPs, I don't have the need to have all the fancy features Asterisk offers, plus I have the additional advantage of having the built-in NAT support in SER. The question I have is if someone can point me to the right place where I can see some sample configs that do more or less the things I need or if someone would be willing to share some of those configs on the list. Basically, I need to have the ability to manage any number of virtual PBX services where each virtual entity can manage their own extension numbering, DIDs, outbound CLID. I would probably just continue using Asterisk for voicemail services since I get the feeling that SEMS is still "unstable" to go into production, even for voicemail services only (unless told otherwise). The idea is that even if two companies define two extensions 101 and two extensions 102, when each company calls ext 101 or 102 internally, the call will be maintained within that company and would not cross over to the other company. Also, having features like one main number (DID) that could ring in multiple extensions simultaneously (forking) and if no one answers, leave a message in the general mailbox, or that each extension could have, additionally, a DID so that they can be reached directly from the outside. Any help would be greatly appreciated. Thanks, Waldo

3 5

[Serusers] SEMS + SER
by c waddy 24 Aug '05

24 Aug '05

Hi, I have version 0.8.14 of ser up and running and would now like to allow conference call bridging. >From what i have read is to use SEMS? Can someone point me to some documentation or provide me with an example to integrate this into my SER server. I cant find much information on this module.

1 0

[Serusers] invite timers
by Jeremy George 24 Aug '05

24 Aug '05

I need to set inv_timer to at least 30 seconds to account for the long failover times apparently used by cellular providers. Not hard to understand; it takes a while to dig that cell phone out of the pocket. But this seems to cause inbound calls to take the same long time to failover to voicemail (an Asterisk system running on a separate box.) Do I understand the timers correctly? Is there a way to cause a relatively short inbound failover and keep a longish outbound failure? - Jeremy

2 2

RE: [Fwd: Re: [Serusers] ul_add flag and serctl]
by Aisling 24 Aug '05

24 Aug '05

Hi Iqbal, Jan, I changed my log to /tmp/fifo in my ser.cfg and the serctl.dat file. I then tried adding an alias: Serctl alias add 7890 sip:aisling@x.x.x.x Again I got the 404 flags expected error and I checked the /tmp/fifo file. It was empty. I then did /opt/ser/sbin/serctl fifo ul_add aliases aisling(a)x.x.x.x 7890 0 1.00 0 128. (It would work unless I added those 4 parameters) It said 200 Added to table and the following appeared in my new log file: :ul_add:ser_receiver_28642 aliases aisling(a)x.x.x.x 7890 However while this said added above. When I dialed 7890 to reach the aisling(a)x.x.x.x client, a 404 was sent to my phone. Many Thanks, Aisling. -----Original Message----- From: Iqbal [mailto:iqbal@gigo.co.uk] Sent: 24 August 2005 12:29 To: Aisling O'Driscoll Subject: [Fwd: Re: [Serusers] ul_add flag and serctl] could try this -------------------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] lcr module
by Sebastian Kühner 24 Aug '05

24 Aug '05

Hello, I have a problem with the lcr module if a gateway has "big" IP Address. I calculate the integer of the IP like this: a.b.c.d ==> IP-Integer = d*pow(2,24) + c*pow(2,26) + b*pow(2,8) + a If the d-value is very big (100 or more), the resulting IP Address doesn't work anymore. What is the greatest that an integer can have? I use postgres with this module. I hope that this won't be a problem! Thanks! Sebastian

4 4

[Serusers] www_challange and proxy_challange
by Thomas Britis 24 Aug '05

24 Aug '05

Hi all, Is it sane to use: if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; } else { if (!www_authorize("", "subscriber")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; }; }; }; Or anything here appears to be wrong ? Thank you. -- Thomas Storino Britis TCNet Inf e Telecom LTDA

4 5

[Serusers] Best bersion of radiusclient to use with ser-0.9.0
by Ashutosh Kumar 24 Aug '05

24 Aug '05

Hi, Which is the best known stable version of radiusclient, radiusclient4.x or radisuclient5.x, or radiusclient-ng to use with ser-0.9.x, more specifically, ser-0.9.0. I have been bewildered by the earlier archived posts of this list some of which advocate radiusclient5.x, and some advocate radisuclient4.x, while in some , it is mentioned that radiusclient-ng works very fine with ser-0.9.x, but my ser is unable to even detect radiusclient-ng (it keeps loading the old radiusclient5.1 so's, despite the fact that rc-ng has been compiled from source and ldconfig'ed ). An authoritative response will be appreciated. Regards, Ashutosh Kumar

2 1

[Serusers] ser reinstall problem connected with mysql module load
by Richard Luo 24 Aug '05

24 Aug '05

Hi all, Thanks for the tips. I tried to reinstall my ser. not by rpm file. but compiling source code and make install. Then I met the problem when i running the ser: ================================================== [root@pc5 ~]# /usr/local/sbin/ser : command not found: line 6: : command not found: line 8: : command not found: line 12: /usr/local/sbin/ser: line 13: /bin: is a directory : No such file or directory6: */ : command not found: line 17: : command not found: line 24: : command not found: line 26: : command not found: line 29: /usr/local/sbin/ser: line 30: loadmodule: command not found /usr/local/sbin/ser: line 31: loadmodule: command not found /usr/local/sbin/ser: line 32: loadmodule: command not found /usr/local/sbin/ser: line 33: loadmodule: command not found /usr/local/sbin/ser: line 34: loadmodule: command not found /usr/local/sbin/ser: line 35: loadmodule: command not found /usr/local/sbin/ser: line 36: loadmodule: command not found : command not found: line 37: : command not found: line 42: : command not found: line 44: : command not found: line 46: /usr/local/sbin/ser: line 47: syntax error near unexpected token `"usrloc",' 'usr/local/sbin/ser: line 47: `modparam("usrloc", "db_mode", 0) =================================== but my ser.cfg is already revised to the required one. as: ========================= [root@pc5 ~]# cd /usr/local/etc/ser [root@pc5 ser]# ls dictionary.ser old ser.cfg ser.cfg~ ser.cfg.edited ser.cfg.old ser.cfg.sample [root@pc5 ser]# cat ser.cfg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "//usr/local/lib/ser/modules/mysql.so" loadmodule "//usr/local/lib/ser/modules/sl.so" loadmodule "//usr/local/lib/ser/modules/tm.so" loadmodule "//usr/local/lib/ser/modules/rr.so" loadmodule "//usr/local/lib/ser/modules/maxfwd.so" loadmodule "//usr/local/lib/ser/modules/usrloc.so" loadmodule "//usr/local/lib/ser/modules/registrar.so" loadmodule "//usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//usr/local/lib/ser/modules/auth.so" loadmodule "//usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("iptel.org", "subscriber")) { www_challenge("iptel.org", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ================================ Sorry about too much copy files. But could anyone can tell me where the problem is and what the solution? Thanks a lot. -- _______________________________________________ Search for businesses by name, location, or phone number. -Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.as…

