sr-users November 2003

sr-users@lists.kamailio.org

116 participants
267 discussions

by Kjell Mathiesen

I downloaded the SIP Express Router on Nov. 6. I run a VoIP course (among others) and want a SIP application to show to my students and I found yours interesting. However, I ran into an obstacle, possibly a problem/bug. The details: System: Red Hat 7.3 Personal on a 200 MHz Intel 'Ser' distr.: 0.8.11 RPM-package for RH 7.3 Clients using Windows Messenger Ser.cfg: As distributed but with option <alias="matell5.no"> added (in front of the 'check_via' option. Works fine. <km(a)matell5.no> and others with the same domain can sign in, contact with other clients established etc. Then I add another domain to 'alias' as in <alias="matell5.no" "f19-21.no">. 'ser' starts OK ('ser start' reports [OK] and /var/log/messages says 'ser' is listening on all IP adresses and both aliases are listed and "ser startup successful"). However, attempted sign-in from clients (with the same domain name as earlier) result in ICMP Destination Unreachable/Port Unreachable (port = 5060) and an attempt to stop 'ser' ('ser stop') reports [Failed] and /var/log/messages says "ser shutdown unsuccessful". I have also tried using two lines for the 'alias' parameters (<alias "matell5.no" \n alias "f19-21.no"> with the same results. Then I remove one 'alias' to return to the successful 'ser.cfg' with only one 'alias', and I get the same IMCP message (Port Unreachable). Then I reboot the computer (no further changes to 'ser.cfg') and the clients can sign in without a hitch. The 'ser' Admin's Guide chapter 6.1 'Core Options' clearly indicates that the 'alias' option can take several domain names. (It actually says hostnames. I tried 'mytikas.matell5.no' which is name of the machine 'ser' is running on but this caused the same erroneous result as above. Changed it back to just the domain name and I could sign in normally again. If I am doing something wrong here (or not doing something I should be doing) I would appreciate if you would tell me. I am relatively new to Linux and Linux applications and sometimes find the documentation difficult to understand. If you want me to supply more information like logs etc. let me know (if this really is a bug you probably will). I'll be happy to do more testing/experimenting. I am not running mysql. I'll try that next. Thanks. (Hope I am sending this to the right place.) Kjell Mathiesen Matell Software AS Fururabben 19 1361 Østerås, Norway Tel. (+47) 6714 6509 Mob: (+47) 9073 2212 km(a)matell.no www.matell.no

20 years, 6 months

[Serusers] RE: Reply to Serusers

by sizif＠karaya.com

20 years, 6 months

[Serusers] t_relay_to howto

by Muhammad Nasim

Hi I am having some problems with the simplest tasks in ser. I can't see to get ser to start. It doesn't like my call to the t_relay_to command. Obviously I am being a bit of a thickie. Anyone care to point out where my error is. The following is the whole of the ser.cfg file. # # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode debug=100 fork=yes # listen="192.168.154.81" log_stderror=yes #*/ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # Set natping_interval parameter #modparam("nathelper", "natping_interval", 10) # ------------------------- request routing logic ------------------- # main routing logic route{ log("\nHELLO WORLD\n"); lookup("aliases"); log("\nAFTER lookup aliases\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"); break; }; log("\nAFTER Too Many Hops\n"); if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; log("\nAfter Message too big\n"); # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); log("\nAFTER RECORD ROUTE\n"); # loose-route processing if (loose_route()) { t_relay(); break; }; log("\nAFTER LOOSE ROUTE\n"); #Call to nathlper module functions for Grandstream # fix_nated_contact(); # fix_nated_sdp(3); log("\nJUST BEFORE PSTN stuff\n"); #added by Muhammad # if requests URI is numerical and starts with zero, forward statelessly to a static destination # zero, forward statelessly to a static destination if (uri=~"^sip:0[0-9]*@telser.dyndns.org") { log("\nFORWARDING TO PSTN\n"); #forward( telgw2.dyndns.org, 5060 ); #forward(82.145.32.73, 5060); t_relay_to("82.145.32.73", "5060"); log("\nABOUT TO BREAK\n"); break; }; #end of Muhammad addition # 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("telser.dyndns.org", "subscriber")) { log(8,"XXXXX We are not authorized"); www_challenge("telser.dyndns.org", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; log("\nEND of ROUTING SCRIPT\n"); }

20 years, 6 months

[Serusers] Extensions of SER

by Ivan Dario Salazar

Hello I want to know, what kind of developments or adaptations can i do with SER,so if anybody has gotten expirience with this or knows the e-mail of someone who has done something on their network enviroment, please write me a mail. thanks a lot. _________________________________________________________________ ¿Estás buscando un auto nuevo? http://www.yupimsn.com/autos/

