sr-users July 2004

sr-users@lists.kamailio.org

178 participants
449 discussions

[Serusers] ERROR when trying to start ser

by Francesco Bottà

Hi all, I've just updated my ser via cvs co sip_router. But I'm getting this error, after make all and make install, when i'm trying to start ser: 0(0) fixing /usr/local/lib/ser/modules/auth.so consume_credentials 0(0) fixing /usr/local/lib/ser/modules/auth.so www_challenge 0(0) fixing /usr/local/lib/ser/modules/uri.so is_user 0(0) fixing /usr/local/lib/ser/modules/uri_db.so check_to 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/registrar.so save 0(0) get_connection(): Connection not found in the pool 0(0) submit_query(): Unknown column 'user_agent' in 'field list' 0(0) submit_query(): Error while submitting query 0(0) preload_udomain(): Error while doing db_query 0(0) register_udomain(): Error while preloading domain 'location' 0(0) release_connection(): Removing connection from the pool 0(0) domain_fixup(): Error while registering domain 0(0) ERROR: fix_expr : fix_actions error ERROR: error ffffffff while trying to fix configuration 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : 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 : destroing tmcb lists 0(0) DEBUG: tm_shutdown : done 0(18452) shm_mem_destroy 0(18452) destroying the shared memory lock What's wrong with my config? Please, help me. Best Regards, Franz

20 years, 3 months

[Serusers] No answer call forwarding does not work

by David Lee

Hello, In our lab environment, we have the SER server (10.0.0.1), 1 Cisco IP phone (#81001, 10.0.0.200), 2 XTEN Pro Softphones (#84000 10.0.0.4 & #85000 10.0.0.5), and voicemail server SEMS+SER (10.0.0.150) on a separate box. The softphone #84000 is setup to do a no-answer forward to softphone #85000. When a call originates from the IP Phone, a 302 Redirecting message is sent from the softphone (10.0.0.4) to the SER (10.0.0.1). However, the SER does not relay it to the originating IP Phone (10.0.0.200)... The SER sends an INVITE message to the voicemail (10.0.0.150) when receiving the 302 message... We don't know why that is happening! Attached is the ngrep captured at the SER, and the ser.cfg file. Can anyone please help us? Thanks Dave