1 0

[Serusers] radiusclient-ng/SER attributes for accounting
by Alberto Cruz 24 Aug '05

24 Aug '05

Hi guys, does anybody knows if it's possible to give a format to the "calling-from" and "calling-to" values that radiusclient-ng/SER send to the Radius server? I have a licensed billing application that is expecting to receive the "calling-from" and "calling-to" values like in a number format in order to apply the rates this mean: if "calling-from" or "calling-to" value is 836572345(a)sip.domain the billing application needs to receive 836572345 without the @sip.domain What should i do at SER or radiusclient-ng in order to achieve this format? Regards Alberto Cruz

3 4

[Serusers] User-Agent: Asterisk PBX.
by fedora＠voice.dyndns.tv 24 Aug '05

24 Aug '05

# U 2005/08/20 00:13:30.111494 221.186.150.67:5060 -> 221.186.150.68:5060 BYE sip:8000@221.186.150.68:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:221.186.150.67;ftag=as1cbee789;lr=on>. Via: SIP/2.0/UDP 221.186.150.67;branch=z9hG4bK1e0d.6051d181.0. Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062. From: <sip:500@voice.dyndns.tv>;tag=as1cbee789. To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543. Contact: <sip:500@221.186.150.67:5062>. Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68. CSeq: 102 BYE. User-Agent: Asterisk PBX. Content-Length: 0. Route: <sip:8000@221.186.150.68:5060>. . # U 2005/08/20 00:13:30.147446 221.186.150.68:5060 -> 221.186.150.67:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 221.186.150.67;branch=z9hG4bK1e0d.6051d181.0. Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062. From: <sip:500@voice.dyndns.tv>;tag=as1cbee789. To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543. Contact: <sip:8000@221.186.150.68:5060>. Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68. CSeq: 102 BYE. Server: X-Lite release 1103m. Content-Length: 0. . # U 2005/08/20 00:13:30.149017 221.186.150.67:5060 -> 221.186.150.67:5062 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062. From: <sip:500@voice.dyndns.tv>;tag=as1cbee789. To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543. Contact: <sip:8000@221.186.150.68:5060>. Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68. CSeq: 102 BYE. Server: X-Lite release 1103m. Content-Length: 0.

4 3

[Serusers] ul_add flag and serctl
by Iqbal 24 Aug '05

24 Aug '05

Hi been having a offlist discussion with Aisling, who has a problem with using serctl to add alias. The error we get is ul_add flag expected error 1. We have looked at mysql logs, and there is nothing sent, hence error must be in script/fifo 2. I have tested the same script on my server, and it works with no errors 3. We have used serctl ul_add to send raw comannds, and again it works on aisling's server also 4. Ran a diff between my working sercl and aisling's and there is no diff. The flow I think of info is serctl ---> fifo --> DB is there anyway of seeing what fifo is doing, or any other thoughts, this error DOES NOT invlove serweb, or its config scripts, its just serctl on its own Iqbal

2 1

[Serusers] t_on_failure and t_on_reply in the same block
by maka 24 Aug '05

24 Aug '05

Hi all, Can I have t_on_reply() and t_on_failure() in the same ser route block, i.e. in the route block processing INVITEs? On-replied transactions are processed in the on_reply route block for nat traversal, but how do I specifically catch the 486 busy message if want to play a voice message for them? cheers

2 1

[Serusers] speeding up db update
by Johan 24 Aug '05

24 Aug '05

I noticed that SER will take some time to update location table everytime I quit from my UA. Is there anything that I can do to speedup this update ? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] Adding "Allow" to the reply
by Federico Giannici 24 Aug '05

24 Aug '05

According to RFC3261, a reply of code 405 (Method Not Allowed) MUST include an "Allow" header field. How can I add this header field? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 1

Re: [Serusers] Distributing "OSP enable SER" binary file w/ source
by vmathur＠transnexus.com 24 Aug '05

24 Aug '05

Hi Jan, You had mentioned in your mail that you provide binary tarballs and packages, which are kept in local repositories. Further to that, I have the following questions: 1. Could you tell me the location of the repository. 2. I am guessing that in order for me to post our package there, I would need to send it to you or is it possible for me to post it directly. 3. When you say "packages", you mean rpm ? 4. Finally, is it possible for me to create a binary of OSP enabled SER and post it there ? Many Thanks, Vikrant ------------------------------------------------------------------------------------------------------------- Yes, we provide binary tarballs and binary packages. In case of binary packages we even have our own repositories for Redhat and Debian distributions (so if you create packages of the library then you could put it there). Jan.

2 1

[Serusers] generating password by hand
by Roger Thomas 24 Aug '05

24 Aug '05

Hi, I have a user registration that writes user info to ldap database. Whenever a new user registers, I would like to populate SER's subscriber table too. How to I generate the HA1 password with PHP ? Please advise. ../zhza --------------------------------------------------- Sign Up for free Email at http://ureg.home.net.my/ ---------------------------------------------------

4 4

[Serusers] unix socket interface documentation?
by Federico Giannici 24 Aug '05

24 Aug '05

I have seen that, besides the fifo interface, SER has an unix socket interface too. Is there any documention or at least any example of usage? What are the advantages over the fifo interface? If I'm correct, serctl uses only the fifo interface, right? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

2 1

[Serusers] Sems troubleshooting
by Pavol Segeč 24 Aug '05

24 Aug '05

Hi, May somebody give me some suggestion how to troubleshoot sems? I need to troubleshhoot sems voicemail, I configured ser (0.9.3) + sems. I'd like use sems voicemail, I configured ser and sems following Configure-Sems-Ser-HOWTO with pasing mail info through fifo_db_url. I'm receiving voicemail announcement about leaving voice message, but e-mail with voice mail is not sent. In syslog I don't see any error, mailq is empty, message wasn't received by postix. I tried also configure mail passing through tw_append, but setting described in manual is probably out of date, because I have some database reading problems with it. Thanks pavol

1 0

[Serusers] (no subject)
by share phone 23 Aug '05

23 Aug '05

Hi All, Mediaproxy help needed is this working with openser. Regards, Sunil --------------------------------- Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