20 years, 6 months

RE: [Serusers] Redirection to voicemail on failure route

by Klaus Darilion

Try the config files of Pablo: http://lists.iptel.org/pipermail/serusers/2003-October/003192.html klaus -----Original Message----- From: jaime.gil(a)orange.co.uk [mailto:jaime.gil@orange.co.uk] Sent: Fri 07.11.2003 13:14 To: serusers(a)lists.iptel.org Cc: Subject: [Serusers] Redirection to voicemail on failure route Hello, We are trying to get voicemail working when an invitation fails. The script is like the following: -------------------------------------------------------------------- On port 5060, SER is running a script containing: <....> if (method==INVITE) { log(1, "INVITE\n"); t_on_failure("1"); }; <..> t_relay(); <...> failure_route[1] { # Redirect to Voicemail ??????????????? } ------------------------------------------------------------------ On port 5080, another SER process launches the voicemail when an incoming request arrives. Does anyone know how would the script look like on the "failure_route" bit? Thanks in advance, Jaime ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. ******************************************************************************* _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

20 years, 6 months

[Serusers] Multiple sers to mysql & asterish voicemail

by Tristan Colgate

Hi, I am currently setting up a SER platform and would like to use Asterisk for pstn and voicemail. I have three hosts, one machine has the mysql db and a dns with an enum zone. I then have the main sip registrar/proy and the final box is the asterisk server. My intention is to have an instance of ser running on the asterisk server to verify authentication and check users priviledges for access to the pstn/voicemail and to enforce the Remote Party ID, this will then nat the connection to the asterisk process' sip channel (either running 127,0.0.1. So is this theoretically sound? And can I have the SER proxy and the SER instance on the * both talking to the same mysql DB? -- Tristan Colgate Inux Technologies E-Mail: tristan(a)inuxtech.co.uk Mobile: 07900 690 912

20 years, 6 months

[Serusers] Redirection to voicemail on failure route

by jaime.gil＠orange.co.uk

Hello, We are trying to get voicemail working when an invitation fails. The script is like the following: -------------------------------------------------------------------- On port 5060, SER is running a script containing: <....> if (method==INVITE) { log(1, "INVITE\n"); t_on_failure("1"); }; <..> t_relay(); <...> failure_route[1] { # Redirect to Voicemail ??????????????? } ------------------------------------------------------------------ On port 5080, another SER process launches the voicemail when an incoming request arrives. Does anyone know how would the script look like on the "failure_route" bit? Thanks in advance, Jaime ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

20 years, 6 months

Re: [Serusers] Unresolveable destination

by Jiri Kuthan

Your server presumably is not configured to handle vfrnd.jp as a served domain and tries to forward to it instead. Apparently, such a domain is not in DNS system. jiri@fox:~$ host vfrnd.jp vfrnd.jp A record not found, server failure jiri@fox:~$ host -t srv _sip._udp.vfrnd.jp _sip._udp.vfrnd.jp SRV record not found, server failure -jiri At 09:21 AM 11/7/2003, Iizuka, Hirotaka, VF-JP wrote: >Content-Class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C3A508.14DFEAEA" > >Hello, > >I have a problem with registering to the SER. >I use X-Lite SIP softphone, and when I try to register to the SER, I get following messages; > >(c)2003 Xten Networks Inc. All rights reserved. >Private build: 1085 > >Established SIP protocol listen on: 192.168.2.101:5060 > >Discovered Port Restricted Cone NAT Firewall > >SIP: 192.168.2.101:5060 >RTP: 192.168.2.101:8000 >NAT: 61.116.104.137 > >PROXY#0: 192.168.2.253:5060 > >SEND >> 192.168.2.253:5060 >REGISTER sip:vfrnd.jp SIP/2.0 >Via: SIP/2.0/UDP 192.168.2.101:5060;rport;branch=z9hG4bK235BF512110411D8A98C0090FE68D625 >From: hiro <sip:iizuka@vfrnd.jp> >To: hiro <sip:iizuka@vfrnd.jp> >Contact: "hiro" <sip:iizuka@192.168.2.101:5060> >Call-ID: 235BF50F110411D8A98C0090FE68D625(a)vfrnd.jp >CSeq: 61073 REGISTER >Expires: 1800 >Max-Forwards: 70 >User-Agent: X-Lite build 1085 >Content-Length: 0 > >SEND >> 192.168.2.253:5060 >REGISTER sip:vfrnd.jp SIP/2.0 >Via: SIP/2.0/UDP 192.168.2.101:5060;rport;branch=z9hG4bK235BF512110411D8A98C0090FE68D625 >From: hiro <sip:iizuka@vfrnd.jp> >To: hiro <sip:iizuka@vfrnd.jp> >Contact: "hiro" <sip:iizuka@192.168.2.101:5060> >Call-ID: 235BF50F110411D8A98C0090FE68D625(a)vfrnd.jp >CSeq: 61073 REGISTER >Expires: 1800 >Max-Forwards: 70 >User-Agent: X-Lite build 1085 >Content-Length: 0 > >RECEIVE << 192.168.2.253:5060 >SIP/2.0 478 Unresolveable destination (478/TM) >Via: SIP/2.0/UDP 192.168.2.101:5060;rport=5060;branch=z9hG4bK235BF512110411D8A98C0090FE68D625 >From: hiro <sip:iizuka@vfrnd.jp> >To: hiro <sip:iizuka@vfrnd.jp>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-0e7b >Call-ID: 235BF50F110411D8A98C0090FE68D625(a)vfrnd.jp >CSeq: 61073 REGISTER >Server: Sip EXpress router (0.8.11 (i386/linux)) >Content-Length: 0 >Warning: 392 192.168.2.253:5060 "Noisy feedback tells: >pid=11731 req_src_ip=192.168.2.101 req_src_port=5060 in_uri=sip:vfrnd.jp out_uri=sip:vfrnd.jp via_cnt==1" > >Thanks in advance. > >Hiro >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