20 years, 3 months

Re: [Serusers] Call Failed: 404 Not Found

by Bogdan-Andrei IANCU

looks like your callee is not in usr location: 2(3358) lookup(): 'dani' Not found in usrloc please check with "serctl ul show" the content of user location mem cache and see if that user is really registered or not. bogdan Pavel Siderov wrote: > 2(3358) SIP Request: > 2(3358) method: <INVITE> > 2(3358) uri: <sip:dani@mydomain.org> > 2(3358) version: <SIP/2.0> > 2(3358) parse_headers: flags=1 > 2(3358) Found param type 235, <rport> = <n/a>; state=6 > 2(3358) Found param type 232, <branch> = ><z9hG4bK6E32A94E8FC742AFAC077B24E64C1319>; state=16 > 2(3358) end of header reached, state=5 > 2(3358) parse_headers: Via found, flags=1 > 2(3358) parse_headers: this is the first via > 2(3358) After parse_msg... > 2(3358) preparing to run routing scripts... > 2(3358) DEBUG : is_maxfwd_present: searching for max_forwards header > 2(3358) parse_headers: flags=128 > 2(3358) end of header reached, state=9 > 2(3358) DEBUG: get_hdr_field: <To> [26]; uri=[sip:dani@mydomain.org] > 2(3358) DEBUG: to body [<sip:dani@mydomain.org> >] > 2(3358) get_hdr_field: cseq <CSeq>: <880> <INVITE> > 2(3358) DEBUG: is_maxfwd_present: value = 70 > 2(3358) DEBUG: add_param: tag=4155995681 > 2(3358) end of header reached, state=29 > 2(3358) parse_headers: flags=256 > 2(3358) DEBUG: get_hdr_body : content_length=308 > 2(3358) found end of header > 2(3358) find_first_route(): No Route headers found > 2(3358) loose_route(): There is no Route HF > 2(3358) check_self - checking if host==us: 13==13 && [mydomain.org] == >[mydomain.org] > 2(3358) check_self - checking if port 5060 matches port 5060 > 2(3358) lookup(): 'dani' Not found in usrloc > 2(3358) parse_headers: flags=-1 > 2(3358) check_via_address(192.168.51.2, 192.168.51.2, 3) > 2(3358) receive_msg: cleaning up > 5(3363) SIP Request: > 5(3363) method: <ACK> > 5(3363) uri: <sip:dani@mydomain.org> > 5(3363) version: <SIP/2.0> > 5(3363) parse_headers: flags=1 > 5(3363) Found param type 235, <rport> = <n/a>; state=6 > 5(3363) Found param type 232, <branch> = ><z9hG4bK6E32A94E8FC742AFAC077B24E64C1319>; state=16 > 5(3363) end of header reached, state=5 > 5(3363) parse_headers: Via found, flags=1 > 5(3363) parse_headers: this is the first via > 5(3363) After parse_msg... > 5(3363) parse_headers: flags=4 > 5(3363) DEBUG: add_param: tag=99d196a8a45a6ad6a584303c392bdd59.db43 > 5(3363) end of header reached, state=29 > 5(3363) DEBUG: get_hdr_field: <To> [68]; uri=[sip:dani@mydomain.org] > 5(3363) DEBUG: to body [<sip:dani@mydomain.org>] > 5(3363) DEBUG: sl_filter_ACK : local ACK found -> dropping it! > 5(3363) receive_msg: cleaning up > >----- Original Message ----- >From: "Bogdan-Andrei IANCU" <iancu(a)fokus.fraunhofer.de> >To: "Pavel Siderov" <pi(a)hostmates.com> >Sent: Tuesday, July 27, 2004 5:07 PM >Subject: Re: [Serusers] Call Failed: 404 Not Found > > > > >>full log of ser when processing the INVITE will be helpful. ser debug=9, >>fork=no, log_stderr=yes in your script and make the call. >>bogdan >> >>Pavel Siderov wrote: >> >> >> >>>I tried this but the same problem exists :( >>>----- Original Message ----- >>>From: "Bogdan-Andrei IANCU" <iancu(a)fokus.fraunhofer.de> >>>To: "Pavel Siderov" <pi(a)hostmates.com> >>>Cc: <serusers(a)lists.iptel.org> >>>Sent: Tuesday, July 27, 2004 3:49 PM >>>Subject: Re: [Serusers] Call Failed: 404 Not Found >>> >>> >>> >>> >>> >>> >>>>try to set "db_mode" to 0 - maybe you have some mysql connection error - >>>>and see if it works. >>>>also, listen=mydomain.org, shouldn't be actually alias=mydomain.org? >>>>bogdan >>>> >>>>Pavel Siderov wrote: >>>> >>>> >>>> >>>> >>>> >>>>>Hello, >>>>>I read moany posts but cannot resolve my problem with ser... >>>>>here it is (i've changed my real ip and domain name) >>>>> >>>>>slackware 10 (actually it is current) >>>>>ser-08.12 >>>>>my ser.cfg >>>>> >>>>># $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ >>>>>check_via=no >>>>>dns=yes >>>>>rev_dns=yes >>>>>port=5060 >>>>>fifo="/tmp/ser_fifo" >>>>> >>>>># ------------------ module loading ---------------------------------- >>>>>loadmodule "/usr/local/lib/ser/modules/mysql.so" >>>>>loadmodule "/usr/local/lib/ser/modules/sl.so" >>>>>loadmodule "/usr/local/lib/ser/modules/tm.so" >>>>>loadmodule "/usr/local/lib/ser/modules/rr.so" >>>>>loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >>>>>loadmodule "/usr/local/lib/ser/modules/usrloc.so" >>>>>loadmodule "/usr/local/lib/ser/modules/registrar.so" >>>>>loadmodule "/usr/local/lib/ser/modules/auth.so" >>>>>loadmodule "/usr/local/lib/ser/modules/auth_db.so" >>>>> >>>>># -- usrloc params -- >>>>>modparam("usrloc", "db_mode", 2) >>>>>modparam("auth_db", "calculate_ha1", yes) >>>>>modparam("auth_db", "password_column", "password") >>>>>modparam("rr", "enable_full_lr", 1) >>>>>listen=mydomain.org >>>>>listen=213.213.213.213 >>>>> >>>>># ------------------------- request routing logic ------------------- >>>>> >>>>>route{ >>>>>if (!mf_process_maxfwd_header("10")) { >>>>>sl_send_reply("483","Too Many Hops"); >>>>>break; >>>>>}; >>>>>if ( msg:len > max_len ) { >>>>>sl_send_reply("513", "Message too big"); >>>>>break; >>>>>}; >>>>>record_route(); >>>>>if (loose_route()) { >>>>>t_relay(); >>>>>break; >>>>>}; >>>>>if (uri==myself) { >>>>>if (method=="REGISTER") { >>>>># if (!www_authorize("mydomain.org", "subscriber")) { >>>>># www_challenge("mydomain.org", "0"); >>>>># break; >>>>># }; >>>>>save("location"); >>>>>break; >>>>>}; >>>>>if (!lookup("location")) { >>>>>sl_send_reply("404", "Not Found"); >>>>>break; >>>>>}; >>>>>}; >>>>>if (!t_relay()) { >>>>>sl_reply_error(); >>>>>}; >>>>>} >>>>> >>>>> >>>>> >>>>>Everything seems to be ok and no error messages in >>>>> >>>>>/var/log/syslog >>>>> >>>>>/var/log/messages >>>>>Jul 27 14:10:39 sf /usr/local/sbin/ser[xxx]: INFO: signal 15 received >>>>> >>>>>Something in /var/lib/mysql/myhost.err >>>>>040727 14:07:01 Aborted connection 1480 to db: 'ser' user: 'ser' host: >>>>>`localhost' (Got an error reading communication packets) >>>>>040727 14:07:01 Aborted connection 1485 to db: 'ser' user: 'serro' >>>>>host: `localhost' (Got an error reading communication packets) >>>>> >>>>>But messages go only when I stop ser. >>>>>SO the problem is that - I use X-Lite sip phone, I login and >>>>>everything is ok but when I try to call another user >>>>>typing user2(a)mydomain.org <mailto:user2@mydomain.org> I got message >>>>>Call Failed: 404 Not Found. User2 is logged in too. Here I use user >>>>> >>>>> >>>>>from internal >>>> >>>> >>>>>network but whatever user log in even with real ip ... same problem. >>>>>serctl moni: >>>>>... >>>>>UsrLoc Stats >>>>>Domain Registered Expired >>>>>'location' 2 0 >>>>>serctl