1 0

[Serusers] missed_calls username and domain
by Ah khng 23 Aug '05

23 Aug '05

Hi all, I want to display missed calls in SERWEB, so i use acc_db_request to store the entry into table missed_calls. However, username and domain weren't recorded into database. While i noticed that serweb is comparing username and domain in sql WHERE cluase (method.get_missed_calls.php) which cause no records being display. Thus, how do i make domain and username recorded into database? # -- accounting parameters -- modparam("acc", "log_level", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) failure_route[1] { # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted acc_db_request("missed calls","missed_calls"); revert_uri(); if (t_check_status("486") || t_check_status("603")) { log(1,"User busy\n"); } else if (t_check_status("480") || t_check_status("408")) { log(1,"User is unavailable\n"); } else { log(1,"User is unavailable for unknown reason\n"); } rewritehostport("202.73.10.7:5060"); append_branch(); t_relay_to_udp("202.73.10.7", "5060"); } mysql> select sip_from,sip_status,sip_method,username,domain from missed_calls; +----------------------------------------------+--------------+------------+----------+--------+ | sip_from | sip_status | sip_method | username | domain | +----------------------------------------------+--------------+------------+----------+--------+ | <sip:10001@202.73.10.8>;tag=ff2015eeb8c23171 | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=ff2015eeb8c23171 | missed calls | CANCEL | n/a | n/a | | <sip:10001@202.73.10.8>;tag=688a799e25d70d76 | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=9da5279e291e12cc | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=1fbde47420f30ff2 | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=bde153a0cf74ce23 | missed calls | INVITE | n/a | n/a | | <sip:10001@202.73.10.8>;tag=bc3362f1cb8b0f64 | missed calls | INVITE | n/a | n/a | +----------------------------------------------+--------------+------------+----------+--------+ 9 rows in set (0.00 sec) Regards Seong

2 1

[Serusers] call through
by Iqbal 23 Aug '05

23 Aug '05

Has anyone implemented a call through scenario eg dial 001 234512367, and thats hits ser, then press # (am thinking this will not work just with SER) dial the next number, and the call is terminated at the second number. 2 stage dialing. Iqbal

2 1

[Serusers] "stun" parameters in Contact HF
by Massimo Barletta 23 Aug '05

23 Aug '05

Hi all, I'm testing a new UA and I've got a Contact Header Field format in REGISTER msg I'm not aware of: REGISTER sip:/sip.mydomain.com/:5060 SIP/2.0. Via: SIP/2.0/UDP *0.0.0.0:0;*branch=z9hG4bK2497750963. From: <sip:/pstn_number/@/sip.mydomain.com/>;tag=3670695360. To: <sip:/pstn_number/@/sip.mydomain.com/>. Call-ID: 2377110845(a)192.168.1.4. CSeq: 1 REGISTER. Contact: <sip:/pstn_number/*@0.0.0.0:0;**stun=4*>. max-forwards: 70. expires: 3600. user-agent: oSIP/Linphone-1.1.0-IMVP. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO. Content-Length: 0. i.e. ip:port is no longer in Contact or Via Header Fields and a stun=4 parameter is specified in Contact HF Have you got any clue on RFC reference for this behaviour? How can I handle this on SER? (SER fills location table as follows: username: /pstn_number/ domain: /sip.mydomain.com/ contact: sip:/pstn_number/@0.0.0.0:0;stun=4 received: sip:/UA_ip_address/:5060 ) THX, Mety

2 1

[Serusers] Developers' Guide
by Ahmed Kellal 23 Aug '05

23 Aug '05

Hello, I was reading the SER DG when I came upon a sentence (twice on the same page) saying that there is a chapter dealing with, and I quote, SIP Message Modifications. However, the said chapter does not exist in the 0.8.8 version of the document. Can someone shed some light on the subject? :-) Cause I really need to understand the workings of that part of the server. Thanks in advance Ahmed _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[Serusers] Compiling SER to use radiusclient-ng library
by Ashutosh Kumar 23 Aug '05

23 Aug '05

Hi, I want to use radiusclient-ng library with my SER. Previously, I was using radiusclient. Now that I have installed from source radiusclient-ng on the machine and also configured SER to sue, it still seems to be using old radiusclient, and not radiusclient-ng, because when I delete the modules libradius.so.* from my machine, then I get the error by SER "libradiusclient.so.2: cannot open shared object file: No such file or directory", which surely indicates that SER is configure to use only radiusclient at compile time . Is there any switch/ directive while compile time to indicate that SER should use radiusclient-ng , and not radiusclient? 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

2 1

[Users] avp_write
by Richard Z 23 Aug '05

23 Aug '05

Hi, If there are multiple branches, will avp_write("$ruri/username","i:123") generate multiple avps with each for one branch? Or it only grabs the first one? Thanks, Richard

3 8

[Serusers] help: fix_nated_contact_f
by Ahmed Kellal 23 Aug '05

23 Aug '05

Hello, I am trying to add a function to SER, in order to only alter the host part of the Contact URI. To do that, I'm trying to draw inspiration from the existing fix_nated_contact_f function written in nathelper.c Problem is there are various points I just don't seem to understand. If anyone can help me answer these questions, I would sure appreciate it!! 1- the parameters declared for the fix_nated_contact_f function are (struct sip_msg* msg, char* str1, char* str2) But the code itself only exploits the first parameter, if I am not mistaken. If that is the case, what are the others here for? 2- the lines: if (uri.port.len==0) uri.port.s = uri.host.s + uri.host.len; What I understand from this is the following: If there isn't a port specified in the original message (the port pointer is not pointing to anything), set the pointer's value (the pointer in the port part of the URI structure) to point to the memory location right after where the host part in the URI ends. Is that correct? And, if so, is that done in order to prepare for an eventual insertion of a port when going through the NAT? 3- the line: offset = c->uri.s - msg->buf; I won't even venture a guess here. A total mystery to me. 4- the line: len = c->uri.len + strlen(cp) + 6 /* :port */ - (uri.port.s + uri.port.len - uri.host.s) + 1; What I gather here is that we're computing the relative difference in size (length) between the already existing ip:port field in the message, and the one we're about to replace it with (source address of the packet), is that right? 5- the part: temp[0] = uri.host.s[0]; temp[1] = c->uri.s[c->uri.len]; Euh!! Is he taking just the first and last characters from the URI host string, putting them in a "safe place" (to be later restored) and putting '\0' characters instead to "delimit" something or other? Or am I totally off mark??? 6- the line: len1 = snprintf(buf, len, "%s%s:%d%s", c->uri.s, cp, msg->rcv.src_port,uri.port.s + uri.port.len); Here, len1 will hold the size of the memory size (buffer) needed to insert the new data, right? 7- the condition: if (len1 < len) len = len1; You'll have to enlighten me on that one, cause I don't see it... I'm sorry to bother you with such a long mail, but I need the help! And if someone has a better approach to my original problem(cf. first line of the mail) please do not hesitate :-) Thank you in advance _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[Serusers] ser register at external SIP-Server
by Müller Thorsten 23 Aug '05

