- sr-users - kamailio.org

[Users] access control list using avps
by jayesh nambiar 29 Sep '05

29 Sep '05

hi daniel, the $avp is an alias to i:800. My main doubt is where in the table should i mention the values 1,2,3 and 4 for the users who got the appropriate privileges. I am sorry, for i get confused manipulating the values of avp. pls help in this. thanks a lot jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

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[Serusers] about my previous question [please help me about how to config radiusclient!]
by zhangshuai 29 Sep '05

29 Sep '05

Dear all, Sorry to bother you again. After googled the error messages: Sep 29 17:11:06 localhost ser[4408]: rc_ip_hostname: couldn't look up host by addr: DA61FC29 Sep 29 17:11:06 localhost ser[4408]: rc_send_server: no reply from RADIUS server unknown:1812 I found this: http://forums.staros.com/showthread.php?p=21895 "The key thing most people miss is that the IP you enter in the radius server for the remote that will talk to it has to be the IP of the device that the server is listening on. Most people simply use the IP of the gateway device and that is not correct unless that device also has Hotspot or PPPoE or some other service that talks to radius." I still don't know how to check his words are right. What do you think? Shuai http://www.goldentek.biz

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[Serusers] please help me about how to config radiusclient!
by zhangshuai 29 Sep '05

29 Sep '05

Hi all, I wish to use softphone X-lite on SER and radius in db_mode. When I installed both freeradius 1.0.4 and radiusclient 0.4.3 on the same server which ser 0.8.14 has installed on, I could log in X-lite through radius authorization. And now I want radiusclient (on server A) to send the request to a remote radius server (on server B) to realize authorization. I have changed those config files: /usr/local/etc/radiusclient/servers, /usr/local/etc/radiusclient/radiusclient.conf and /usr/local/etc/raddb/clients.conf as the document (http://www.iptel.org/ser/doc/ser_radius/ser_radius.html#AEN193) did. Then I test the radius server as the above document said on server A: radclient -f digest server B auth testing123 and I received expected replied message from server B. But when I log in x-lite, there are error messages: Sep 29 17:11:06 localhost ser[4408]: rc_ip_hostname: couldn't look up host by addr: DA61FC29 Sep 29 17:11:06 localhost ser[4408]: rc_send_server: no reply from RADIUS server unknown:1812 How come? Did I mis-config anything? Many thanks for your reply! Shuai http://www.goldentek.biz

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RV: [Users] RV: Nat Help
by Fernando Rodriguez 29 Sep '05

29 Sep '05

Does any one has any idea or can point me to a place to read about this .. Thanks .. _____ De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre de Fernando Rodriguez Enviado el: Miércoles, 28 de Septiembre de 2005 11:18 a.m. Para: users(a)openser.org Asunto: [Users] RV: Nat Help Hello y have 3 interfaces on my box 1 Lan to Internet 2 Lan to 192.168.1.x 3 Lan to 192.168.108.x I only have one domain in my openser I want to know a method to use the mediaproxy only when sending packet tru interfaces for example When making a call from lan 108 to internet or 192.168.1.x Im putting this kind of rules I set different number for number for each lan 1 for eth0 2 for eth1 3 for eth2 so each client has different number regarding the interface he sees .. Is this valid ?? if (from_uri=~"sip:|1[1]|2[0-9]|3[0-9][:@]" && to_uri=~"sip:|1[2]|2[0-9]|3[0-9]|4[0-9][:@]") { log("de 1 a 2\n"); use_media_proxy(); }; if (from_uri=~"sip:|1[2]|2[0-9]|3[0-9]|[0-9][:@]" && to_uri=~"sip:|1[1]|2[0-9]|3[0-9][:@]") { log("de 2 a 1\n"); use_media_proxy(); }; And not to use mediaproxy when calling within the same lan. Thanks for your help Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

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[Users] load_credentials
by Richard Z 29 Sep '05

29 Sep '05

Hi, I am trying to use load_credentials modparam in auth_db to load some colums in the subscriber table. In mysql db table, what type should I use the columns? For example, modparam("auth_db", "load_credentials", "i:15=timeout") If I define timeout as varchar, it gets the value. If as int, it doesn't generate the avp. Is this supposed to be this way? Do I miss anything? Thanks, Richard

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[Users] avp error
by jayesh nambiar 29 Sep '05

29 Sep '05

hi all, can somebody pls tell me what this error is!!! ERROR:avpops_init: "AVP_DB" present but "AVP_TABLE" found empty my db scheme is as follows: modparam("avpops","db_scheme","scheme0:username_col=username;domain_col=domain;value_col=acl;value_type=integer;table=subscriber") I have added a column called acl in the subscriber table and added integer value in it. for eg: for user 201 the acl column value is 4 so that he is privileged to make international calls. Is this correct or some mistake in my understanding. pls let me know.. thanks a lot. jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

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[Serusers] Re: Sems IVR Module
by Nicky 29 Sep '05

29 Sep '05

Please help. ----- Original Message ----- From: "Nicky" <nicky(a)caliber.com.sg> To: <serusers(a)lists.iptel.org>; <sems(a)lists.iptel.org> Sent: Wednesday, September 28, 2005 10:38 PM Subject: Sems IVR Module > Hi all, > > I am testing the sems version from the Head and when i uses a SIP Phone to > call into > the IVR module for recording when I am in the middle of recording, the > program detected DTMF > signal and jump out of the script. Why is it so? as I never pressed on any > key? How can I resolve this > problem? Why is the voice signal become the DTMF signal? Is there any > adjustment that > i can do to solve this problem , as my voice goes louder the more DTMF > signal is being detected. > > Please help. > > regards, > nicky >

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[Serusers] can not open dictionary from freeradiusd!
by Manoranjan Pandey 29 Sep '05

29 Sep '05

hello, guys: I installed freeradiusd now, i want to connect the two together. the problem is that when i use the radiusd-xyz raddb/dictionary, and run radiusd -X, then the messages came out as follow: ------------------------------------- [root@localhost ~]# radiusd -xyz Starting - reading configuration files ... Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary[14]: Couldn't open dictionary " /usr/local/share/freeradius/dictionary": No such file or directory Errors reading radiusd.conf again i am using this command--- [root@localhost ~]# radiusd -x Starting - reading configuration files ... Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary[14]: Could n't open dictionary " /usr/local/share/freeradius/dictionary": No such file o r directory Errors reading radiusd.conf again i am using .. [root@localhost ~]# radiusd -y Thu Sep 29 11:03:19 2005 : Info: Starting - reading configuration files ... again i am using .. [root@localhost ~]# radiusd Thu Sep 29 10:54:56 2005 : Info: Starting - reading configuration files ... WHEN I AM TRYING WITH RADPWTST ----------------------- [root@localhost ~]# radpwtst Bad format in dictionary '/usr/local/etc/raddb/dictionary' at line 14: $INCLUDE /usr/local/share/freeradius/dictionary Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 2 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Access-Request... No reply Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 44 is not defined in your dictionary Attribute number 40 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 41 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Accounting-Request Start... No reply Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 44 is not defined in your dictionary Attribute number 40 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 41 is not defined in your dictionary Attribute number 46 is not defined in your dictionary Attribute number 42 is not defined in your dictionary Attribute number 43 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Accounting-Request Stop... No reply [root@localhost ~]# radpwtst Bad format in dictionary '/usr/local/etc/raddb/dictionary' at line 14: $INCLUDE /usr/local/share/freeradius/dictionary Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 2 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Access-Request... No reply Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 44 is not defined in your dictionary Attribute number 40 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 41 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Accounting-Request Start... No reply Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 44 is not defined in your dictionary Attribute number 40 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 41 is not defined in your dictionary Attribute number 46 is not defined in your dictionary Attribute number 42 is not defined in your dictionary Attribute number 43 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Accounting-Request Stop... No reply please try to mail me olution of this problem. thanking you and waiting for ur reply. bye.

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[Serusers] Changing contents of TO address
by Russell Cozens 29 Sep '05

29 Sep '05

Hi, I am trying to setup a PSTN gateway, the documentation is fine for this and I have it working successfully with a gateway. However, the gateway I would like to use is a bit fussy and seems to be looking at the SIP TO address as well as the INVITE line. I am using rewritehost to change the domain in the uri and this changes the INVITE header but not the SIP TO address within the INVITE. Is it possible to change this ? I've even tried various replace commands from textops.so but the "to" field is never changed. Thanks Russell

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[Serusers] Arbitrary queries
by sip 28 Sep '05

28 Sep '05

Is the only way to make arbitrary queries from the DB to use an external script? I'm having a devil of a time creating something like: SELECT from_uri FROM acc WHERE to_uri LIKE '%<username@domain>%' AND sip_method='INVITE' ORDER BY TIME DESC LIMIT 1 Any ideas? Or am I stuck with external scripts? N.

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[Users] access control list using avps
by jayesh nambiar 28 Sep '05

28 Sep '05

hi daniel, thanx a lot for the help. I'll definitely try creating a new table. Yeah the example is exactly what i need to achieve at present. Based on that i am planning to enhance my features. But what would the column 'acl' contain, will it contain the integer value based on the user privileges. I did'nt try yet as I am unable to understand the bitwise operation. Sorry if it seems rather stupid. thanx again jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

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[Serusers] Regular Expression Problems in this if statement?
by Sherwood McGowan 28 Sep '05

28 Sep '05

Hello all, very glad to have been a user on this list, though I have been lurking. I've checked the documentation and have had trouble finding out information on how to translate my standard regexes (perl) into SER's routing logic I'm trying to use the following logic: if (uri =~ "^sip:1[0-9]{10}\@[0-9]{3}\.[0-9]{3}\.1\.foo\.bar\.net") { #perform actions } Now, that SHOULD match the following scenario: uri has 1 + 10 digits + @ + 3digits + . + 3digits + . + 1 + .foor.bar.net Unfortunately, it's not working.Any help translating the regex, or even just a link to the syntax allowed, maybe it's not allowing {n} to match n times? Thanks, Sherwood McGowan

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[Users] RV: Nat Help
by Fernando Rodriguez 28 Sep '05

28 Sep '05

Hello y have 3 interfaces on my box 1 Lan to Internet 2 Lan to 192.168.1.x 3 Lan to 192.168.108.x I only have one domain in my openser I want to know a method to use the mediaproxy only when sending packet tru interfaces for example When making a call from lan 108 to internet or 192.168.1.x Im putting this kind of rules I set different number for number for each lan 1 for eth0 2 for eth1 3 for eth2 so each client has different number regarding the interface he sees .. Is this valid ?? if (from_uri=~"sip:|1[1]|2[0-9]|3[0-9][:@]" && to_uri=~"sip:|1[2]|2[0-9]|3[0-9]|4[0-9][:@]") { log("de 1 a 2\n"); use_media_proxy(); }; if (from_uri=~"sip:|1[2]|2[0-9]|3[0-9]|[0-9][:@]" && to_uri=~"sip:|1[1]|2[0-9]|3[0-9][:@]") { log("de 2 a 1\n"); use_media_proxy(); }; And not to use mediaproxy when calling within the same lan. Thanks for your help Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

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RE: [Serusers] Regular Expression Problems in this if statement?
by Sherwood McGowan 28 Sep '05

28 Sep '05

Thanks guys for your reponses _____ From: Ramin NIkaeen [mailto:Rnikaeen@goldline.net] Sent: Wednesday, September 28, 2005 12:29 PM To: Sherwood McGowan Subject: RE: [Serusers] Regular Expression Problems in this if statement? Try uri =~ "^sip:1[0-9]{10}@[0-9]{3}\.[0-9]{3}\.1\.foo\.bar\.net") ie. Remove "\" before the "@". Let me know if it works. ramin -----Original Message----- From: Sherwood McGowan [mailto:madprofzero@yahoo.com] Sent: Wednesday, September 28, 2005 12:22 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Regular Expression Problems in this if statement? Hello all, very glad to have been a user on this list, though I have been lurking. I've checked the documentation and have had trouble finding out information on how to translate my standard regexes (perl) into SER's routing logic I'm trying to use the following logic: if (uri =~ "^sip:1[0-9]{10}\@[0-9]{3}\.[0-9]{3}\.1\.foo\.bar\.net") { #perform actions } Now, that SHOULD match the following scenario: uri has 1 + 10 digits + @ + 3digits + . + 3digits + . + 1 + .foor.bar.net Unfortunately, it's not working.Any help translating the regex, or even just a link to the syntax allowed, maybe it's not allowing {n} to match n times? Thanks, Sherwood McGowan