ul show: >>>>>q : 0.00 >>>>>Call-ID: '83F62A146F1849D38374C4FE587C96E7(a)mydomain.org' >>>>><mailto:%2783F62A146F1849D38374C4FE587C96E7@mydomain.org%27> >>>>>CSeq : 62404 >>>>>replic : 0 >>>>>State : CS_SYNC >>>>>Flags : 0 >>>>>next : (nil) >>>>>prev : (nil) >>>>>~~~/Contact~~~~ >>>>>.../Record... >>>>>...Record(0x402e4aa0)... >>>>>domain: 'location' >>>>>aor : 'dani(a)mydomain.org' <mailto:%27dani@mydomain.org%27> >>>>> >>>>>~~~Contact(0x402e4af0)~~~ >>>>> >>>>>domain : 'location' >>>>>aor : 'dani(a)mydomain.org' <mailto:%27dani@mydomain.org%27> >>>>>Contact: 'sip:dani%40mydomain.org@192.168.1.2:5060' >>>>>Expires: 1714 >>>>>q : 0.00 >>>>>Call-ID: '919B7F440210436E848ED6EE743C472C(a)mydomain.org' >>>>><mailto:%27919B7F440210436E848ED6EE743C472C@mydomain.org%27> >>>>>CSeq : 40853 >>>>>replic : 0 >>>>>State : CS_SYNC >>>>>Flags : 0 >>>>>next : (nil) >>>>>prev : (nil) >>>>>~~~/Contact~~~~ >>>>>.../Record... >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>X-lite diagnostic log: >>>>> >>>>>v=0 >>>>>o=petko(a)mydomain.org <mailto:o=petko@mydomain.org> 16984234 16984250 >>>>>IN IP4 192.168.51.2 >>>>>s=X-Lite >>>>>c=IN IP4 192.168.51.2 >>>>>t=0 0 >>>>>m=audio 8000 RTP/AVP 0 8 3 98 97 101 >>>>>a=rtpmap:0 pcmu/8000 >>>>>a=rtpmap:8 pcma/8000 >>>>>a=rtpmap:3 gsm/8000 >>>>>a=rtpmap:98 iLBC/8000 >>>>>a=rtpmap:97 speex/8000 >>>>>a=rtpmap:101 telephone-event/8000 >>>>>a=fmtp:101 0-15 >>>>>SEND TIME: 16890921 >>>>> >>>>>SEND >> 213.213.213.213:5060 >>>>>ACK sip:dani@mydomain.org SIP/2.0 >>>>>Via: SIP/2.0/UDP >>>>>192.168.51.2:5060;rport;branch=z9hG4bK2C881ABD05574ECFB9ECA868ED909189 >>>>>From: Pavel <sip:petko%40mydomain.org@mydomain.org>;tag=251597520 >>>>>To: <sip:dani@mydomain.org>;tag=99d196a8a45a6ad6a584303c392bdd59.a444 >>>>>Contact: <sip:petko%40mydomain.org@192.168.51.2:5060> >>>>>Call-ID: D8AAB24B-799F-4485-B18A-A52C996E2BFA(a)192.168.51.2 >>>>><mailto:D8AAB24B-799F-4485-B18A-A52C996E2BFA@192.168.51.2> >>>>>CSeq: 25255 ACK >>>>>Max-Forwards: 70 >>>>>Content-Length: 0 >>>>> >>>>> >>>>>SEND TIME: 16984250 >>>>>SEND >> 213.213.213.213:5060 >>>>>INVITE sip:dani@mydomain.org SIP/2.0 >>>>>Via: SIP/2.0/UDP >>>>>192.168.51.2:5060;rport;branch=z9hG4bKE1A01C567E6E4C4D8E97F16D05498E79 >>>>>From: Pavel <sip:petko%40mydomain.org@mydomain.org>;tag=1337044785 >>>>>To: <sip:dani@mydomain.org> >>>>>Contact: <sip:petko%40mydomain.org@192.168.51.2:5060> >>>>>Call-ID: 3FC4393F-8B1A-49F2-BF30-9426F6C1F1AE(a)192.168.51.2 >>>>><mailto:3FC4393F-8B1A-49F2-BF30-9426F6C1F1AE@192.168.51.2> >>>>>CSeq: 9818 INVITE >>>>>Max-Forwards: 70 >>>>>Content-Type: application/sdp >>>>>User-Agent: X-Lite release 1103m >>>>>Content-Length: 308 >>>>> >>>>> >>>>>Please help me! >>>>>Thanks in advance >>>>> >>>>> >>>>> >>>>------------------------------------------------------------------------ >>>> >>>> >>>>>_______________________________________________ >>>>>Serusers mailing list >>>>>serusers(a)lists.iptel.org >>>>>http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>__________ NOD32 1.578 (20031212) Information __________ >>>> >>>>This message was checked by NOD32 antivirus system. >>>>http://www.nod32.com >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> >>> >> >>__________ NOD32 1.578 (20031212) Information __________ >> >>This message was checked by NOD32 antivirus system. >>http://www.nod32.com >> >> >> >> > > > >