23 Aug '05

Hello all, is it possible to send a REGISTER with username and password to an external SIP-Server right after starting up of ser? I thought at uac-Module or tm-Module, any other suggestions? Thanks Thorsten

2 1

[Serusers] serweb on different server
by Johan 23 Aug '05

23 Aug '05

I have ser running on svr1. Would it be possible to run serweb from another server, say, svr2 ? What do I need to achieve this ? TIA. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] openser :
by share phone 23 Aug '05

23 Aug '05

Hi All, lookup(): '@192.168.1.66' Not found in usrloc my SIP_DOMAIN was sharephone.networks.com and my IP Address is 192.168.1.66 and i added the FQDN in the /etc/hosts file and when i use the UA to register it registers but after a while i get the folloing message in the log. Thanks, Sunil. 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) lookup(): '@192.168.1.66' Not found in usrloc 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) lookup(): '@192.168.1.66' Not found in usrloc 2(14038) parse_headers: flags=ffffffffffffffff 2(14038) check_via_address(192.168.1.65, 192.168.1.65, 0) 2(14038) DEBUG:destroy_avp_list: destroying list (nil) 2(14038) receive_msg: cleaning up 2(14038) SIP Request: 2(14038) method: <OPTIONS> 2(14038) uri: <sip:192.168.1.66:5060> 2(14038) version: <SIP/2.0> 2(14038) parse_headers: flags=2 2(14038) Found param type 235, <rport> = <n/a>; state=6 2(14038) Found param type 232, <branch> = <z9hG4bKc0a801410000000b430a4e92000005dd000005ec>; state=16 2(14038) end of header reached, state=5 2(14038) parse_headers: Via found, flags=2 2(14038) parse_headers: this is the first via 2(14038) After parse_msg... 2(14038) preparing to run routing scripts... 2(14038) parse_headers: flags=100 2(14038) DEBUG: get_hdr_body : content_length=0 2(14038) get_hdr_field: cseq <CSeq>: <456> <OPTIONS> 2(14038) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(14038) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 2(14038) DEBUG: add_param: tag=286578432823134 2(14038) end of header reached, state=29 2(14038) parse_headers: flags=200 2(14038) end of header reached, state=9 2(14038) DEBUG: get_hdr_field: <To> [25]; uri=[sip:192.168.1.66:5060] 2(14038) DEBUG: to body [<sip:192.168.1.66:5060> ] 2(14038) found end of header 2(14038) find_first_route: No Route headers found 2(14038) loose_route: There is no Route HF 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) lookup(): '@192.168.1.66' Not found in usrloc 2(14038) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(14038) grep_sock_info - checking if port 5060 matches port 5060 2(14038) lookup(): '@192.168.1.66' Not found in usrloc 2(14038) parse_headers: flags=ffffffffffffffff 2(14038) check_via_address(192.168.1.65, 192.168.1.65, 0) 2(14038) DEBUG:destroy_avp_list: destroying list (nil) 2(14038) receive_msg: cleaning up --------------------------------- Start your day with Yahoo! - make it your home page --------------------------------- Start your day with Yahoo! - make it your home page

2 1

[Users] ruri, dset, branches ...
by Klaus Darilion 23 Aug '05

23 Aug '05

Daniel-Constantin Mierla wrote: > Looking at this issue, I am wondering if worth to add the > pseudovariables $dh and $dp for destination host and port ... Also I > am considering to change a bit the $du reference so when the dst_uri > is not set to point to request URI. Would be these changes usefully? Hi Daniel! This reveals another question: What is the difference between: request URI, first branch, all branches, destionation sets ..... I used xlog to log all this pseude variables just before t_relay. first branch: $br=sip:klaus@10.10.0.50:8718 all branches: $bR=sip:klaus@10.10.0.50:8718 destination set: $ds=Contact: sip:klaus@83.136.33.19:5060, sip:klaus@10.10.0.50:8718 destination uri: $du=<null> request uri: $ru=sip:klaus@83.136.33.19:5060 In the above call scenario, there is just lookup() which reveals 2 contacts, one with public IP and one with NATed IP (received structure used). Can you please describe the difference of these variables? - Why does the destination set contains the NATed IP instead of the received-IP? - Why does all branches only shows the first one? - Why is the request URI different to the first branch? regards klaus

4 5

[Serusers] ser.cfg problem cont.
by Richard Luo 23 Aug '05

23 Aug '05

Hi, I tried to uncomment the ser.cfg file as the instruction to use mysql. but i met some problem. the sys log file shows: Aug 22 19:13:52 pc5 ser: ERROR: load_module: could not open module </usr/lib/ser/modules/mysql.so>: /usr/lib/ser/modules/mysql.so: cannot open shared object$ Aug 22 19:13:52 pc5 ser: parse error (28,13-14): failed to load module Aug 22 19:13:52 pc5 ser: set_mod_param_regex: No module matching auth_db found | Aug 22 19:13:52 pc5 ser: parse error (56,21-22): Can't set module parameter Aug 22 19:13:52 pc5 ser: set_mod_param_regex: No module matching auth_db found | Aug 22 19:13:52 pc5 ser: parse error (61,20-21): Can't set module parameter could anyone can help? ============================ You won't admit you love me, and so How am I ever to know You always tell me... Perhaps, perhaps, perhaps -- _______________________________________________ Search for businesses by name, location, or phone number. -Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.as…

4 3

[Serusers] ser+mediaproxy behind nat
by jeff kwong 23 Aug '05

23 Aug '05

Hi everyone! I am currently trying out the examples from the Getting Started manual version 04a from Onsip.org. The reference design there shows that SER itself is nated. My question is how will a UA from the public internet or somewhere outside the SER's nat register with the SIP server? Will I put the SER's nat public ip as the sip proxy and do I have to set some kind of port forwarding on the SER's nat? Also, doesn't mediaproxy have to listen on a public ip? The document is actually a good tutorial and just need additional clarifications for issues like nat. Thanks Guys!