20 years, 6 months

[Serusers] Unresolveable destination

by Iizuka, Hirotaka, VF-JP

Hello, I have a problem with registering to the SER. I use X-Lite SIP softphone, and when I try to register to the SER, I get following messages; (c)2003 Xten Networks Inc. All rights reserved. Private build: 1085 Established SIP protocol listen on: 192.168.2.101:5060 Discovered Port Restricted Cone NAT Firewall SIP: 192.168.2.101:5060 RTP: 192.168.2.101:8000 NAT: 61.116.104.137 PROXY#0: 192.168.2.253:5060 SEND >> 192.168.2.253:5060 REGISTER sip:vfrnd.jp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.101:5060;rport;branch=z9hG4bK235BF512110411D8A98C0090FE68D625 From: hiro <sip:iizuka@vfrnd.jp> To: hiro <sip:iizuka@vfrnd.jp> Contact: "hiro" <sip:iizuka@192.168.2.101:5060> Call-ID: 235BF50F110411D8A98C0090FE68D625(a)vfrnd.jp CSeq: 61073 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite build 1085 Content-Length: 0 SEND >> 192.168.2.253:5060 REGISTER sip:vfrnd.jp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.101:5060;rport;branch=z9hG4bK235BF512110411D8A98C0090FE68D625 From: hiro <sip:iizuka@vfrnd.jp> To: hiro <sip:iizuka@vfrnd.jp> Contact: "hiro" <sip:iizuka@192.168.2.101:5060> Call-ID: 235BF50F110411D8A98C0090FE68D625(a)vfrnd.jp CSeq: 61073 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite build 1085 Content-Length: 0 RECEIVE << 192.168.2.253:5060 SIP/2.0 478 Unresolveable destination (478/TM) Via: SIP/2.0/UDP 192.168.2.101:5060;rport=5060;branch=z9hG4bK235BF512110411D8A98C0090FE68D625 From: hiro <sip:iizuka@vfrnd.jp> To: hiro <sip:iizuka@vfrnd.jp>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-0e7b Call-ID: 235BF50F110411D8A98C0090FE68D625(a)vfrnd.jp CSeq: 61073 REGISTER Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0 Warning: 392 192.168.2.253:5060 "Noisy feedback tells: pid=11731 req_src_ip=192.168.2.101 req_src_port=5060 in_uri=sip:vfrnd.jp out_uri=sip:vfrnd.jp via_cnt==1" Thanks in advance. Hiro

20 years, 6 months

Re: [Serusers] SER Voicemail

by jaime.gil＠orange.co.uk

Jan, Sesha, Ser cannot initiate the transfer by itself, but a simple application talking to the fifo_server in SER can. That's how we worked it out. The drawback is that it only works with IP phones with REFER support (Mitel, Cisco, Pingtel...), not with softclients. Jaime > > (1) How do I set the SER standalone Voicemail config, so that I can reach > Voicemail after 4 rings? You can't reach voicemail after 4 rings because rings are usually generated locally by the caller's phone. You can reach the voicemail after some time, but not measured by number of rings. You can use fr_inv_timer of the tm module for that. > (2) Is SER capable of transferring a call originating from a SIP phone to a > phone on the PBX? If yes, What do I need to set in the configuration? The originating phone must do the transfer, ser can't initiate the transfer. I suppose you mean the transfer of an established call ? Jan. ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

20 years, 6 months

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sr-users November 2003