20 years, 3 months

[Serusers] Asterisks to ser to asterisk (voicemail)

by jon＠bostontech.com

I'm running into a little problem with forwarding. This is what's happening: I have a PBX connected into an asterisk box, allowing the PBX users to make SIP calls from their regular phones. I'm also running a SER box with my own sip domain. I have the SER box send unavailable calls to voicemail on the Asterisk server. Now, when someone uses the PBX phone to call a sip user from my sip domain and they are unavailable, i'm getting a "loop detected" from ser. PBX -> Asterisk -> Ser -> Asterisk I can see why i am getting this (since from ser's perspective it's doing a loop), even though once Ser forwarded the call to asterisk, it won't receive it again, since the incoming call is in a different context. My question is, is there any way to have ser receive a call from asterisk and then reroute it back to the same asterisk server without giving a "loop detected" error? Thanks -Jon

20 years, 3 months

[Serusers] Call Failed: 404 Not Found

by Pavel Siderov

Hello, I read moany posts but cannot resolve my problem with ser... here it is (i've changed my real ip and domain name) slackware 10 (actually it is current) ser-08.12 my ser.cfg # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ check_via=no dns=yes rev_dns=yes port=5060 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) listen=mydomain.org listen=213.213.213.213 # ------------------------- request routing logic ------------------- route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { # if (!www_authorize("mydomain.org", "subscriber")) { # www_challenge("mydomain.org", "0"); # break; # }; save("location"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (!t_relay()) { sl_reply_error(); }; } Everything seems to be ok and no error messages in /var/log/syslog /var/log/messages Jul 27 14:10:39 sf /usr/local/sbin/ser[xxx]: INFO: signal 15 received Something in /var/lib/mysql/myhost.err 040727 14:07:01 Aborted connection 1480 to db: 'ser' user: 'ser' host: `localhost' (Got an error reading communication packets) 040727 14:07:01 Aborted connection 1485 to db: 'ser' user: 'serro' host: `localhost' (Got an error reading communication packets) But messages go only when I stop ser. SO the problem is that - I use X-Lite sip phone, I login and everything is ok but when I try to call another user typing user2(a)mydomain.org I got message Call Failed: 404 Not Found. User2 is logged in too. Here I use user from internal network but whatever user log in even with real ip ... same problem. serctl moni: ... UsrLoc Stats Domain Registered Expired 'location' 2 0 serctl ul show: q : 0.00 Call-ID: '83F62A146F1849D38374C4FE587C96E7(a)mydomain.org' CSeq : 62404 replic : 0 State : CS_SYNC Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0x402e4aa0)... domain: 'location' aor : 'dani(a)mydomain.org' ~~~Contact(0x402e4af0)~~~ domain : 'location' aor : 'dani(a)mydomain.org' Contact: 'sip:dani%40mydomain.org@192.168.1.2:5060' Expires: 1714 q : 0.00 Call-ID: '919B7F440210436E848ED6EE743C472C(a)mydomain.org' CSeq : 40853 replic : 0 State : CS_SYNC Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... X-lite diagnostic log: v=0 o=petko(a)mydomain.org 16984234 16984250 IN IP4 192.168.51.2 s=X-Lite c=IN IP4 192.168.51.2 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 SEND TIME: 16890921 SEND >> 213.213.213.213:5060 ACK sip:dani@mydomain.org SIP/2.0 Via: SIP/2.0/UDP 192.168.51.2:5060;rport;branch=z9hG4bK2C881ABD05574ECFB9ECA868ED909189 From: Pavel <sip:petko%40mydomain.org@mydomain.org>;tag=251597520 To: <sip:dani@mydomain.org>;tag=99d196a8a45a6ad6a584303c392bdd59.a444 Contact: <sip:petko%40mydomain.org@192.168.51.2:5060> Call-ID: D8AAB24B-799F-4485-B18A-A52C996E2BFA(a)192.168.51.2 CSeq: 25255 ACK Max-Forwards: 70 Content-Length: 0 SEND TIME: 16984250 SEND >> 213.213.213.213:5060 INVITE sip:dani@mydomain.org SIP/2.0 Via: SIP/2.0/UDP 192.168.51.2:5060;rport;branch=z9hG4bKE1A01C567E6E4C4D8E97F16D05498E79 From: Pavel <sip:petko%40mydomain.org@mydomain.org>;tag=1337044785 To: <sip:dani@mydomain.org> Contact: <sip:petko%40mydomain.org@192.168.51.2:5060> Call-ID: 3FC4393F-8B1A-49F2-BF30-9426F6C1F1AE(a)192.168.51.2 CSeq: 9818 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 308 Please help me! Thanks in advance

20 years, 3 months

[Serusers] alternative for cpl-c No Answer timer

by Arne Scheffer

Bogdan and other who are interested, What would you advice at present if I want to use SER and offer a forward on No Answer feature ? As TM and thus CPL-C does not support the timeout (timer) option I understand it can not be used to implentent this feature. Other options : use SER -> asterisk (FS) In this case all calls are sent to the FS maybe some smart thing can be done to only send users with a no answer script. If I want to enable PSTN forwarding + Billing I think I would then end up with the following: SER -> Asterisk (FS) -> SER -> Asterisk (PSTN) -> PSTN GW Is this the only way ? Is there something better ? any ideas are welcome. regards, Arne.