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[Users] access control list using avps
by jayesh nambiar 28 Sep '05

28 Sep '05

hi all, I tried going through the avp documents but I have a few doubts related to avp_db_load and avp_check functions. I wanted to implement the access control list. the access control values may be jus like: 1 => local calls 2 => long distance calls 3 => international calls. I am using the default table usr_preferences. when I do the avp_check as: if(!avp_check("$acl", "and/0x01")) I get an error that it cannot be parsed and fix the avp_check. And can someone pls explain how the bitwise check "and", "or" etc works.. thanks, jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

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[Users] business mailing list
by Daniel-Constantin Mierla 28 Sep '05

28 Sep '05

Hello everybody, a new mailing list was created to provide a forum for parties who wish to discuss commercial or business related activities, opportunities, etc. related to OpenSER, VoIP, SIP ... The email address of the mailing list is "business(a)openser.org". To subscribe/unsubscribe, follow the link: http://openser.org/cgi-bin/mailman/listinfo/business It is list where everybody can subscribe but no archiving is done. This should force anyone interested in some business to get up-to-date information, avoiding questions related to past mails found on web archive. Also, it should prevent some spaming from users trying to increase search engine's ranking via links and message posts. The usual mails sent to this list should be in the form of "I need _this_, who does it?" or "I do _this_, who needs it?", but should not be limited to such pattern. Basic rules for posting on the mailing list: - keep a fair conduit of messages, no bad remarks about others or counter-advertising - do not send images, flash or other graphics - post links instead - it is recommended to reply directly to the sender when it is a one-to-one business, the others may not be interested, otherwise they should post a message - if nobody answers, then nobody needs/provides what you offer/need, it is no reason to send the message hourly, wait for some days or weeks Hopefully, it will help everybody for a mutual benefit. Cheers, Daniel

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[Serusers] How to terminate a session?
by Federico Giannici 28 Sep '05

28 Sep '05

Suppose that we know which calls are open, based on the accounting data, so we have all dialog parameters (Call-ID, From, To, C-Seqs, etc...). How can I terminate a session that is lasting too long? If I suppose that a session is actually closed (for example because one UA has crashed), how can I be sure that it is really closed (eventually actually closing it)? I'm thinking to send two BYE messages to both UAs (impersonating the other UA), so to make them close the session (if it is still open). Is this tecnically possible? Are there any contraindications? Is there a better way to obtain it? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Serusers] Aliases and Location Problem... please help
by Ryan Pagquil 28 Sep '05

28 Sep '05

Hi SER users, Why does my aliases and locations are not persistent? Considering that they appear in my database (aliases and location), when I restarted SER all aliases and locations are not being read by SER. What will I do is to restart my X-lite softphone for me to re-register. And for my aliases I will run again serctl alias add <alias> sip:<uri> for the aliases to work. It seems like SER is not reading my database and all contact informations are only stored on its memory. But all these contacts appears in my location and aliases tables of the ser database. Is there any known bugs about this? Please help. I'm using SER-0.9.3 in SPARC64 Debian linux with mediaproxy as nat traversal solution. Any help is greately appreciated... 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

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[Serusers] 483 Too Many Hops problem ;;;;
by 송 미진 27 Sep '05

27 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 dear, I'm working on a project and i'm using ser as a sip server. actually i have this problem: every time i send an instant message to a subscriber i get this error message: 483 too many hops i tried to fix it but i couldn't and i don't know what to do ! (My computer connect many computers. Is it reason?!) if any one can help me i'll be very grateful i'm using ser 0.9.3 When I use sipsak, I received following messages. root]# sipsak -T -s sip:user@mydomain.com warning: IP extract from warning activated to be more informational 0: 192.168.178.194 (0.148 ms) SIP/2.0 483 Too Many Hops 1: 192.168.178.194 (1.915 ms) SIP/2.0 483 Too Many Hops 2: 192.168.178.194 (25.673 ms) SIP/2.0 483 Too Many Hops 3: 192.168.178.194 (4.855 ms) SIP/2.0 483 Too Many Hops 4: 192.168.178.194 (9.286 ms) SIP/2.0 483 Too Many Hops 5: 192.168.178.194 (20.285 ms) SIP/2.0 483 Too Many Hops 6: 192.168.178.194 (17.068 ms) SIP/2.0 483 Too Many Hops 7: 192.168.178.194 (15.038 ms) SIP/2.0 483 Too Many Hops 8: 192.168.178.194 (16.379 ms) SIP/2.0 483 Too Many Hops 9: timeout after 500 ms 9: timeout after 1000 ms 9: timeout after 2000 ms 9: 192.168.178.194 (1480.452 ms) SIP/2.0 483 Too Many Hops 10: 192.168.178.194 (20.068 ms) SIP/2.0 483 Too Many Hops 11: 192.168.178.194 (21.916 ms) SIP/2.0 483 Too Many Hops 12: 192.168.178.194 (22.478 ms) SIP/2.0 483 Too Many Hops 13: 192.168.178.194 (22.984 ms) SIP/2.0 513 Message too big without Contact header ====================================== here's my code: # ----------- 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=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://root:vlftmd@localhost/ser" alias=sam.cse # ------------------ 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_url","mysql://root:vlftmd@localhost/ser") #modparam("usrloc", "db_mode", 0) modparam("registrar","use_domain",1) # 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","secret","123123123") #modparam("auth_db","db_url","mysql://root:vlftmd@localhost/ser") #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("sam.cse", "subscriber")) { www_challenge("sam.cse", "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(); }; } _________________________________________________________________ 싸게 싸게 MSN공동구매 http://www.waawaa.com/cobuy/cobuy_default.asp?siteid=10160

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[Users] Nat Help
by Fernando Rodriguez 27 Sep '05

27 Sep '05

Hello y have 3 interfaces on my box 1 Lan to Internet 2 Lan to 192.168.1.x 3 Lan to 192.168.108.x I only have one domain in my openser I want to know a method to use the mediaproxy only when sending packet tru interfaces for example When making a call from lan 108 to internet o 192.168.1.x And not to use mediaproxy when calling within the same lan. Thanks for your help Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

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[Serusers] problem detectec: Not found in usrloc
by Ángel Gómez 27 Sep '05

27 Sep '05

Hi, I have found the problem, but i don't know ho to solve it. To search the problem I started SER with a debug level 9, redirectin STDERROR to a file ( if some one want to do it is: /etc/init.d/ser start 2>>/ser.log1 ) and when I was reading the file I found it: 2(1926) lookup(): 'ninguna' Not found in usrloc This is what the server say when i send a MESSAGE to ninguna(a)10.22.1.249, but then I used: serctl ul show, and got this: ...Record(0x4045d1b8)... domain: 'location' aor : 'ninguna' ~~~Contact(0x4045d218)~~~ domain : 'location' aor : 'ninguna' Contact : 'sip:ninguna@10.22.1.249' Expires : 3583 q : Call-ID : '1549255136(a)10.22.1.5' CSeq : 1 User-Agent: 'sipsak 0.8.9_pre' received : '' State : CS_SYNC Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... can any one give me a solution? BEFORE of sending the message I send a REFISTER for the user ninguna.

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[Serusers] Adding Aliases for inbound PSTN
by Ashutosh kumar 27 Sep '05

27 Sep '05

Hi, I want to configure my ser.cfg so as to bind a user john to a DID 1234 from my pstn switch. John is a user in my ser DB and i want to assign him the phone number 1234, so that whenever my pstn GW forwards this call for 1234 to SER, SER should forward this call to user john.Which command do i use ni this case. serctl alias add john 1234, OR serctl alias add 1234 john, beacuse in my case, none of these is working, although i can see the switch forwarding the call to SER. How can i use save("aliases") in this scenario? I have used lookup ("alaises") at appropriate places. Thanks Ashutosh

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[Users] mail to sobomax
by harry gaillac 27 Sep '05

27 Sep '05

(open)ser users, Forget this post I just try to get a reply from Mr Maxim Sobolev. Dear Maxim, Please to reply to my mails I've been sending since several months. http://mail.iptel.org/pipermail/serusers/2005-August/022745.html Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] RTPproxy and SER
by Alexandre Aractingi 27 Sep '05

27 Sep '05

Hi all, I'm trying to use RTPproxy (CVS) and SER to enable locally-routed endpoints to talk to an external public gateway. When a call comes from the external gateway, the SDP in the INVITE gets rewritten properly with SER's IP address, but then when my phone hangs off, the SDP in the 200 OK doesn't get rewritten, so an RFC1918 IP address is passed to the external gateway (so I get no audio). I'm using nathelper with: modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") and for my tests I put "force_rtp_proxy()" at the very beginning of the routing logic in order to make sure it gets invoked. But the SDP in the 200 OK is not rewritten no matter what. Is this a know issue? Is there a known workaround? Thanks in advance for any help, -- Alexandre Aractingi <aaractingi(a)libertysurf.fr>

1 0

[Serusers] cisco 7905G
by Alberto 27 Sep '05

27 Sep '05

Hi, I have the next scheme: 7905G ------- NAT ------- Internet ------ SER+mediaproxy When the 7905G sends a INVITE the ser responds: Status: 407 Proxy Authentication Required. This is right, but the destination port this packet not is right because the ser sends the packet with destination port 5060 (private port) and not the (public port). If I open this port all it is right, I do not want to open no port. This single problem happens to me with 7905G (no linksys, no X-Ten, ...). I do not understand what it is happening. Thanks for your aid. -- Alberto

3 5

RE: [Serusers] Restriction on port number??
by pankaj.munjal＠wipro.com 27 Sep '05

27 Sep '05

Hi, Yes, I've set the SIP proxy address as <proxy-IP>:7065 - Pankaj. -----Original Message----- From: sip [mailto:sip@arcdiv.com] Sent: Tuesday, September 27, 2005 8:32 PM To: Pankaj Munjal (WT01 - Voice & Next Generation Networks); serusers(a)lists.iptel.org Subject: Re: [Serusers] Restriction on port number?? Is x-lite set to use your proxy:7065 ? There's no restriction as far as I know. N. On Tue, 27 Sep 2005 20:17:58 +0530, pankaj.munjal wrote > Hi everyone, > > I had a basic question about SER. > Is there a restriction that we have to bring up SER on port number 5060 only? > > When I tried changing the port number in the ser config file, ser comes up normally, but it doesn't answer the REGISTER message from the UA. > I am using XLite as a UA and bringing up the ser on port 7065. > > Any pointers as to what can be reason for this? > > Cheers, > > Pankaj Munjal > Automation - Nortel Networks > Wipro Technologies, Bangalore. > ESN : 6-877-5666 > PBX: 080- 28520408, Extn:- 83265 > > > Confidentiality Notice > > The information contained in this electronic message and any attachments to this message are intended > for the exclusive use of the addressee(s) and may contain confidential or privileged information. If > you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately > and destroy all copies of this message and any attachments. > Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

2 1

[Serusers] Restriction on port number??
by pankaj.munjal＠wipro.com 27 Sep '05

27 Sep '05

Hi everyone, I had a basic question about SER. Is there a restriction that we have to bring up SER on port number 5060 only? When I tried changing the port number in the ser config file, ser comes up normally, but it doesn't answer the REGISTER message from the UA. I am using XLite as a UA and bringing up the ser on port 7065. Any pointers as to what can be reason for this? Cheers, Pankaj Munjal Automation - Nortel Networks Wipro Technologies, Bangalore. ESN : 6-877-5666 PBX: 080- 28520408, Extn:- 83265 Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

2 1

[Serusers] Database queries?
by sip 27 Sep '05

27 Sep '05

I'm attempting to figure out how to use AVPops to grab data from the calls_forwarding database table so I can avoid having to write my own module to handle that sort of thing, and I can avoid doing what I'm CURRENTLY doing which is using a pretty hideous exec_msg SQL query hack. I've run into a snag that I can't figure a way around.... In the calls_forwarding table, I have these fields used in this fashion: username == username domain == domain uri_re == username@domain information (for use in a couple of ways) purpose == purpose for this entry (callblock, callfwd, screening, etc) action == action to take (for callblock, could be reply or relay, etc) param1 == option for the action above param2 == additional option So... if I have a call block entry, it would be like this: username = bob domain = mydomain.com uri_re = motherinlaw(a)another.domain.com purpose = callblock action = reply param1 = 486 param2 = "Busy" The logic would be along the lines of, when a call comes in for bob(a)mydomain.com, I check to see if he has any callblock entries. If so, I check those to see if any match the From user. If there's a match, take the action in action (in this case reply with 486 - Busy... could be relay to voicemail, etc). The problem I'm having is in the grabbing of all the data from the DB and comparing it to the right values. For instance, with this avpops query avp_db_load("$ruri/username", "$urire/$urire_scheme"); # Load the urire column for this and check to see if # it matches the $from address if(avp_check("$urire", "eq/$from/I")) { send reply... take action... etc. } I'm calling from the number motherinlaw(a)another.domain.com. If I use the above, the avp_check test fails. If I put if(avp_check("$urire", "eq/motherinlaw(a)another.domain.com/I")) { send reply... take action... etc. } Then the test succeeds and the appropriate action is taken. Any guesses as to what I'm missing here... why the test is failing when I use the $from variable for AVPops, but not when I hard-code the $from address into the test (which implies to me that it's grabbing the right data from the database, but there's an issue in either the $from variable or the comparison statement)? I'd love a little shove in the right direction. I thought about writing a module that returned what I need without the need for complex logical paths, but my C is pretty rusty, and I'm trying to save myself the hair-pulling. :) N.