2 2

[Users] UAC module: parameter not found
by Franz Edler 22 Aug '05

22 Aug '05

Hi, I am just testing UAC module (development version). Using the parameter "from_restore_mode" is not possible because it is not found in module uac. Debug log: set_mod_param_regex: parameter <from_restore_mode> not found in module <uac> This is strange, because it is defined in the list of exported parameters of uac.c module v 1.4. Any ideas? Franz

4 6

[Serusers] Radius packets not being sent to radius server
by Ashutosh Kumar 22 Aug '05

22 Aug '05

Hi, I m using radius with SER-0.9.0 and also using all the reqd functions viz. www_raidus_authorize etc... , but when I ngrep for any connection with my radius server, I don't get any radius-packets in the result. Also the log file contains "rc_send_server: no reply from RADIUS server a.b.c.d.:1812" (a.b.c.d is the radius server's IP). This is despite the fact that the radius server is running and hosting another running service, also radisuclient is configured fine. Kindly guide. 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

2 3

[Serusers] Mediaproxy
by Juan 22 Aug '05

22 Aug '05

Hi. I am trying to have mediaproxy starting on reboot but when doing chkconfig --add mediaproxy I am getting: service mediaproxy does not support chkconfig. How can I make it to start on reboot, then? Thanks Juan

1 0

[Serusers] Missed Calls not being displayed in Serweb
by Rafael R. GV 22 Aug '05

22 Aug '05

Hi I have ser-0.9.3 and Serweb-0.9.3, mysql4 and accounting works correcty but missed calls not being displayed in serweb. I have read in the list that method.get_missed_calls.php script from serweb require domain column in aliases tables and username/domain columns in missed_calls table and I have the values "n/a" in those columns so if that the case, How can I set these values correctly? any ideas? thanks Rafael Risco -- rrgv

1 1

[Serusers] Timeout
by Sebastian Kühner 22 Aug '05

22 Aug '05

Hi! Does anyone of you know how to stop the timeout timer manually? Thanks! Sebastian

3 7

[Serusers] reroute
by Jason Mohler 22 Aug '05

22 Aug '05

Hello everyone, I am novice with SER reroute. Can anyone help explain to me how I can set my SER up to reroute when I get a particular SIP message back (e.g. 503 ) and route to another IP. Your help with this matter will be greatly appreciated.

1 0

[Users] trusting src_ips using AVP
by jayesh nambiar 22 Aug '05

22 Aug '05

hi all, I went through the avp docs in voicesystem.ro and i thought the source_ip trusting in multi domain environment would be useful to me. I used permissions to do this before, but i think multidomain support is not possible in permissions module. But I jus wanted to know that on updating the table how does openser get updated without restarting the server. Is there a FIFO command or something. pls let me kno cause I did not understand the example mentioned in the docs properly. One more thing, Does Openser 0.9.5 support use_domain parameter mentioned in the docs. thanx.. jayesh ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

3 4

[Serusers] Problem with natted X-lite and public gateway
by Carlo 22 Aug '05

22 Aug '05

Hi all, I've a problem beetween X-lite softphone and Cisco Sip Gateways. Both SER and Cisco Gateways are configured with public ip addresse, while the x-lite is behind a nat router. It seems that the x-lite doesn't take into account the ser ip address in SDP connection information field but it sends rtp to the ip of the cisco gateway. Can someone help me ? How can I try to solve the problem? Thanks Carlo

1 0

[Serusers] fifo aliases working, serctl aliases not working
by Aisling 22 Aug '05

22 Aug '05

Hello, I need help troubleshooting a problem I am having adding aliases. It works fine using direct FIFO commands but not with my serctl script. I have not modified the serctl script so its just the standard script. Basically I am continually told that flags are expected. See below: 1) serctl alias add test1 sip:3456@serveraddress sip:3456@serveraddress 400 ul_add: flags expected 2) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 400 ul_add: expires expected 3) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 400 ul_add: q expected 4) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 1.00 400 ul_add: replicate expected 5) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 1.00 0 400 ul_add: flags expected 6) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 1.00 0 128 400 ul_add: flags expected 7) /opt/ser/sbin/serctl fifo ul_add aliases sip:3456@serveraddress 0 1.00 0 128 200 Added to table ('sip:3456', 'test1') to 'aliases' Any help appreciated, 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.

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[Serusers] problem with serctl command
by Richard Luo 22 Aug '05

22 Aug '05

Hi guys, i met the problem with using serctl to add the new users. I have the FC4 on my pc. I have uncommented the ser.cfg file for using mysql. and the error information is like follow: [cluo@pc5 ~]$ /usr/sbin/serctl add richard 000000 richard_luo(a)lycos.com /usr/sbin/serctl: line 764: gen_ha1: command not found HA1 calculation failed And if i changed to the root user, the error information is changed like these: /usr/sbin/serctl: line 401: mysql: command not found could anyone help me for this problems? best richard ============================ You won't admit you love me, and so How am I ever to know You always tell me... Perhaps, perhaps, perhaps -- _______________________________________________ Search for businesses by name, location, or phone number. -Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.as…

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[Serusers] configuration for public ser and natted pstn gateway
by Simone Cittadini 22 Aug '05

22 Aug '05

I've the following configuration : internet --- ser -||-- asterisk where ser has two network interfaces, one with public ip and one with 192.168.1.1, asterisk sits in the 192.168.1.0 net, authorization and forwarding of calls from internet works well with this ser.cfg (I omit parameters and obvious stuff): if (uri==myself) { if (method=="REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; lookup("aliases"); }; if (method=="INVITE") { record_route(); }; rewritehost("192.168.1.101"); <--- asterisk ip if (!t_relay()) { sl_reply_error(); }; but of course you can't hear nothing with this config... Googling I only find configurations relative to the ser machine also being natted, can someone tell me what I need to do in my particular case ?

4 3

[Serusers] accounting question
by Nhadie 22 Aug '05

22 Aug '05

Hi All, Is my query correct SELECT t1.from_uri as orig_number, t1.to_uri as term_number, TIMEDIFF(t2.time, t1.time) as duration, t1.timestamp as calldate FROM acc t1, acc t2 WHERE t1.sip_callid = t2.sip_callid AND ((t1.fromtag = t2.fromtag and t1.totag = t2.totag) OR (t1.fromtag = t2.totag and t1.totag = t2.fromtag)) AND t1.sip_method='INVITE' AND t2.sip_method='BYE' also this query displays all calls fron all user and it shows also all the previos month, what can I add here to show only calls from a certain user e.g 2009 and the current month? I tried this: SELECT t1.from_uri as orig_number, t1.to_uri as term_number, TIMEDIFF(t2.time, t1.time) as duration, t1.timestamp as calldate FROM acc t1, acc t2 WHERE t1.sip_callid = t2.sip_callid AND ((t1.fromtag = t2.fromtag and t1.totag = t2.totag) OR (t1.fromtag = t2.totag and t1.totag = t2.fromtag)) AND t1.sip_method='INVITE' AND t2.sip_method='BYE' AND t1.username="2009" and t1.time REGEXP '-08-'; <--- I added this line but i had an error, hope someone can help me. thank you. regards, nhadie