20 years, 3 months

[Serusers] server monitoring

by Sanjiv Ranjan

Hi, I am not getting any value in/ admin/server monitoring page. I have already enabled corn_job as said in installation/ and given command line entr of serweb cron jobs like --php -q /usr/local/apache/htdocs/serweb/html/admin/cron_job/read_ser_moni.php | cat >> read_ser_moni.log but, still not getting value in /admin/server monitoring page./ Any help. Thanks in advance. sanjiv

20 years, 3 months

RE: [Serusers] problem with serctl add

by Arne Scheffer

Jan, Adding lookup("aliases") to the config has helped. Thanks, Arne. -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: dinsdag 27 juli 2004 11:36 To: Arne Scheffer Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] problem with serctl add I think that the problem is that you do not have lookup("aliases") in the configuration file. SER does not load the aliases table into memory and the check in serctl fails. That deserves to be fixed. Jan. On 27-07 10:56, Arne Scheffer wrote: > Jan, > > The user DB's (subscriber and alias) are both empty (exept for admin account). > > Please find attached config file. > > Could it be I am having a problem because the current config does not user authentication yet ? > > This way I can still do call tests until I have the authentication working. > > Arne. > > -----Original Message----- > From: Jan Janak [mailto:jan@iptel.org] > Sent: maandag 26 juli 2004 19:34 > To: Arne Scheffer > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] problem with serctl add > > > Do you have lookup("aliases") in your configuration file ? Serweb tries > to check if the username does not collide with existing aliases. > > Jan. > > On 26-07 12:09, Arne Scheffer wrote: > > Hello all, > > > > I am using latest CVS head (friday 23) of ser and serweb. > > > > I am having problems with adding users using the serctl > > > > export SIP_DOMAIN=212.x.x.x > > serctl add 200 test test(a)nobody.net > > > > I always get the following error: > > > > error: overlap with an existing alias > > > > I have tried alle types of names als with someone(a)212.x.x.x > > > > Any ideas ? > > > > I also ran into a problem trying to add users with serweb (see email on 400 ul_add) > > > > thanks, > > Arne. > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers

20 years, 3 months

[Serusers] call signalling between voice mail and answer_machine

by Koyama Zehi

Dear All, I want to know the call signaling between sip user agent and answering_machine. Suppose I made a call for user "matsu" and if she wants to recive only voicemail then then this call will be forawarded to answer_machine by SER proxy. Now what will the call signaling going on before to start the rtp session..either this new call signaling will handle by answer_machine or SER. as when i made a call to the user then who wants only voice mail, I get first the message as "100 try and ringing" on my user agent and after that i get 408 request time out... I need to know that if every thing will fine then what is the call sequence... thanks & regards koyama --------------------------------- Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish.

20 years, 3 months

RE: [Serusers] problem with serctl add

by Arne Scheffer

Jan, The user DB's (subscriber and alias) are both empty (exept for admin account). Please find attached config file. Could it be I am having a problem because the current config does not user authentication yet ? This way I can still do call tests until I have the authentication working. Arne. -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: maandag 26 juli 2004 19:34 To: Arne Scheffer Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] problem with serctl add Do you have lookup("aliases") in your configuration file ? Serweb tries to check if the username does not collide with existing aliases. Jan. On 26-07 12:09, Arne Scheffer wrote: > Hello all, > > I am using latest CVS head (friday 23) of ser and serweb. > > I am having problems with adding users using the serctl > > export SIP_DOMAIN=212.x.x.x > serctl add 200 test test(a)nobody.net > > I always get the following error: > > error: overlap with an existing alias > > I have tried alle types of names als with someone(a)212.x.x.x > > Any ideas ? > > I also ran into a problem trying to add users with serweb (see email on 400 ul_add) > > thanks, > Arne. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

20 years, 3 months

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sr-users July 2004