4 9

[Serusers] new voip service [ot] FYI
by Iqbal 27 Sep '05

27 Sep '05

I know I know not supposed to advertise, but this is not my company its dixons, a 1400 retail unit in the UK, will have vonage running for cover in the UK http://www.theregister.co.uk/2005/09/27/dixons_freetalk/ £79.99 includes hardware and unlimited calls to UK landline for a year..... plusnet £3.99 for 3500 mins per month (but need there bandwidth) and ipclouds.co.uk (£69.99) Iqbal

1 0

[Serusers] SEMS+IVR
by Abdul Lateef 27 Sep '05

27 Sep '05

Hi all, I wanted configure IVR with SEMS. I have a samall question. Is SEMS need SER, or SEMS can work with another SIP GK. i have already SIP GK and i wanted to use SEMS priticular for IVR. Please let me know if SEMS can work withou SER? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

[Serusers] Seperate SER and Oubound Proxy
by Kevin Brennan 27 Sep '05

27 Sep '05

We would like to use STUN as preferred NAT traversal method but for those situations where STUN does not work we would offer an outbound proxy (configured in user agent). We would like to keep the NAT traversal proxy completely separate from SER. I wonder has anybody implemented this ? how ? and any other suggestions/feedback. Br /Kevin Brennan.

3 2

RE: [Serusers] is SER ( 0.9.4) 100% 3GPP SIP call flow compliant ?
by Ashish Gilotra 27 Sep '05

27 Sep '05

Hi Klaus Thanks for answer. What is missing in SER which makes it non-compliant to 3GPP IMS call flow ? Br Ashish -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Tuesday, September 27, 2005 2:51 PM To: ashish gilotra Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] is SER ( 0.9.4) 100% 3GPP SIP call flow compliant ? ashish gilotra wrote: > Hi > > We are evaluating SER for our purpose. > > We need to use 3GPP SIP call flows. > > Is SER compliant ? No. regards klaus _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Users] Problem with ACK's (branch=0)
by Rodrigo P. Telles 27 Sep '05

27 Sep '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've been experiencing some troubles with ACK's with branch=0. I found a thread about it but I didn't find a 'solution' folowing the thread. http://mail.iptel.org/pipermail/serdev/2005-April/004296.html Can some one point me to the correct answer for that question? Thanks in advance. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> TI Manager Devel-IT - http://www.devel.it ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDJbvNiLK8unYgEMQRAoATAJwJSLGiN1/XuAUk2aVm4rm5oGD00ACfYJf0 XbL8Vv4unK6U3j974UOtYU0= =gZzI -----END PGP SIGNATURE-----

4 6

[Serusers] use of SRV records for failover
by Markus Storm 27 Sep '05

27 Sep '05

Dear all, we would like to do loadbalancing and failover with SRV in our multi-SER setup. I.e. ser-1 gets a call and can forward the call to either ser-2 or ser-3. There's a DNS SRV record that points to both, ser-2 and ser-3 (same weight, same priority). On ser-1, I use the SRV name to forward calls to. Now I'd like ser-1 to distribute calls according to weight and priority, i.e. 50% to ser-2 and 50% to ser-3. But it doesn't, it keeps sending to ser-2 (the first entry in the SRV record's list). Also, if say ser-2 goes down, I'd like ser-1 to notice that and only ever try to send subsequent requests to ser-2 until <something> happens (e.g. until the SRV record TTL expires). I found a one-year old message in the archives stating that neither of this is possible (unless I use a DNS A record that points to multiple IP addresses, but that's not what I want). Is it meanwhile possible with the current SER software ? thanks Markus

5 4

[Serusers] SER 0.9.3 core dumps
by Darren Nay 27 Sep '05

27 Sep '05

Hey All, I installed SER 0.9.3 about 3 weeks ago and since then have been getting core dumps about every 7-8 days consistently. Acts almost like a progressive memory leak. So .. I've noticed that release 0.9.4 has also been released about 2 weeks ago with a few bug fixes (including a memory leak or two). However, the CVS head is also available with 0.9.5-pre1. So .. What I really need to know is.. Which version should I use? :) Thanks, Darren Nay

3 3

[Serusers] lcr
by Sebastian Kühner 27 Sep '05

27 Sep '05

Hello, I'm using the lcr module with prefix capability to make failover. When I use a "big" IP, the lcr module gives back a wrong IP. It was very hard for me to compile lcr with prefix capability and ser-0.9.3... has anyone of you a lcr module with prefix-support for ser-0.9.3 and without this bug? Thanks! Sebastian

3 4

[Users] Route, Record Route and quintum
by Iqbal 27 Sep '05

27 Sep '05

Hi I have a strange looking dialogue, after sending a 200 OK to a quintum box, and get back an ACK, but the Route field seems to be wrong, can someone confirm this. -------- U 2005/09/06 15:23:16.613003 111.222.333.444:5060 -> 10.20.30.40:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.20.30.40;branch=z9hG4bK-tenor-c3ac-7b24-0018. From: <sip:046677@10.20.30.40>;tag=c3ac7b24-a. To: <sip:123456789@111.222.333.444>;tag=1A4297F4-2C9. Date: Tue, 06 Sep 2005 14:07:59 gmt. Call-ID: call-00CC9E9D-4FBE-D311-0207-A(a)10.20.30.40. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:0044123456789@212.156.5.134:5060>. Record-Route: <sip:212.156.5.135;ftag=c3ac7b24-a;lr=on>,<sip:111.222.333.444;ftag=c3ac7b24-a;lr=on>. Content-Type: application/sdp. Content-Length: 206. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7438 1091 IN IP4 212.156.5.134. s=SIP Call. c=IN IP4 212.156.5.134. t=0 0. m=audio 18590 RTP/AVP 18. c=IN IP4 212.156.5.134. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U 2005/09/06 15:23:16.726095 10.20.30.40:5060 -> 111.222.333.444:5060 ACK sip:111.222.333.444;ftag=c3ac7b24-a;lr=on SIP/2.0. CSeq: 1 ACK. Call-ID: call-00CC9E9D-4FBE-D311-0207-A(a)10.20.30.40. Contact: <sip:046677@10.20.30.40>. From: <sip:046677@10.20.30.40>;tag=c3ac7b24-a. Route: <sip:0044123456789@212.156.5.134:5060>. Session-GUID: 859321956-876164920-926430820-875782963. To: <sip:123456789@111.222.333.444>;tag=1A4297F4-2C9. Via: SIP/2.0/UDP 10.20.30.40;branch=z9hG4bK-tenor-c3ac-7b24-0018. User-Agent: Quintum/1.0.0. Quintum: 0c01030b0232360501000715000000000000000d006c0a828180803034363637371301000f0b413031322d313033353941. Max-Forwards: 70. ---------------- I thought the Route field should have the data which it pulls from the Record Route headers tks iqbal

7 21

[Serusers] SER callerid/ANI removal
by Jason Mohler 27 Sep '05

27 Sep '05

I am trying to find a way to set my SER caller ID to unknown. What I trying to accomplish is to not have calls coming with an ANI. It can be unknown , anonymous, etc. But I do not want the ANI/caller ID number showing. Can anyone help me answer this question? I think it would be an rpid setting but not certain.

5 4

[Users] use_domain not found in avpops
by Franz Edler 27 Sep '05

27 Sep '05

Hi, I just wanted to use the parameter use_domain in avpops, but I got the error "<use_domain> not found in module <avpops>". A quick check in module avpops.c shows that the variable declaration for use_domain is not declared as a pointer. Might this be the cause? Regards Franz

3 3

[Serusers] is SER ( 0.9.4) 100% 3GPP SIP call flow compliant ?
by ashish gilotra 27 Sep '05

27 Sep '05

Hi We are evaluating SER for our purpose. We need to use 3GPP SIP call flows. Is SER compliant ? Thanks Ash __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] SER Vs. SIPsak = 483, incoerence!!!!
by Ángel Gómez 27 Sep '05

27 Sep '05

Nils, thank for making me se this. Despite of this i thin this is starnger, because i maque a REGISTER or INVITE and this doesn´t happen. Now I´m sending my ser.cfg files. To see wath is wrong. # $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" listen=10.22.1.249 alias="10.22.1.249" alias="213.194.156.186" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" 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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/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", 1) # -- 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("16")) { sl_send_reply("483","Too Many Hops"); }; 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; 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(); }; } Thack for all.

1 0

[Serusers] Windows Messenger 5.1 -> SER authentication fails
by davor jovanovic 27 Sep '05

27 Sep '05

Hi all, I have a problem with using Windows Messenger 5.1, that is, I can't log in with SER 0.9.3. Everything is working OK with Xlite client. I try to set up my realm value to equal server name as it says in Admin guide for SER, but it doesn't help. If someone had similar problem and solve it, I would be very thankful if you can share solution with me. Thanks This is output from SER when trying to authenticate with WM. 0(22883) parse_headers: flags=-1 0(22883) check_via_address(192.168.0.101, 192.168.0.101, 0) 0(22883) DEBUG:destroy_avp_list: destroying list (nil) 0(22883) receive_msg: cleaning up 0(22883) SIP Request: 0(22883) method: <REGISTER> 0(22883) uri: <sip:debtest.stu.du> 0(22883) version: <SIP/2.0> 0(22883) parse_headers: flags=1 0(22883) end of header reached, state=5 0(22883) parse_headers: Via found, flags=1 0(22883) parse_headers: this is the first via 0(22883) After parse_msg... 0(22883) preparing to run routing scripts... 0(22883) parse_headers: flags=128 0(22883) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(22883) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(22883) parse_headers: flags=256 0(22883) end of header reached, state=9 0(22883) DEBUG: get_hdr_field: <To> [21]; uri=[sip:admin@debtest.stu.du] 0(22883) DEBUG: to body [<sip:admin@debtest.stu.du> ] 0(22883) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 0(22883) DEBUG: get_hdr_body : content_length=0 0(22883) found end of header 0(22883) find_first_route: No Route headers found 0(22883) loose_route: There is no Route HF 0(22883) grep_sock_info - checking if host==us: 7==12 && [debtest.stu.du] == [161.53.0.131] 0(22883) grep_sock_info - checking if port 5060 matches port 5060 0(22883) grep_sock_info - checking if host==us: 7==12 && [debtest.stu.du] == [161.53.0.131] 0(22883) grep_sock_info - checking if port 5060 matches port 5060 0(22883) parse_headers: flags=-1 0(22883) check_via_address(192.168.0.101, 192.168.0.101, 0) 0(22883) parse_headers: flags=4096 0(22883) pre_auth(): Credentials with given realm not found 0(22883) build_auth_hf(): 'WWW-Authenticate: Digest realm="debtest.stu.du", nonce="43390f58ff776ef2b6700bd7e5cb53ea5b79453b" ' 0(22883) parse_headers: flags=-1 0(22883) check_via_address(192.168.0.101, 192.168.0.101, 0) 0(22883) DEBUG:destroy_avp_list: destroying list (nil) 0(22883) receive_msg: cleaning up