1 0

[Serusers] Module Avpops
by Hernando Maisano 22 Aug '05

22 Aug '05

Hello, I need to do, group of line, rotating lines. Somebody I advise myself that it uses I modulate AVPOPS. Somebody this using? Somebody can help me? I am reading http://www.voicesistem.ro/index.phplang=ro&page=tehnologii&sub_page=diferit… but it does not obtain it. Thanks Nando

2 4

[Users] OpenSER and SEMS ?
by Ozan Blotter 22 Aug '05

22 Aug '05

Dear List, Does anyone playing with SEMS 0.9.0 with OpenSER 0.9.X ? I couldn't find VM module to use with SEMS. Thanks, Ozan

3 2

[Users] Proper FROM mangling
by Bogdan-Andrei Iancu 22 Aug '05

22 Aug '05

Hi everybody, Based on the latest dialog supported (which was added in RR module), I was able to implement in the "uac" module proper FROM replacement. The missing issue was correct restoring/replacement of FROM in the sequential requests. As restoring information was kept in FROM param, this information didn't show up in all sequential requests (for example in ones from original caller to callee). Now, based on the new RR API, the module keeps the restoring info in the Route header which will show up for sure in all sequential request; so the module will be able to track and fix all sequential replies and requests. This implies fixing FROM or TO hdr, depending of the request direction. For people not interested in implementation details, I put first the usage description: There are 3 restore mode for the module to work (for FROM mangling): "none" -> NO RESTORE; the FROM URI will be changed and no further effort or info will be done/store in order to restore its value neither is sequential replies, nor requests; "manual" -> information needed for restoring the value will stored in RR parameter; all the replies and local generated requests (ACK and CANCEL) will be automatically fixed, but no sequential request will be restored; the admin has full liberty to do it from script by using the restore_from() function. "auto" -> FULL AUTO RESTORING -> information needed for restoring the value will stored in RR parameter and all sequential request and replies will be fixed *automatically* NOTE: for restore modes different than "none", the module will try to bind itself to RR and TM modules, so take care and load them. For people interested in implementation details. Shortly, here is how it works: Let's consider a call initiated from UA A to UA B (or from A to B) Now, since sequential requests may flow from A to B or B to A we need to be able both to restore original TO in requests from B to A and to replace with new FROM in request from A to B; which means that during the dialog we need to know both old and new FROM uris. As in each situation on of the value is known (depending of the direction, either the new, either the new URI appears), the simplest and most efficient way to store both in minimum size is to store the XOR value of them; having the XOR value and one of the uris, we can determine the other URI ( (x^y)^y=x :) ) A Proxy B 1) initial request: From:OLD_URI FROM:NEW_URI RR:(OLD_URI^NEW_URI) ---------------------------------> -----------------------------------> 2) final reply From:OLD_URI FROM:NEW_URI RR:(OLD_URI^NEW_URI) RR:(OLD_URI^NEW_URI) <--------------------------------- <----------------------------------- 3) re-invite (direction downstream) From:OLD_URI FROM:NEW_URI Route:(OLD_URI^NEW_URI) ---------------------------------> -----------------------------------> 4) final reply From:OLD_URI FROM:NEW_URI <--------------------------------- <----------------------------------- 5) BYE (direction upstream) TO:OLD_URI TO:NEW_URI Route:(OLD_URI^NEW_URI) <--------------------------------- <----------------------------------- 6) final reply TO:OLD_URI TO:NEW_URI ---------------------------------> -----------------------------------> 1) initial equest has the FROM URI changed from scrip via replace_from() ; XOR between NEW_URI and OLD_URI is added to RR as parameter 2) the reply (actualy all replies) is fixed via TM callbacks 3) the re-invite is detected via RR callback as having the RR param; as the direction is downstream,it will have the OLD_URI *FROM* changed with the NEW_URI = (OLD_URI^NEW_URI)^OLD_URI 2) the reply is fixed via TM callbacks 3) the BYE is detected via RR callback as having the RR param; as the direction is upstream,it will have the NEW_URI *TO* changed with the OLD_URI = (OLD_URI^NEW_URI)^NEW_URI and this is all the magic behind :) regards, bogdan

3 4

[Serusers] relaying fifo
by Iqbal 22 Aug '05

22 Aug '05

Hi I cant recall the name of the proggie to help relay ser_fifo from one machine to another, has anyone installed it, and does it work properly, I want to move fifo away from the central install of ser iqbal

1 0

[Users] SDP
by ggtel 22 Aug '05

22 Aug '05

Hello, Any use of SDP media attribute in conjunction with SIP /SER ? I would appreciate any insight! George ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] 2 SER network using 1 PSTN provider
by smy＠addm.com 22 Aug '05

22 Aug '05

Hi all... Just came across a problem with running 2 individual SER for 2 separate network. They both work fine independently, yet I have subscribed for PSTN landing of calls for only 1 SER server (IP restrictions). Now, for my second SER server, I have forwarded calls to the first server (much like pointing a call to it using "if uri=~" command........ and of course, in the trusted table of SER 1, I have placed the IP of the SER 2. Now... the call goes through and the can be landed, yet the PSTN user can not hear anything while the IP Phone user can hear perfectly from the PSTN user. The 2 SER servers are setup based on onsip.org with mediaproxy configuration. Everything works fine, except for this 1 way audio problem. Could some please point me to the right direction to where to solve this?!?! By the way, if my IP phone were to use a REAL IP on SER 2 and try to land calls via SER 1, everything works perfectly!!! There will be 2 way audio. So I would imagine this has something to do with mediaproxy or NAT issues. PLEASE HELP!! :>

2 1

[Serusers] ser configuration issues
by AChin63457＠aol.com 22 Aug '05

22 Aug '05

I just started installing ser the other day and the installation was smooth and painless. I am know not quite sure about about the configuration files and how to set them. Here is my current setup -- I have a fedora 3 linux box obtaining a dhcp assigned address with currently no hostname set and no FQDN-- localhost.localdomain. I signed up the other day to iptel.org to register for a username and password -- ie, _example(a)iptel.org_ (mailto:example@iptel.org) . So i have 2 issues. 1. What should i specify in my /etc/profile for SIP_DOMAIN -- should i use my _example(a)iptel.org_ (mailto:example@iptel.org) or my internal lan address or my wan address available to the outside? 2. Same question when it comes to configuring the /usr/local/etc/ser/ser.cfg what should be listed on the listen parameters listen=127.0.0.1 listen=192.168.x.x or should it be the wan address also can i use the alias alias=_example(a)iptel.org_ (mailto:example@iptel.org) or should this be my machines hostname. Regards, Alfred