1 0

[Serusers] Register problem
by Sebastian Kühner 26 Sep '05

26 Sep '05

Hi, I installed SER on a new system and now I can't REGISTER anymore. With a little bit luck, the users can REGISTER after a restart of the system, but the register expires and the user is unregistered for ever. SER sends back a Unauthorized, although the user is registerin correctly. It's a phenomenon that I don't understand. May be the database? Here's a log of the database: Sep 26 18:00:22 pbx5 postgres[286]: [13-1] LOG: 00000: statement: BEGIN Sep 26 18:00:22 pbx5 postgres[286]: [13-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[286]: [14-1] LOG: 00000: statement: select userpass,rpid from subscriber where username='43750' Sep 26 18:00:22 pbx5 postgres[286]: [14-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[286]: [15-1] LOG: 00000: statement: COMMIT Sep 26 18:00:22 pbx5 postgres[286]: [15-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[307]: [25-1] LOG: 00000: statement: BEGIN Sep 26 18:00:22 pbx5 postgres[307]: [25-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[307]: [26-1] LOG: 00000: statement: select userpass,rpid from subscriber where username='43750' Sep 26 18:00:22 pbx5 postgres[307]: [26-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[307]: [27-1] LOG: 00000: statement: COMMIT Sep 26 18:00:22 pbx5 postgres[307]: [27-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[282]: [10-1] LOG: 00000: statement: BEGIN Sep 26 18:00:22 pbx5 postgres[282]: [10-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[282]: [11-1] LOG: 00000: statement: select userpass,rpid from subscriber where username='43750' Sep 26 18:00:22 pbx5 postgres[282]: [11-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:00:22 pbx5 postgres[282]: [12-1] LOG: 00000: statement: COMMIT Sep 26 18:00:22 pbx5 postgres[282]: [12-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:01:00 pbx5 postgres[332]: [19-1] LOG: 00000: statement: BEGIN Sep 26 18:01:00 pbx5 postgres[332]: [19-2] LOCATION: pg_parse_query, postgres.c:464 Sep 26 18:01:00 pbx5 postgres[332]: [20-1] LOG: 00000: statement: delete from location where username='43750' AND contact='si p:43750@xxx.xxx.xxx.xxx:63713' AND Can anyone of you help me?? Thanks!!! Sebastian

1 0

[Users] branch route
by Richard Z 26 Sep '05

26 Sep '05

Hi, I am trying to figure out how to use branch route for nat traversal. If a subscriber registers from multiple locations, a branch route is called after lookup. For example, if locations include UAs behind same NAT and different NAT, i'd like to invoke any rtp/media-proxy accordingly in branch_route[], . However they can't be called inside a branch_route. What's the best way to do this? Thanks, Richard

5 10

[Serusers] Try to do effective routing using sip server
by Prajyot 26 Sep '05

26 Sep '05

Hi all, Can anyone please tell me how to configure SIP ser server for effective routing based on DNIS. Right now its pretty static in nature, meaning that we have if queries based on the DNIS and the invite is forwarded to a server. What I am trying to achieve is load balancing in which I can specify more than one server for routing to a particular DNIS. Please guide me, thanks in advance. Regards, Prajyot Shirsolkar

1 0

Re: [Serusers] ONSIP Script + Java Rockx ser 0.9.0 script
by Iqbal 26 Sep '05

26 Sep '05

Hi @Mark: I agree, but some people only really need ser, or should I say if your route to much to asterisk its pointless having ser, I keep my Ip__IP traffic in ser, in fact my billing there also, asterisk as you mentioned for features, and when my upstream gateway wont support REFER to do transfers, and cool things like call pickup. @Aisling: if (method=="INVITE"){ if(is_user_in("Request-URI", "voicemail")){ setflag(31); }; }; what this will do is to see if the username you have in RURI is in your grp table and has an entry of voicemail next to it, if so it will set a flag 31 if (method=="INVITE") { > avp_write("$from", "i:34"); > if (avp_check("i:34", "eq/$ruri/i")) { > > if (isflagset(31)) { > route(8); > break; > } else { > sl_send_reply("486", "Busy"); > break; > }; > }; > }; what this does is basically to see if your are calling yourself, i.e if the from and the RURI is the same, first it stores the from setting, and the compares that to your ruri.Now if they are the same, it will see if flag 31 is set, i.e if you have voicemail attached to the RURI in grp table. If both are true then goto route8, at which point it seems as if your are relaying to another server. You seem to be getting busy, which means that flag 31 is not being set, check your ngrep dump see what your RURI has, and see what entries your grp table has. The problem with matching like this (or at least the problem(s) I had) was that there are alot of different call scenarios which all need to be taken into account. EG my usernames are not e164 numbers, they are part of that number, which means my ruri, and from can be different to the e164 format. Now when a caller calls from 040600, to 0845040602, it should see if 0845xxx is busy, send to voicemail, but in grp ther is no 0845xxx, hence I need to make sure that the check/match is done after the alias lookup. I had to move lookup(aliases) near the to to make sure all my lookups had been done at the start, and then I can start matching. Your best friend will be ngrep, if you start doing the diff call scenarios, and seeing what needs to be matched, you can set 1 run settings, once it all works work out a tidier logic. If you throw asterisk into the loop, you add another headache (depending on what you do), cause asterisk was really designed for ip devices to register with it, but if you register in ser, you could lose some of that info (although latest beta, has some nice SIP header parsing, to pull info from). The other problem you may hit with voicemail, depends on how you set your username, initially my subscriber table had names for username...eg Iqbal, and the alias would be 040600, or whatever, problem with this was that voicemail in asterisk is better with number, especially when you have to dial the mailbox number from your phone :-), so I switched the subscriber table to hold the number, cause I think unless you will be using software dialers, no one will type in names. "voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type >=1:702") All this does is to define aliases, you can use it, or not iqbal Mark Aiken wrote: > I see a lot of people trying to use SER and Asterisk together but > mixing the roles in strange and complex ways. > > Asterisk is an excellent SIP "feature server" and "b2bua" where SER is > an excellent SIP proxy, registrar, and far-end NAT solution. > > If you design your network based on those strengths you can create a > solution that rivals 7 figure commercial solutions. People that do > otherwise seem to just fall over themselves. > > Mark > > > On 9/21/05, *Aisling* <ashling.odriscoll(a)cit.ie > <mailto:ashling.odriscoll@cit.ie>> wrote: > > Many thanks Iqbal- That was the problem. > > I think the issue is that I don't even really understand what that > line > does or how the code that I took from Paul's config affected my config > file.....SER now displays no errors but is still not working as > expected. For example I expect that if a phone (e.g. 2092) dials > its own > extension, it should be caught by > > if (method=="INVITE"){ > if(is_user_in("Request-URI", "voicemail")){ > setflag(31); > }; > }; > > This then should be caught by the URI Compare code below and routed to > route[8].... > > However that's not happening and a 486 Busy is being returned to the > phone. I find this hard to troubleshoot because I think the code > should > be correct...given my current understanding anyhow... > > If Iqbal (or anyone else) has a clearer understanding of this and can > shed some light on it, that would be great. > > Kindest Regards, > Aisling. > > -----Original Message----- > From: Iqbal [mailto:iqbal@gigo.co.uk <mailto:iqbal@gigo.co.uk>] > Sent: 21 September 2005 15:20 > To: Aisling O'Driscoll > Cc: serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > Subject: Re: [Serusers] ONSIP Script + Java Rockx ser 0.9.0 script > > "voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type > =1:702") > > try a voicemail=i:500 not 1:500 > > Iqbal > > > > Aisling O'Driscoll wrote: > > >Hello everyone, > > > >I am trying to modify the Onsip call forward script to forward to > >asterisk voicemail given a users preference. I am using the script > >that Java RockX (Paul) posted in January... > >http://lists.iptel.org/pipermail/serusers/2005-January/014968.html > > > >I think I am getting very confused though with the logic...I want > >users to be able to load their preference using the avpops module of > >whether they want call forward on no answer or voicemail. They get > >directed to voicemail if they dial their own extension number....I > >have posted the modified script which is a combination of the Pauls > >script and the Onsip script...When I start SER I currently have these > >errors but I'd say these are a consequence of my messed up logic > >;).... > > > >0(20031) ERROR:parse_avp_name: unsupported type [1] > > 0(20031) ERROR:add_avp_galias_str: <500> not a valid AVP name > > 0(20031) parse error (70,20-21): Can't set module parameter > > > >The config is below.... > >If anyone can offer some insight, Id be very grateful. > >Many thanks, > >Aisling. > > > ># $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. > >Teigre/emailAddress= greger(a)onsip.org <mailto:greger@onsip.org> $ > >#debug=3 > >#fork=yes > >#log_stderror=no > > > >#listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE > >#port=5060 > >#children=4 > > > >check_via=no > >dns=no > >rev_dns=no > >fifo="/tmp/ser_fifo" > >fifo_db_url="mysql://root:password@localhost/ser" > > > >alias="x.x.x.x:5060" > > > >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" > >loadmodule "/usr/local/lib/ser/modules/uri.so" > >loadmodule "/usr/local/lib/ser/modules/uri_db.so" > >loadmodule "/usr/local/lib/ser/modules/nathelper.so" > >loadmodule "/usr/local/lib/ser/modules/textops.so" > >loadmodule "/usr/local/lib/ser/modules/cpl-c.so" > >loadmodule "/usr/local/lib/ser/modules/avpops.so" > >loadmodule "/usr/local/lib/ser/modules/permissions.so" > >loadmodule "/usr/local/lib/ser/modules/speeddial.so" > >loadmodule "/usr/local/lib/ser/modules/group.so" > > > >modparam("auth_db|permissions|uri_db|usrloc", "db_url", > >"mysql://root:password@localhost/ser") > >modparam("auth_db", "calculate_ha1", 1) > >modparam("auth_db", "password_column", "password") > > > >modparam("nathelper", "natping_interval", 30) #ping every 30 seconds > >modparam("nathelper", "ping_nated_only", 1) > >modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") > > > >modparam("cpl-c", "cpl_db", "mysql://root:password@localhost/ser") > >modparam("cpl-c", "cpl_table", "cpl") > >modparam("cpl-c", "cpl_dtd_file", > >"/tmp/ser-0.9.3/modules/cpl-c/cpl-06.dtd") > >modparam("cpl-c", "lookup_domain", "location") > > > >modparam("tm", "fr_inv_timer", 27) > >modparam("tm", "fr_inv_timer_avp", "inv_timeout") > > > >modparam("permissions", "db_mode", 1) > >modparam("permissions", "trusted_table", "trusted") > > > >modparam("usrloc", "db_mode", 2) > > > >modparam("registrar", "nat_flag", 6) > > > >modparam("rr", "enable_full_lr", 1) > > > >modparam("speeddial", "db_url", > "mysql://root:password@localhost/ser") > >modparam("speeddial", "user_column", "userid") > >modparam("speeddial", "sd_user_column", "short_user") > >modparam("speeddial", "sd_domain_column", "short_domain") > >modparam("speeddial", "new_uri_column", "real_uri") > > > >modparam("avpops", "avp_url", "mysql://root:password@localhost/ser") > >modparam("avpops", "avp_table", "usr_preferences") > >modparam("avpops", "avp_aliases", > >"voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type > >=1:702") > > > >route { > > > > # > ----------------------------------------------------------------- > > # Sanity Check Section > > # > ----------------------------------------------------------------- > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483", "Too Many Hops"); > > break; > > }; > > > > if (msg:len > max_len) { > > sl_send_reply("513", "Message Overflow"); > > break; > > }; > > > > > #------------------------------------------------------------------ > > # NOTIFY Keep Alive Section > > > #------------------------------------------------------------------ > > if ((method=="NOTIFY") && search("^Event: keep-alive")){ > > sl_send_reply("200", "OK"); > > break; > > }; > > > > > #------------------------------------------------------------------ > > # Speed Dialing Section > > > #------------------------------------------------------------------ > > if ((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*")) { > > sd_lookup("speed_dial"); > > }; > > > > > #------------------------------------------------------------------ > > # Do not Disturb Section > > > #------------------------------------------------------------------ > > #if(avp_db_load("$ruri/username", "s:donotdisturb")){ > > # if(avp_check("s:donotdisturb", "eq/y/i")){ > > # route(x) #whereever asterisk voicemail is > > # break; > > # }; > > #}; > > > > # > ----------------------------------------------------------------- > > # Record Route Section > > # > ----------------------------------------------------------------- > > if (method!="REGISTER") { > > record_route(); > > }; > > > > if (method=="BYE" || method=="CANCEL") { > > unforce_rtp_proxy(); > > }; > > > > if (method=="INVITE"){ > > if(is_user_in("Request-URI", "voicemail")){ > > setflag(31); > > }; > > }; > > > > # > >---------------------------------------------------------------------- > >-- > > # URI Compare Section > > # > > # Here we compare the "from" and "to" to see if the > caller is > >dialing > > # their own extension. If so then we route to voicemail if > >needed > > # > >---------------------------------------------------------------------- > >-- > > if (method=="INVITE") { > > avp_write("$from", "i:34"); > > if (avp_check("i:34", "eq/$ruri/i")) { > > > > if (isflagset(31)) { > > route(8); > > break; > > } else { > > sl_send_reply("486", "Busy"); > > break; > > }; > > }; > > }; > > > > # > ----------------------------------------------------------------- > > # Loose Route Section > > # > ----------------------------------------------------------------- > > if (loose_route()) { > > > > if (has_totag() && (method=="INVITE" || > method=="ACK")) > { > > if (nat_uac_test("19")) { > > setflag(6); > > force_rport(); > > fix_nated_contact(); > > }; > > force_rtp_proxy("l"); > > }; > > route(1); > > break; > > }; > > > > # > ----------------------------------------------------------------- > > # Call Type Processing Section > > # > ----------------------------------------------------------------- > > > > if (uri!=myself) { > > route(4); > > route(1); > > break; > > }; > > > > if (method=="CANCEL") { > > route(1); > > break; > > } else if (method=="INVITE") { > > if(!cpl_run_script("incoming", "is_stateless")) > > { > > # script execution failed > > t_reply("500", "CPL script execution failed"); > > }; > > route(3); > > break; > > } else if (method=="REGISTER") { > > #handle REGISTER messages with CPL script > > cpl_process_register(); > > route(2); > > break; > > }; > > > > lookup("aliases"); > > if (uri!=myself) { > > route(4); > > route(1); > > break; > > }; > > > > if (!lookup("location")) { > > > > if(isflagset(31)){ > > setflag(19); > > }; > > > > #sl_send_reply("404", "User Not Found"); > > #break; > > }; > > > > route(1); > >} > > > >route[1] { > > > > # > ----------------------------------------------------------------- > > # Default Message Handler > > # > ----------------------------------------------------------------- > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > if (method=="INVITE" && isflagset(6)) { > > unforce_rtp_proxy(); > > }; > > sl_reply_error(); > > }; > >} > > > >route[2] { > > > > # > ----------------------------------------------------------------- > > # REGISTER Message Handler > > # > ---------------------------------------------------------------- > > > > if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { > > setflag(6); > > fix_nated_register(); > > force_rport(); > > }; > > > > sl_send_reply("100", "Trying"); > > > > if (!www_authorize("","subscriber")) { > > www_challenge("","0"); > > break; > > }; > > > > if (!check_to()) { > > sl_send_reply("401", "Unauthorized"); > > break; > > }; > > > > consume_credentials(); > > > > if (!save("location")) { > > sl_reply_error(); > > }; > >} > > > >route[3] { > > > > # > ----------------------------------------------------------------- > > # INVITE Message Handler > > # > ----------------------------------------------------------------- > > > > if (!proxy_authorize("","subscriber")) { > > proxy_challenge("","0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > > > #consume_credentials(); > > > > if (nat_uac_test("19")) { > > setflag(6); > > } > > > > lookup("aliases"); > > if (uri!=myself) { > > route(4); > > route(1); > > break; > > }; > > > > #Blind Call Forwarding > > if (avp_db_load("$ruri/username", "s:callfwd")){ > > setflag(22); > > avp_pushto("$ruri", "s:callfwd"); > > route(6); > > break; > > }; > > > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); > > break; > > }; > > > > if (avp_db_load("$ruri/username", "s:fwdbusy")){ > > if(!avp_check("s:fwdbusy", "eq/$ruri/i")){ > > setflag(26); > > }; > > }; > > > > if (avp_db_load("$ruri/username", "s:fwdnoanswer")){ > > if(!avp_check("s:fwdnoanswer", "eq/$ruri/i")){ > > setflag(27); > > }; > > }; > > > > t_on_failure("1"); > > > > route(4); > > route(1); > >} > > > >route[4] { > > > > # > ----------------------------------------------------------------- > > # NAT Traversal Section > > # > ----------------------------------------------------------------- > > > > if (isflagset(6)) { > > force_rport(); > > fix_nated_contact(); > > force_rtp_proxy(); > > }; > >} > > > >route[5]{ > > > > > #----------------------------------------------------------------- > > # PSTN Handler > > > #----------------------------------------------------------------- > > > > revert_uri(); > > rewritehostport(" x.x.x.x:5064"); > > append_branch(); > > t_relay_to_udp("x.x.x.x", "5064"); > > break(); > > > > t_on_failure("1"); > > > > route(4); > > route(1); > >} > > > >route[6]{ > > > > > #------------------------------------------------------------------ > > # Call Forwarding Handler > > # > > # This must be done as a route block because sl_send_reply() > cannot > > # be called from the failure_route block > > > #------------------------------------------------------------------- > > > > if(uri=~"^sip:1[0-9][10]@"){ > > strip(1); > > }; > > > > lookup ("aliases"); > > > > if(uri!=myself){ > > if(!isflagset(22)){ > > append_branch(); > > }; > > > > route(4); > > route(1); > > break; > > }; > > > > if(uri=~"^sip:011[0-9]*@"){ > > route(4); > > route(5); > > break; > > }; > > > > if(!lookup("location")){ > > if (uri=~"^sip:[0-9][10]@"){ > > route(4); > > route(1); > > break; > > }; > > sl_send_reply("404", "User Not Found"); > > }; > > > > route(4); > > route(1); > >} > > > >route[8] { > > > > # voicemail route #2 > > # > > # this path this executed during these conditions: > > # > > # cond 1) the called number is not in the location > table > > # cond 2) the from_uri == to_uri (ie, caller==callee) > > > > if (method == "INVITE" || method == "ACK"){ > > force_rtp_proxy(); > > }; > > > > rewritehostport("x.x.x.x:5064"); > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > > > if(method == "INVITE" || method == "ACK"){ > > unforce_rtp_proxy(); > > }; > > > > sl_reply_error(); > > }; > >} > > > >route[9]{ > > > > #voicemail route 1 > > # > > #this path is executed during these conditions: > > # > > # cond 1) the called number is in the location table > > # but the callee did not answer the phone ( i.e. > failover > to > >voicemail) > > > > rewritehostport("x.x.x.x:5064"); > > append_branch(); > > > > t_on_reply("1"); > > > > if (!t_relay()){ > > if(method == "INVITE" || method=="ACK"){ > > unforce_rtp_proxy(); > > }; > > sl_reply_error(); > > }; > >} > > > >onreply_route[1] { > > > > if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { > > if (!search("^Content-Length:[ ]*0")) { > > force_rtp_proxy(); > > }; > > }; > > > > if (nat_uac_test("1")) { > > fix_nated_contact(); > > }; > >} > > > >failure_route[1]{ > > > > if(t_check_status("487")){ > > break; > > }; > > > > if(isflagset(26) && t_check_status("486")){ > > if(avp_pushto("$ruri", "s:fwdbusy")){ > > avp_delete("s:fwdbusy"); > > resetflag(26); > > append_branch(); > > route(6); > > break; > > }; > > }; > > > > #Here we can have either voicemail OR forward no answer > > #forward on no answer is flag 27 > > #voicemail is flag 31 > > > > if(isflagset(27) && t_check_status("408")){ > > if(avp_pushto("$ruri", "s:fwdnoanswer")){ > > avp_delete("s:fwdnoanswer"); > > resetflag(27); > > append_branch; > > route(6); > > break; > > }; > > } > > else if(isflagset(31) && avp_pushto("$ruri", "$voicemail")){ > > avp_delete("$voicemail"); > > route(9); > > break; > > }; > >} > > > > > > > >-------------------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. > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > >http://lists.iptel.org/mailman/listinfo/serusers > > > >. > > > > > > > > -------------------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. > > > -------------------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. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > <http://lists.iptel.org/mailman/listinfo/serusers> > >

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[Serusers] Does VOVIDA B2BUA works with SER+MEDIAPROXY?
by Thomas Tomter 26 Sep '05

26 Sep '05

Hello, We are now trying to integrate SER+Vovida B2BUA with Alepo RBS version 4.2 and I was wondering whether you succeeded in making it work? Would it be possible for you to share with us how you configured SER+ B2UBA in order to make this work? We are particularly interested in making SER+B2BUA act as a Cisco device towards the Alepo billing system. Thanks in advance. Best regards, Thomas Tomter

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[Serusers] Field STATE
by Alberto 26 Sep '05

26 Sep '05

Hi, Where can I get information by field STATE? When I execute the command "serctl alias show", I can see the field STATE and I have watched the values: CS_SYNC, CS_DIRTY and CS_NEW, which is the meaning of these values? Thanks, -- Alberto

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[Serusers] cisco 7905G
by Alberto 26 Sep '05

26 Sep '05

Hi, Somebody has experience with cisco7905G? The 7905G is registered OK and can receive calls but not send call. Thanks, -- Alberto

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RE: [Serusers] allow_trusted removed from permission module (HEAD)?
by Bjorn Asmul 26 Sep '05

26 Sep '05

The README should be updated then. Unless I'm really off here, the allow_trusted-section has been taken out. Bjorn -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Monday, September 26, 2005 2:03 AM To: Bjorn Asmul Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] allow_trusted removed from permission module (HEAD)? No, it is still there. Jan. On 25-09-2005 17:55, Bjorn Asmul wrote: > It looks like "allow_trusted" was removed from the permission module > in CVS HEAD. > Can someone confirm this? Or was it moved to a different module? > > Bjorn > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Alias routing section in http://lists.iptel.org/pipermail/serusers/2005-February/015700.html
by Ramin Nikaeen 26 Sep '05

26 Sep '05

I have looked at the sample ser.cfg: http://lists.iptel.org/pipermail/serusers/2005-February/015700.html Can someone tell me what the following section in this file is for: # ------------------------------------------------------------------------ # Alias Routing Section # ------------------------------------------------------------------------ lookup("aliases"); if (!uri==myself) { route(2); break; }; thanks ramin

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[Serusers] SER Vs. SIPsak = 483, incoerence!!!!
by Ángel Gómez 26 Sep '05

26 Sep '05

I´m trying to test my SER server, but... wen i use sipsak to test it, with this comand: sipsak -s sip:ninguna@10.22.1.249 -vvv -f mensage.txt I got this output with sipsak: fqdnhostname: local our Via-Line: Via: SIP/2.0/UDP local:2288;rport New message with Via-Line: MESSAGE sip:prueba@10.22.1.249 SIP/2.0 Via: SIP/2.0/UDP local:2288;rport From: <sip:prueba@10.22.1.249> To: <sip:ninguna@10.22.1.249> Call-ID: 917549908@local CSeq: 2 MESSAGE Content-Type: text/plain Content-Length: 48 Max-Forwards: 7 User-Agent: sipsak 0.8.9_pre usrloc test message from SIPsak for user ninguna0.sending message to ninguna... ** request ** MESSAGE sip:prueba@10.22.1.249 SIP/2.0 Via: SIP/2.0/UDP local:2288;rport From: <sip:prueba@10.22.1.249> To: <sip:ninguna@10.22.1.249> Call-ID: 917549908@local CSeq: 2 MESSAGE Content-Type: text/plain Content-Length: 48 Max-Forwards: 7 User-Agent: sipsak 0.8.9_pre usrloc test message from SIPsak for user ninguna0.sending message to ninguna... message received: SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP local:2288;received=10.22.1.5;rport=2288 From: <sip:prueba@10.22.1.249> To: <sip:ninguna@10.22.1.249>;tag=b27e1a1d33761e85846fc98f5f3a7e58.191f Call-ID: 917549908@local CSeq: 2 MESSAGE Server: Sip EXpress router (0.9.3 (i386/linux)) Content-Length: 0 Warning: 392 10.22.1.249:5060 "Noisy feedback tells: pid=1649 req_src_ip=10.22. 1.249 req_src_port=5060 in_uri=sip:prueba@10.22.1.249 out_uri=sip:prueba@10.22.1 .249 via_cnt==8" ** reply received after 0.000 ms ** SIP/2.0 483 Too Many Hops final received And the incoerence is because in ser.cfg I have THIS line (I think): if (!mf_process_maxfwd_header("16")) { sl_send_reply("483","Too Many Hops"); break; }; the mensage isn´t looping (or i thing so), i have alias in the config file and both user of the mensage (from and to) are registered in the server. Can some one help me???