2 1

[Users] Re: Re: Re: Re: MediaProxy & PSTN GW
by Ozan Blotter 22 Aug '05

22 Aug '05

Dear Klaus, As i get ideas from three people here i should close this subject. Codec i use is g729 (it supports g711 too) as it comes default with Zyxel Prestige 2002, even i have a special firmware (engineered one) for this product so it supports g723 too :P, i think problem is high latencies in ADSL or CableNet services here, even they're 512/128, 1024/256, 2048/512 in ADSL and 512/256, 1024/512, 2048/1024 in CableNet... Anyway i will try to find another solution and thanks for your replies... Ozan... ------------------------------ Message: 3 Date: Sat, 20 Aug 2005 23:03:59 +0200 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> Subject: Re: [Users] Re: Re: MediaProxy & PSTN GW To: cosmocid(a)ispro.net.tr Cc: users(a)openser.org Message-ID: <43079ABF.9030405(a)pernau.at> Content-Type: text/plain; charset=us-ascii; format=flowed That does not help you - inserting another hop can not compensate the bottleneck between the UAs and the Internet (the ADSL connection). BTW: 600km is not far away. Which Codec you you use? Try low-babdwidth codecs like G729, ilbc, GSM and G723 klaus Ozan Blotter wrote: > Dear Iqbal and Bogdan, > > That is exactly what i need to improve QoS, because the UAs ADSL > connection is not fast enough, 512/128k in general, OpenSER and UA's are > located in same city but PSTN GW is nearly 600 km (350 miles nearly) away > from this point. I thought since OpenSER is on a good connection (1Mbit > Fiber) and can relay to PSTN faster then UA's themselves. > > Thanks, > Ozan > > Also in terms of ping times how far is the PSTN gateway, as Bogdan said, > media proxy will add even more delay, as for sending pstn via > mediaproxy, you could do all yourprocessing, and once ser detects that > call must goto PSTN, then do all the NAT checking ...this is a > oversimplified view... > > Iqbal > > Bogdan-Andrei Iancu wrote: > > >>Hi Ozan, >> >>so, if I get it right, you have a PSTN gw locate far away from the >>UAs...and instead of letting the media traffic to do directly between >>GW and UAs, you want to insert in the middle a RTP relay (mediaproxy) >>in order to improve the QoS, right? >>If so, I would say this will not help you - having a media relay on >>your media path leads to QoS degradation due additional delays.... >> >>If you still want to go for it....I will take a look on tour script. >> >>regards, >>bogdan >> >>Ozan Blotter wrote: >> >> >>>Dear List, >>> >>>At below config. i setted up OpenSER on Public IP and PSTN Gateway is >>>again on another Public IP. I use Zyxel Prestige Series 660-HW ADSL >>>modems on both client side and note that they have SIP ALG, also i've >>>completed port forwarding in each modem, so SIP2SIP call happens >>>(voice traffic goes from UA to UA) without need of anything extra as >>>STUN or RTP RELAY. But since those are very away from each other if a >>>UA needs to call PSTN the voice traffic flows from UA to PSTN and >>>that lowers QoS :( >>> >>>I would like to activate MediaProxy again but just for one case which >>>is PSTN, so whenever a UA needs to call PSTN the voice traffic flows >>>like : >>> >>>UA (ATA) >>> OpenSER + MediaProxy >>> PSTN >>> >>>How can i setup this like above without touching SIP2SIP calls >>>between UA's... >>>or adding the lines beginning with # these to my existing config is >>>enough ? >>> >>>Thanks, >>>Ozan Blotter

1 0

[Users] Re: Re: MediaProxy & PSTN GW
by Ozan Blotter 22 Aug '05

22 Aug '05

Dear Iqbal and Bogdan, That is exactly what i need to improve QoS, because the UAs ADSL connection is not fast enough, 512/128k in general, OpenSER and UA's are located in same city but PSTN GW is nearly 600 km (350 miles nearly) away from this point. I thought since OpenSER is on a good connection (1Mbit Fiber) and can relay to PSTN faster then UA's themselves. Thanks, Ozan Also in terms of ping times how far is the PSTN gateway, as Bogdan said, media proxy will add even more delay, as for sending pstn via mediaproxy, you could do all yourprocessing, and once ser detects that call must goto PSTN, then do all the NAT checking ...this is a oversimplified view... Iqbal Bogdan-Andrei Iancu wrote: > Hi Ozan, > > so, if I get it right, you have a PSTN gw locate far away from the > UAs...and instead of letting the media traffic to do directly between > GW and UAs, you want to insert in the middle a RTP relay (mediaproxy) > in order to improve the QoS, right? > If so, I would say this will not help you - having a media relay on > your media path leads to QoS degradation due additional delays.... > > If you still want to go for it....I will take a look on tour script. > > regards, > bogdan > > Ozan Blotter wrote: > >> Dear List, >> >> At below config. i setted up OpenSER on Public IP and PSTN Gateway is >> again on another Public IP. I use Zyxel Prestige Series 660-HW ADSL >> modems on both client side and note that they have SIP ALG, also i've >> completed port forwarding in each modem, so SIP2SIP call happens >> (voice traffic goes from UA to UA) without need of anything extra as >> STUN or RTP RELAY. But since those are very away from each other if a >> UA needs to call PSTN the voice traffic flows from UA to PSTN and >> that lowers QoS :( >> >> I would like to activate MediaProxy again but just for one case which >> is PSTN, so whenever a UA needs to call PSTN the voice traffic flows >> like : >> >> UA (ATA) >>> OpenSER + MediaProxy >>> PSTN >> >> How can i setup this like above without touching SIP2SIP calls >> between UA's... >> or adding the lines beginning with # these to my existing config is >> enough ? >> >> Thanks, >> Ozan Blotter > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > . >

3 2

Re: [Users] SDP
by Bogdan-Andrei Iancu 22 Aug '05

22 Aug '05

Hi George, OpenSER is a signalling SIP proxy and it has very few implications over the SDP (media) part - only regarding NAT traversal. the SDP info is mostly used end-to-end (only by user-agents) regards, Bogdan ggtel wrote: >Hello, > >Any use of SDP media attribute in conjunction with SIP /SER ? >I would appreciate any insight! > >George > > > > > > > >___________________________________________________________________________ >Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger >Téléchargez cette version sur http://fr.messenger.yahoo.com > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

1 0

Re:[Serusers] Centralized routing for managing multiple SER proxies
by Samuel Osorio Calvo 22 Aug '05