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[Serusers] Seeking Advise With Caller ID Block
by Chris Tsai 26 Sep '05

26 Sep '05

hi, I have a question about blocking caller id at SER (public proxy) if I want to blocking my caller id to invite other user at IPtel.org How do I do ? How about the INVITE message format needs? I try it the messae as follows: but it's fail.........>< Request-Line: INVITE sip:8201@iptel.org SIP/2.0 Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-hrs-21193 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org> Call-ID: gnA-11414(a)61.216.39.201 CSeq: 8 INVITE Contact: <sip:christsai12@61.216.39.201> Max-Forwards: 70 Remote-Party-ID: Chris <sip:christsai12@iptel.org>;privacy=full;party=calling supported: replaces User-Agent: DrayTek UA-1.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 Message body Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-hrs-21193;rport=5060;received=61.216.35.30 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.8aa7 Call-ID: gnA-11414(a)61.216.39.201 CSeq: 8 INVITE P-Behind-NAT: Yes Proxy-Authenticate: Digest realm="iptel.org", nonce="42bf93c8d6bc77f7ef3e6e8f24513796fda5bccc" Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Content-Length: 0 Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=30610 req_src_ip=61.216.35.30 req_src_port=5060 in_uri=sip:8201@iptel.org out_uri=sip:8201@iptel.org via_cnt==1" Session Initiation Protocol Request-Line: ACK sip:8201@iptel.org SIP/2.0 Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-hrs-21193 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.8aa7 Call-ID: gnA-11414(a)61.216.39.201 CSeq: 8 ACK Content-Length: 0 Session Initiation Protocol Request-Line: INVITE sip:8201@iptel.org SIP/2.0 Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-mWf-3372 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org> Call-ID: gnA-11414(a)61.216.39.201 CSeq: 9 INVITE Contact: <sip:christsai12@61.216.39.201> Proxy-Authorization: Digest username="christsai12", realm="iptel.org", nonce="42bf93c8d6bc77f7ef3e6e8f24513796fda5bccc", uri="sip:8201@iptel.org", response="d1cb74bf6a92df9edf18e565ee2eac79", algorithm=MD5 Max-Forwards: 70 Remote-Party-ID: Chris <sip:christsai12@iptel.org>;privacy=full;party=calling supported: replaces User-Agent: DrayTek UA-1.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 Message body Session Initiation Protocol Status-Line: SIP/2.0 403 That is ugly -- use From=id next time (OB) Message Header Via: SIP/2.0/UDP 61.216.39.201:5060;branch=z9hG4bK-mWf-3372;rport=5060;received=61.216.35.30 From: Anonymous <sip:anonymous@iptel.org>;tag=DBb-18789 To: <sip:8201@iptel.org>;tag=b51ece8fd8c195776737473dd6552d43.b29b Call-ID: gnA-11414(a)61.216.39.201 CSeq: 9 INVITE P-Behind-NAT: Yes Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Content-Length: 0 Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=30606 req_src_ip=61.216.35.30 req_src_port=5060 in_uri=sip:8201@iptel.org out_uri=sip:8201@iptel.org via_cnt==1"

2 1

Re: [Users] rewriteuser
by Nelson Silva 26 Sep '05

26 Sep '05

1 0

[Serusers] is SER ( 0.9.4) 100% 3GPP SIP call flow compliant ?
by Ashish Gilotra 26 Sep '05

26 Sep '05

Hi We are evaluating SER for our purpose. We need to use 3GPP SIP call flows. Is SER compliant ? Thanks Ash

1 0

Re: [Users] rewriteuser
by Daniel-Constantin Mierla 26 Sep '05

26 Sep '05

Hello, you can implement this with avpops module (http://www.voice-system.ro/docs/avpops/). Functions of interest for you are avp_write() and avp_pushto(). ... avp_write("1234", "s:client_n"); alias_db_lookup("dbaliases"); lookup("location); avp_pushto("$ruri/username", "s:client_n"); t_relay(); ... For optimization reasons, you may consider replacing the name of avp "s:client_n" to an integer id, e.g., "i:500" (faster operations). Cheers, Daniel On 09/26/05 14:14, Nelson Silva wrote: > Have been using openser for a couple of months and i run into a problem. > > Currently i'm using alias_db_lookup, location and t_relay to do the > routing. > > Since many clients have many numbers, i need to to rewrite the contact > to the right client. It's doing it but only for a client with one > number. Need for clients with more numbers. > > This is what i want: > > > $client_n = 1234 > > alias_db_lookup("db_aliases"); -> client_number(a)provider.com > > location("location"); -> client_number@client_ip > > rewriteuser($client_n); > > t_relay(); > > This is what i have: > > alias_db_lookup("dbaliases"); > > lookup("location); > > t_relay(); > > > > Anyone have this implemented? > > > Nelson Silva > ----------------- > email: nelson.silva(a)neuvex.com > website: http://www.neuvex.com > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Users] rewriteuser
by Nelson Silva 26 Sep '05

26 Sep '05

1 0

[Users] nathelper force_rtp_proxy onreply_route
by OpenSer Users 26 Sep '05

26 Sep '05

hello, in SDP body sent out by the proxy the original media IP address is still present, the rtpproxy ip is added as a another line. is this the correct behavior ? should I add some flags to remove the original media IP ? for the invite this does not happen, the c= field is REWRITTEN with the ip received from the rtpproxy. message received by openser: ============================================ v=0. o=CiscoSystemsSIP-GW-UserAgent 4176 788 IN IP4 GW-IP-ADDRESS. s=SIP Call. c=IN IP4 GW-IP-ADDRESS. t=0 0. m=audio 17644 RTP/AVP 18. c=IN IP4 GW-IP-ADDRESS. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=direction:passive. ============================================ message sent out by openser: ============================================ v=0. o=CiscoSystemsSIP-GW-UserAgent 4176 788 IN IP4 GW-IP-ADDRESS. s=SIP Call. c=IN IP4 GW-IP-ADDRESS. t=0 0. m=audio 35784 RTP/AVP 18. c=IN IP4 RTPPROXY-IP-ADDRESS. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=direction:passive. a=nortpproxy:yes. ============================================ onreply_route block ============================================ if(status =~ "(183)|2[0-9][0-9]") { if(!force_rtp_proxy()) { log(3, "force_rtp_proxy FAILED"); }; }; ============================================ thanks Razvan Radu

2 1

[Users] one way audio problem with stun
by Diego Valencia 26 Sep '05

26 Sep '05

Hi everybody, I have a "one way audio" problem. My enviroment is: xx.xx.63.75/27 = SER xx.xx.63.68/27 = Natbox (linux iptables masquerade) xx.xx.70.2/25 = STUN server xx.xx.63.78/27 = PAP1 linksys AOR 2000 10.1.1.233/24 = PAP2 linksys AOR 2001 (behind NATbox with stun enable) I have no audio from PAP1 to PAP2 when the call is made from PAP1 to PAP2, but when the call is made from PAP2 to PAP1 it works correctly. I see on the natbox that the RTP packet is droped with unreachable cause (icmp messages). This is the trace (ngrep) for the call from PAP1 to PAP2 on external interface of the natbox, where I can see STUN packets and the number udp port that the PAP willl be use (16444 to 16447). The problem is natbox use other port (1173). I don't know if this problem is on natbox or stun server or PAP. Can you helpme? Thanks!! U xx.xx.63.75:5060 -> xx.xx.63.68:5060 INVITE sip:2001@xx.xx.63.68:5060 SIP/2.0..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr=on>..Via: SIP/2.0/UDP xx.xx.63.75;branch=z9hG4bK52fb.41985863.0..Via: SIP/2.0/UDP xx.xx.63.78:5060;branch=z9hG4bK-57b1577f..From: 2000 <sip:5 890000(a)xx.xx.63.75>;tag=d1ccd3db9e31eb1fo0..To: <sip:2001@xx.xx.63.75>..Call-ID: d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 INVITE..Max-Forwards: 16..Contact: 2000 <sip:2000@xx.xx.63.78:5060>..Expires: 240..User-Agent: Linksys/PAP2-2.0.10(L Sc)..Content-Length: 420..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: app lication/sdp....v=0..o=- 29375 29375 IN IP4 xx.xx.63.78..s=-..c=IN IP4 xx.xx.63.78..t=0 0..m=audio 16472 RTP/AVP 18 0 2 4 8 9 6 97 98 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:8 PCMA/8 000..a=rtpmap:96 G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpmap:100 NSE/8000..a=rtpmap:101 telephon e-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. # U xx.xx.63.68:5060 -> xx.xx.63.75:5060 SIP/2.0 100 Trying..To: <sip:2001@xx.xx.63.75>..From: 2000 <sip:2000@xx.xx.63.75>;tag=d1ccd3db9e31eb1fo0..Call-ID : d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 INVITE..Via: SIP/2.0/UDP xx.xx.63.75;branch=z9hG4bK52fb.41985863.0..Via: SIP/2.0/U DP xx.xx.63.78:5060;branch=z9hG4bK-57b1577f..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr=on>..Server: Linksys/P AP2-2.0.10(LSc)..Content-Length: 0.... # U xx.xx.63.68:5060 -> xx.xx.70.2:3478 .....*f..*f..*f..*f. # U xx.xx.63.68:16444 -> xx.xx.70.2:3478 .....OB..OB..OB..OB. # U xx.xx.63.68:16445 -> xx.xx.70.2:3478 .................... # U xx.xx.63.68:16446 -> xx.xx.70.2:3478 .......&...&...&...& # U xx.xx.63.68:16447 -> xx.xx.70.2:3478 ....#...#...#...#... # U xx.xx.70.2:3478 -> xx.xx.63.68:16447 ...$#...#...#...#.........@?.+?D.........+F..........+F. # U xx.xx.63.68:5060 -> xx.xx.70.2:3478 .....*f..*f..*f..*f. # U xx.xx.63.68:16444 -> xx.xx.70.2:3478 .....OB..OB..OB..OB. # U xx.xx.70.2:3478 -> xx.xx.63.68:5060 ...$.*f..*f..*f..*f..........+?D.........+F..........+F. # U xx.xx.63.68:16445 -> xx.xx.70.2:3478 .................... # U xx.xx.63.68:16446 -> xx.xx.70.2:3478 .......&...&...&...& # U xx.xx.70.2:3478 -> xx.xx.63.68:16444 ...$.OB..OB..OB..OB.......@<.+?D.........+F..........+F. # U xx.xx.70.2:3478 -> xx.xx.63.68:16446 ...$...&...&...&...&......@>.+?D.........+F..........+F. # U xx.xx.63.68:5060 -> xx.xx.63.75:5060 SIP/2.0 180 Ringing..To: <sip:2001@xx.xx.63.75>;tag=156298945b7d27i0..From: 2000 <sip:2000@xx.xx.63.75>;tag=d1ccd 3db9e31eb1fo0..Call-ID: d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 INVITE..Via: SIP/2.0/UDP xx.xx.63.75;branch=z9hG4bK52fb.4198 5863.0..Via: SIP/2.0/UDP xx.xx.63.78:5060;branch=z9hG4bK-57b1577f..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr= on>..Server: Linksys/PAP2-2.0.10(LSc)..Content-Length: 0.... # U xx.xx.70.2:3478 -> xx.xx.63.68:16444 ...$.OB..OB..OB..OB.......@<.+?D.........+F..........+F. # U xx.xx.63.68:5060 -> xx.xx.63.75:5060 SIP/2.0 200 OK..To: <sip:2001@xx.xx.63.75>;tag=156298945b7d27i0..From: 2000 <sip:2000@xx.xx.63.75>;tag=d1ccd3db9e 31eb1fo0..Call-ID: d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 INVITE..Via: SIP/2.0/UDP xx.xx.63.75;branch=z9hG4bK52fb.41985863. 0..Via: SIP/2.0/UDP xx.xx.63.78:5060;branch=z9hG4bK-57b1577f..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr=on>.. Contact: 2001 <sip:2001@xx.xx.63.68:5060>..Server: Linksys/PAP2-2.0.10(LSc)..Content-Length: 236..Allow: ACK, BYE, CANCE L, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: application/sdp....v=0..o=- 659101 659101 IN IP4 20 0.43.63.68..s=-..c=IN IP4 xx.xx.63.68..t=0 0..m=audio 16444 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100 NSE/8000. .a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. # U xx.xx.63.75:5060 -> xx.xx.63.68:5060 ACK sip:2001@xx.xx.63.68:5060 SIP/2.0..Record-Route: <sip:xx.xx.63.75;ftag=d1ccd3db9e31eb1fo0;lr=on>..Via: SIP/2.0/UDP 200 .43.63.75;branch=0..Via: SIP/2.0/UDP xx.xx.63.78:5060;branch=z9hG4bK-5a4c9c84..From: 2000 <sip:2000@xx.xx.63.75>;tag =d1ccd3db9e31eb1fo0..To: <sip:2001@xx.xx.63.75>;tag=156298945b7d27i0..Call-ID: d6e2df8b-5193d32f@xx.xx.63.78..CSeq: 101 AC K..Max-Forwards: 16..Contact: 2000 <sip:2000@xx.xx.63.78:5060>..User-Agent: Linksys/PAP2-2.0.10(LSc)..Content-Length: 0.... # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8...-..L....@.....tF.34..V.|B..n..a.k(.R # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8...-..L....@.....tF.34..V.|B..n..a.k(.R # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!o..vy.F.!...|...&.T.BP..Z..n...:.3..d.. # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8......L... .s..._exR..Kc.....+.O<.RZ.O{ # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8......L... .s..._exR..Kc.....+.O<.RZ.O{ # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!p..wi.F.!.'D.F...K....@C..e.Z.[~P...Et, # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8.../..L......@H..._...,[.P...rU.A..P... # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8.../..L......@H..._...,[.P...rU.A..P... # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!q..xY.F.!....Mf.....Fu.Q.....~.}....v.. # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8...0..L..\.~..h..+..|S.C.M..~6W}....6sR # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8...0..L..\.~..h..+..|S.C.M..~6W}....6sR # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!r..yI.F.!x|..Eo..._.1...;..GRz....*.x.Z # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8...1..L..x.....n."..6.v....S.D.......S. # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2.!..8...1..L..x.....n."..6.v....S.D.......S. # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!s..z9.F.!0...ooPE..F. .5...E....v...... # U xx.xx.63.78:16472 -> xx.xx.63.68:16444 ..8...2..L..z....+.b..r?...r....|..$.x...^ # I xx.xx.63.68 -> xx.xx.63.78 3:3 ....E..F.........+?N.+?D@X@<.2....8...2..L..z....+.b..r?...r....|..$.x...^ # U xx.xx.63.68:1173 -> xx.xx.63.78:16472 ..!t..{).F.!.o...{..r..3....F.$.r.. V..by. #

2 1

[Serusers] Problem/Question regarding mediaproxy
by Ronald Voermans 26 Sep '05

26 Sep '05

Hello, I'm using Ser in combination with Mediaproxy. The situation is as follow: My SER has a private ip address (10.254.254.1), and my UAs also have private IP addresses, they can all see SER, and SER can see the UAs. Next, our company has a Voice Interconnect with a telco. (it's using Cirpack as softswitch). SER can see Cirpack and vice versa. However, the UAs cannot see the Cirpack. So I want to use Mediaproxy to overcome this. I can make outbound calls, but when calling inbound, I sometimes have one-way-voice of no voice at all. I think the problem is related to Cirpack, but I have a question thouh: - When calling mediaproxy, how does mediaproxy decide which one is the caller and which one is the called? Is this based on the From and To header? Because, if i call from PSTN -> SIP, i get serveral thousands (!) of messages saying that the called person signed in (no caller signs in in mediaproxy). This could be due to the fact that the from and to headers are incorrect. My ser.cfg is the one from onsip.org (pstn with nat). I can post configs/traces if someone needs them! Regards, Ronald

1 0

[Serusers] allow_trusted removed from permission module (HEAD)?
by Bjorn Asmul 26 Sep '05

26 Sep '05

It looks like "allow_trusted" was removed from the permission module in CVS HEAD. Can someone confirm this? Or was it moved to a different module? Bjorn

2 1

[Serusers] SER + Mysql.so problem
by smy＠addm.com 26 Sep '05

26 Sep '05

Hi all, I've been working with SER + Mysql for over a month without any problems, yet all these (including the DB) are installed on the same server (LocalHost). I'm currently trying to centralize and securing all Data, thus would like to move all data to a centralized DB offsite. Now, I have SER DB migrated to this new DB server, and have changed the SER.cfg so that all mysql related loadmodule would be pointing to the proper server (ip address) instead of localhost. This is where the nightmare begins. The SER server can not seem to be connecting to the Mysql Server properly. Thus it took me over 3 days to debug and see if there were any auth problems from a remote site to the Mysql server. Apparently, remote auth & connect to the DB is working perfectly since I've installed other application which can connect to this DB server. So I've checked with the SER log + ngrep, of which to my amazement that the SER never sent any signals to the DB server. As soon as I moved the DB back to localhost, everything worked! My question is, is there something I need to configure in the mysql module so that it can work with a remote server???? Any insight is greatly appreciated!!! Best Regards, Samuel Yeung

3 7

[Serusers] accounting and radius/sql
by Bjorn Asmul 25 Sep '05

25 Sep '05

The best solution is to use two different tables for start and stop accounting. Have a look in the sql.conf file in FreeRADIUS Problem is that SER replaces the Username in the stop-message, with whoever hung up... Therefore the default queries in FreeRADIUS won't find the corresponding record. To get the CDR just join the two tables on AcctSessionId. I have experienced problems joining on the AcctSessionId in MySQL versions prior to 4.1, for some reason. Bjorn

1 0

[Serusers] More random avpops questions....
by sip 25 Sep '05

25 Sep '05

Okay... I think with Steve's help, I've been getting somewhere with my avpops stuff, but I have a couple of questions. 1) What is the format of the $from and $to headers as far as avpops is concerned? How can I find that out? 2) How does avp_print() work (i.e. to where does it print) ? Is there any way I can get intermediate values of things so I can see just what it is my little scripts are doing? N.

3 4

[Serusers] [about SER]networking among three SIP servers.....
by 송 미진 25 Sep '05

25 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear........ I want to networking among three SIP servers. But now I am trying using two servers likes three servers. That is, - source : computer A ip address ; 192.168.178.194 sipuri : sip:mj@192.168.178.194 - via : computer B ip address : 192.168.178.193 sipuri : sip:hj@192.168.178.193 - destination : computer A ip address :192.168.178.194 sipuri : sip:mj@192.168.178.194 And I am using sipp to test (I modified the "via" part based on original uac module). ===================================================== << computer B's test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="kims.myserver.sampro.cse" alias="myserver.sampro.cse" alias="192.168.178.193" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes 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/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\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; }; if (msg:len>=204822) { 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=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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(uri=="sip:mijin@sam.cse"){ # forward(192.168.178.194, 5060); # }; if (uri=~"sip:192.168.178.193"){ if (method=="REGISTER") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save ì„±ê³µ\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3ìœ¼ë¡œ ì�´ë�™ \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DBì—�ì„œ ì°¾ ì�„ ìˆ˜ ì—†ì�Œ \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:hjkim@myserver.sampro.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.193" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 ë�� \n"); } As you see, I added modparam("rr", "enable_full_lr", no) if(uri=="sip:mijin@192.168.178.194"){ forward(192.168.178.194, 5060); append_branch("sip:hjkim@192.168.178.193:5060"); log(1, "next hop...\n"); break; }; but, I received error message in computer B(via): 8(12253) ERROR: forward_reply: no 2nd via found in reply What does it mean? What can I do for doing what I want ??? Is there any problem in my test.cfg file ????? I would like to know in detail . please tell me what to do............................... sipp module can't support that???? And Please tell me about forward funtion. I don't know whether I understand forward function usage well....... sorry ............ Song _________________________________________________________________ 고.. 감.. 도.. 사.. 랑.. 만.. 들.. 기.. MSN 러브 http://www.msn.co.kr/love/

1 0

[Serusers] Radius Authorization and AVP Retrieval
by Tavis P 24 Sep '05

24 Sep '05

I'm having some trouble trying to integrate both Radius authentication and Radius AVP storage. I'm using freeradius and there doesn't seem to be much documentation regarding configuration semantics and such. Could anyone share some information on how they were able to use a Radius server to authenticate requests and (with avp_radius) as an AVP retrieval mechanism? I'm uncertain as to how I can stack these two uses of data on the freeradius server tavis

3 4

[Serusers] [about SER]networking among three SIP servers.....
by 송 미진 24 Sep '05

24 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear........ I want to networking among three SIP servers. But now I am trying using two servers likes three servers. That is, - source : computer A ip address ; 192.168.178.194 sipuri : sip:mj@192.168.178.194 - via : computer B ip address : 192.168.178.193 sipuri : sip:hj@192.168.178.193 - destination : computer A ip address :192.168.178.194 sipuri : sip:mj@192.168.178.194 And I am using sipp to test (I modified the "via" part based on original uac module). ===================================================== << computer B's test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="kims.myserver.sampro.cse" alias="myserver.sampro.cse" alias="192.168.178.193" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes 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/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\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; }; if (msg:len>=204822) { 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=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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(uri=="sip:mijin@sam.cse"){ # forward(192.168.178.194, 5060); # }; if (uri=~"sip:192.168.178.193"){ if (method=="REGISTER") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save ì„±ê³µ\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3ìœ¼ë¡œ ì´ë™ \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DBì—ì„œ ì°¾ ì„ ìˆ˜ ì—†ìŒ \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:hjkim@myserver.sampro.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.193" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 ë \n"); } As you see, I added modparam("rr", "enable_full_lr", no) if(uri=="sip:mijin@192.168.178.194"){ forward(192.168.178.194, 5060); append_branch("sip:hjkim@192.168.178.193:5060"); log(1, "next hop...\n"); break; }; but, I received error message in computer B(via): 8(12253) ERROR: forward_reply: no 2nd via found in reply What does it mean? What can I do for doing what I want ??? Is there any problem in my test.cfg file ????? I would like to know in detail . please tell me what to do............................... Song _________________________________________________________________ 싸게 싸게 MSN공동구매 http://www.waawaa.com/cobuy/cobuy_default.asp?siteid=10160