22 Aug '05

Take a look at RFC3263 Locating SIP Servers.The idea is to use DNS to locate the right server. If you configure your DNS to serve your different SER proxies you just have to use t_relay and the request will be forwarded to the right destination. Hope it helps. Samuel Unclassified. >>> <vmathur(a)transnexus.com> 08/19/05 06:20PM >>> Hi Ashutosh, Thanks for your response. The readme file of mediaproxy talks about load balancing the RTP streams across multiple proxies. Does the signaling also follow the same path as the media (for the same call). Also, The readme file of the mediaproxy says that: "Once it has determined which domain to use to obtain the list of available proxy servers, it will make a DNS lookup for the following SRV records: _mediaproxy._tcp.selected.domain.name" Does this mean that the DNS lookup serves as the routing mechanism for the call? Also, the document says: "However it should be fairly easy to implement another dispatcher that uses a different mean to determine the proxy server (like using a database mapping), or even to distribute the traffic by other means than SIP domain name." I guess that this means that I could use a 3rd party routing engine to route the calls. Is that correct? If so, is there an interface in the dispatcher that I could use to make the dispatcher talk to my routing DB and not do a DNS lookup. Thanks, Vikrant

1 0

Re: [Users] Bus error
by Samuel Osorio Calvo 22 Aug '05

22 Aug '05

Try to compile SER without any optimization parameter set. Bus error usually refers to a bad alignemnt and usually the optimization paramters of the compiler sometimes do not properly behave with the alignemnt. Hope it helps. Samuel. Unclassified. >>> Marc Haisenko <haisenko(a)be-ok.com> 08/17/05 06:07PM >>> On Tuesday 16 August 2005 22:53, Andréa Giordanna wrote: > Hello, does anybody here already have this problem > while starting openser? > And, what does this Aliases exactly means? > > [...] > > Bus error > > Thanks, > Andrea Giordanna "Bus error" is a very critical OS error message, it has nothing to do with OpenSER. You either have faulty hardware or hit a kernel bug ;-) C'ya, Marc -- Marc Haisenko Linux Solutions Be O.K. service group GmbH Rüdesheimer Straße 7 D-80686 München Tel: +49 (0)89 - 548 43 33 21 Fax: +49 (0)89 - 548 43 33 29 e-mail: haisenko(a)be-ok.com http://www.be-ok.com _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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RE: [Serusers] Serweb
by Marc Khayat 22 Aug '05

22 Aug '05

Anybody got the Log.php file from serweb (ser-0.9.3), coz it seems like it's missing from my package. Thx, Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marc Khayat Sent: Friday, August 19, 2005 11:25 AM To: Karel Kozlik Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Serweb /tmp/ser_fifo/ is wrong since it'll be looking for a folder... Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marc Khayat Sent: Friday, August 19, 2005 11:03 AM To: Karel Kozlik Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Serweb Hi Karel, Thx for bearing with me :) In config_paths.php, I changed it from "/tmp/ser_fifo" to "tmp/ser_fifo/" The errors at the top disappeared but I'm still getting the "FIFO not running or bad path to it" message. Is it wrong to put "tmp/ser_fifo/"? knowing that in ser.cfg, it's /tmp/ser_fifo Thanks again, Marc -----Original Message----- From: Karel Kozlik [mailto:karel@iptel.org] Sent: Friday, August 19, 2005 9:28 AM To: Marc Khayat Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Serweb sorry Marc, haven't idea what may be wrong. Please check again setting of fifo in serweb. Check variable $config->fifo_server in config_paths.php. It's value must be '/tmp/ser_fifo'. Try enable logging in serweb ($config->enable_loging = true in config.php), set log level to debug ($config->log_level = "PEAR_LOG_DEBUG") and examine the log file if there are some messages refered fifo. Also try search for string 'fifo_server' throught log files (or whole serweb) and check if it's value isn't changed somewhere. You should find this string only in file config_paths.php (one times) and in file functions.php (four times). Karel Marc Khayat napsal(a): > Hi Karel, > I already tried this one by manually changing the permissions on > ser_fifo to 666; I just tried adding fifo_mode=0666 to ser.cfg, but > still the same problem. > > Marc > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: Friday, August 19, 2005 8:21 AM > To: Marc Khayat > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Serweb > > Hi, > try add: > fifo_mode=0666 > > to your ser.cfg > > Karel > > Marc Khayat napsal(a): > >>/tmp/ser_fifo exists, and is pipe type >>User running Apache is "apache". Owner of ser_fifo is User: "root", >>Group: "root". >>I joined user "apache" to group "root", same. >> >>I created group "ser" and joined "root" "apache" to that group, and >>added fifo_group=ser to the ser.cfg, but still same problem: >> >>Warning: filetype(): Lstat failed for (null) (errno=13 - Permission >>denied) in /var/www/html/serweb/html/functions.php on line 306 >> >>Warning: Cannot modify header information - headers already sent by >>(output started at /var/www/html/serweb/html/functions.php:306) in >>/var/www/html/serweb/html/page.php on line 35 >> >> >>Thanks, >> >>Marc >> >>-----Original Message----- >>From: Karel Kozlik [mailto:karel@iptel.org] >>Sent: Thursday, August 18, 2005 1:38 PM >>To: Marc Khayat >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Serweb >> >>Hello, >>please check if file /tmp/ser_fifo exists and if it's type is pipe. > > Also > >>check rights - if user under which apache is running have access to > > read > >>and write to this file. >> >>Karel >> >>Marc Khayat napsal(a): >> >> >>>The domain thing is working, but problem 2 is still bugging me. >>>Help anyone? >>> >>>Marc >>> >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >> >>On >> >> >>>Behalf Of Marc Khayat >>>Sent: Thursday, August 18, 2005 11:15 AM >>>To: serusers(a)lists.iptel.org >>>Subject: RE: [Serusers] Serweb >>> >>>Never mind guys... it's working! >>> >>>Marc >>>-----Original Message----- >>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >> >>On >> >> >>>Behalf Of Marc Khayat >>>Sent: Thursday, August 18, 2005 10:53 AM >>>To: serusers(a)lists.iptel.org >>>Subject: [Serusers] Serweb >>> >>>Hi all, >>>I'm using SER-0.9.3 and its serweb version. >>>I have 2 problems with it: >>>1- On the login page, when I enter the username, it automatically >>>completes it with the FQDN, while I need only the domain name; i.e. I >>>login to http://ser.xyz.com, I enter the username aloha, it's >> >>completed >> >> >>>as aloha(a)ser.xyz.com while I need it as aloha(a)xyz.com. Currently, I'm >>>writing in manually >>> >>>2- When I login, I'm getting an error on the first page only as "FIFO >>>not running or bad path to it". The path to it (in config_paths.php >> >>file >> >> >>>is /tmp/ser_fifo, which is correct), and when I enter the command >> >>serctl >> >> >>>monitor, the counter isn't constant, and same path. >>> >>>Can anyone help me plz, >>> >>>Thanks, >>> >>>Marc >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> > > _______________________________________________ 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

2 1

[Serusers] Error during accounting - val2str()
by Carlo 22 Aug '05

22 Aug '05

1 0

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