1 0

[Serusers] about SER
by 송 미진 24 Sep '05

24 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear. All Hi~ I was adviced many times by "seruser" but my problems has not been solved. I,m afraid that I have many questions. So I intend to specify my present situation. Operating System: Linux Fedora 3 ================================================================================ [root@cse3 ~]# serctl start Starting SER : started pid(8370) [root@cse3 ~]# serctl ps 0 8370 attendant 1 8375 fifo server 2 8377 receiver child=0 sock= 127.0.0.1:5060 3 8379 receiver child=1 sock= 127.0.0.1:5060 4 8384 receiver child=2 sock= 127.0.0.1:5060 5 8386 receiver child=3 sock= 127.0.0.1:5060 6 8387 receiver child=0 sock= 192.168.178.193:5060 7 8390 receiver child=1 sock= 192.168.178.193:5060 8 8393 receiver child=2 sock= 192.168.178.193:5060 9 8397 receiver child=3 sock= 192.168.178.193:5060 10 8401 timer 11 8403 tcp receiver 12 8406 tcp receiver 13 8410 tcp receiver 14 8412 tcp receiver 15 8415 tcp main process ============================================================================== my_ip: 192.168.178.193 sip_uri: sip:hjkim@192.168.178.193 I added my hostname and my ip address in /etc/host file. And I added alias="hostname.myserver.sampro.cse" alias="192.168.178.193" in /usr/local/etc/ser/ser.cfg file. ==================================== 1 ============================================ [root@cse3 ~]# serctl ping sip:hjkim@192.168.178.193 [1]+ Done cat <$path | filter_fl 483 =========================================================================================== ==================================== 2 =========================================== I received "SIP/2.0 483 Too Many Hops" . but I don't know the reason. Well...My computer is connected other computers including computer has 192.168.178.194 as ip address. And my /etc/host file has 127.0.0.1 cse4 localhost.localdomain localhost 192.168.178.194 kims myserver.sampro.cse Is there reason in here??? I don't know . =========================================================================================== [root@cse3 ~]# sipsak -T -s sip:hjkim@192.168.178.193 warning: IP extract from warning activated to be more informational 0: 192.168.178.193 (0.355 ms) SIP/2.0 483 Too Many Hops 1: 192.168.178.193 (0.982 ms) SIP/2.0 483 Too Many Hops 2: 192.168.178.193 (0.459 ms) SIP/2.0 483 Too Many Hops 3: 192.168.178.193 (0.520 ms) SIP/2.0 483 Too Many Hops .................... 177: timeout after 500 ms 177: timeout after 1000 ms 177: timeout after 2000 ms 177: 192.168.178.193 (3306.067 ms) SIP/2.0 483 Too Many Hops 178: 192.168.178.193 (3.073 ms) SIP/2.0 483 Too Many Hops 179: 192.168.178.193 (3.104 ms) SIP/2.0 483 Too Many Hops .................. 313: 192.168.178.193 (3.345 ms) SIP/2.0 483 Too Many Hops 314: 192.168.178.193 (3.494 ms) SIP/2.0 483 Too Many Hops 315: 192.168.178.193 (3.038 ms) SIP/2.0 483 Too Many Hops 316: 192.168.178.193 (3.518 ms) SIP/2.0 483 Too Many Hops 317: 192.168.178.193 (5.048 ms) SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) without Contact header ============================================================================================ ======================================== 3 ====================================== In response part, I found the message "SIP/2.0 401 Unauthorized" and "authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above" What happened?? ============================================================================================ [root@cse3 ~]# sipsak -U -vv -s sip:hjkim@192.168.178.193 warning: ignoring -i option when in usrloc mode registering user hjkim... authorizing registering user hjkim... request: REGISTER sip:192.168.178.193 SIP/2.0 Authorization: Digest username="hjkim", uri="sip:192.168.178.193", algorithm=MD5, realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87", response="f58e11aff01e5123bcf5010405e91278" Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193 Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.1 Expires: 15 Contact: sip:hjkim@127.0.0.1:32860 response: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport=32860;received=192.168.178.193 From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193;tag=b27e1a1d33761e85846fc98f5f3a7e58.ad3a Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER WWW-Authenticate: Digest realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87" Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.193:5060 "Noisy feedback tells: pid=8397 req_src_ip=192.168.178.193 req_src_port=32860 in_uri=sip:192.168.178.193 out_uri=sip:192.168.178.193 via_cnt==1" error: authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above ============================================================================================ [root@cse3 ~]# serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0xf50b44c8 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0xf50b2218 d_ll { n : 2 first: 0xf50b4220 last : 0xf50b4348 } ...Record(0xf50b4220)... domain: 'location' aor : 'kim' ~~~Contact(0xf50b4260)~~~ domain : 'location' aor : 'kim' Contact : 'sip:hjkim@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50b4348)... domain: 'location' aor : 'mj' ~~~Contact(0xf50b4388)~~~ domain : 'location' aor : 'mj' Contact : 'sip:mj@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ===/Domain list=== ==================================================================================================================================== 4 ===================================== I received the message "received ICMP packet (type: 3, code: 10): from 192.168.178.194" What does means this message?? ============================================================================================ [root@cse3 ~]# sipsak -T -vv -s sip:iptel@192.168.178.194 warning: IP extract from warning activated to be more informational Max-Forwards set to 0 received ICMP packet (type: 3, code: 10): from 192.168.178.194 ======================================================= ======================================================= << my 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) alias="192.168.178.193" alias="myserver.sampro.cse" alias="kims.myserver.sampro.cse" /* 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 >= 2000048 ) { 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=="sip:192.168.178.193") { # 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=="sip:192.168.178.193") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("myserver.sampro.cse", "subscriber")) { www_challenge("myserver.sampro.cse", "0"); break; }; save("location"); log(1, "save \n"); break; }; lookup("aliases"); if (!uri=="sip:192.168.178.193") { 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(); }; } ###################################################################################### I’d appreciate it if you would give me some solutions. Thank you in advance..... mi-jin Song _________________________________________________________________ 행운의 주인공이 이번엔 나일꺼야, 진짜루... 인터넷 복권 http://www.msn.co.kr/money/interlotto/

2 1

[Serusers] cannot receive
by rajesh volluru 24 Sep '05

24 Sep '05

hi I am using Sip communicator. I cant receive any voice from the other calling party but the calling party can hear whatever I talk. The ReceiveStreamEvent is not getting generated. Would anyone help me reg this please? Thanks Rajesh --------------------------------- Yahoo! India Matrimony: Find your partner now.

2 1

RE: [Serusers] Permanent aliases
by Igor Balk 23 Sep '05

23 Sep '05

Thank you guys for the clarification. If you do not mind I would ask one more question. Let say I have aliases 1234 of the account user@server and I'm getting call to the 1234 alias but there is no one registered at the moment with URI name@server. I also have SER configured in a way that if call will come to name@server URI and no user is availeble the call will be forwarded to voice mail box. Will same scenario work in case of call to the alias if I do lookup("aliases") before forwarding or I should do something more sofisticated to make it work? Thank you in advance, Igor -----Original Message----- From: Atle Samuelsen [mailto:clona@cyberhouse.no] Sent: Friday, September 23, 2005 1:14 PM To: Igor Balk Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Permanent aliases Hi Igor, I think your wrong there. AS far as you register with your box to your sipserver, and have a alias pointing to the username@yoursipserver then you should'nt have a problem. -Atle * Igor Balk <IBalk(a)intermedia.net> [050923 19:01]: > > Sorry if I'm asking stupid question but I'm trying to set up alias for > example 1234<->john.doe and make it permanent. As far as I understand > if I use serctl add alias command and then contact for the person will > change (dynamic ip, login from other coputer etc) my alias would not > work anymore. Please tell me if I'm worng and if there is a way to fix > this problem. > > Thank you, > Igor > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Serusers] Question regarding UDP TCP translation
by Weiler, Frank 23 Sep '05

23 Sep '05

Hi all, I'm new to SER and I started to play around with it as we need a solution to connect UPD UA's connected to Asterisk to a Microsoft LCS and we tough to use SER for this purpose. I've read so far through the Admin guide and it seems that the TCP support is natively embedded into SER. However I've added to the ser.cfg file the following lines (within the route[0] block just before branching to route [1]): #Forward all traffic with the uri*(a)172.16.0.201 <mailto:uri*@172.16.0.201> If (uri=~"sip:*@172.16.0.201) { forward_tcp(172.16.0.201,5060); Break; }; However while sniffing the packets it still appears that the INVITE requests are sent to the LCS using UDP and I receive an ICMP port unreachable. The original request for the UA is sent in UDP. I assume that I configured something wrong and I've checked the mailing lists and I didn't find any related documents. I'm using the latest version of SER available. Needless to say that UDP SIP calls between ASTERISK and SER work fine whatever clients I use. Any clue or help would be highly appreciated. Thank you very much in forecast for your help. Cheers, Frank

2 1

[Serusers] Getting starting with ser...
by IS2 - Florent BARBIER 23 Sep '05

23 Sep '05

Hello all, Not really easy ton understand how to setup... If somebody can help, I'll really appreciate. My config : redhat 7.3 (SME E/Smith edition) Asterisk 1.0.9 ser serweb IP phone 1 IP phone 2 IP public 1 IP public 2 ! ! ! Internet ! --------------------------------------------- ! ! ! IP public 3 ! ! Lan Asterisk ! ------------------------------------------------- ! ! IP Phone 3 IP Phone 4 IP Local 3 IP Local 4 Every IP Phone are registered in the AsterisK IP Phone 3 comunique with IP Phone 4 whithout PB IP Phone 1 comunique with IP Phone 3 or 4 whithout PB IP Phone 1 comunique with IP Phone 2 whith delay ! So audio stream doesn't go directly from IP Public 1 to IP Public 2. As I understand S.E.R. should help me to redirect audio stream... Well I setup S.E.R. and SERWEB under the Asterisk box (wich is the gateway & the firewall) I add some user whith serweb I add register in my sip.conf to register my asterisk to ser I try to do an echo test and nothing append. Well, how to setup SER to build this config... I'm sure this question certainly basic for most of you, but I've good Knowledge of AsterisK and I start with SIP proxies. I'm really not find F.A.Q. or userguide to build this... Thank you for your help, Florent.

2 1

[Serusers] about SER
by 송 미진 23 Sep '05

23 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear. All Hi~ I was adviced many times by "seruser" but my problems has not been solved. I,m afraid that I have many questions. So I intend to specify my present situation. Operating System: Linux Fedora 3 ================================================================================ [root@cse3 ~]# serctl start Starting SER : started pid(8370) [root@cse3 ~]# serctl ps 0 8370 attendant 1 8375 fifo server 2 8377 receiver child=0 sock= 127.0.0.1:5060 3 8379 receiver child=1 sock= 127.0.0.1:5060 4 8384 receiver child=2 sock= 127.0.0.1:5060 5 8386 receiver child=3 sock= 127.0.0.1:5060 6 8387 receiver child=0 sock= 192.168.178.193:5060 7 8390 receiver child=1 sock= 192.168.178.193:5060 8 8393 receiver child=2 sock= 192.168.178.193:5060 9 8397 receiver child=3 sock= 192.168.178.193:5060 10 8401 timer 11 8403 tcp receiver 12 8406 tcp receiver 13 8410 tcp receiver 14 8412 tcp receiver 15 8415 tcp main process ============================================================================== my_ip: 192.168.178.193 sip_uri: sip:hjkim@192.168.178.193 I added my hostname and my ip address in /etc/host file. And I added alias="hostname.myserver.sampro.cse" alias="192.168.178.193" in /usr/local/etc/ser/ser.cfg file. ==================================== 1 ============================================ [root@cse3 ~]# serctl ping sip:hjkim@192.168.178.193 [1]+ Done cat <$path | filter_fl 483 =========================================================================================== ==================================== 2 =========================================== I received "SIP/2.0 483 Too Many Hops" . but I don't know the reason. Well...My computer is connected other computers including computer has 192.168.178.194 as ip address. And my /etc/host file has 127.0.0.1 cse4 localhost.localdomain localhost 192.168.178.194 kims myserver.sampro.cse Is there reason in here??? I don't know . =========================================================================================== [root@cse3 ~]# sipsak -T -s sip:hjkim@192.168.178.193 warning: IP extract from warning activated to be more informational 0: 192.168.178.193 (0.355 ms) SIP/2.0 483 Too Many Hops 1: 192.168.178.193 (0.982 ms) SIP/2.0 483 Too Many Hops 2: 192.168.178.193 (0.459 ms) SIP/2.0 483 Too Many Hops 3: 192.168.178.193 (0.520 ms) SIP/2.0 483 Too Many Hops .................... 177: timeout after 500 ms 177: timeout after 1000 ms 177: timeout after 2000 ms 177: 192.168.178.193 (3306.067 ms) SIP/2.0 483 Too Many Hops 178: 192.168.178.193 (3.073 ms) SIP/2.0 483 Too Many Hops 179: 192.168.178.193 (3.104 ms) SIP/2.0 483 Too Many Hops .................. 313: 192.168.178.193 (3.345 ms) SIP/2.0 483 Too Many Hops 314: 192.168.178.193 (3.494 ms) SIP/2.0 483 Too Many Hops 315: 192.168.178.193 (3.038 ms) SIP/2.0 483 Too Many Hops 316: 192.168.178.193 (3.518 ms) SIP/2.0 483 Too Many Hops 317: 192.168.178.193 (5.048 ms) SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) without Contact header ============================================================================================ ======================================== 3 ====================================== In response part, I found the message "SIP/2.0 401 Unauthorized" and "authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above" What happened?? ============================================================================================ [root@cse3 ~]# sipsak -U -vv -s sip:hjkim@192.168.178.193 warning: ignoring -i option when in usrloc mode registering user hjkim... authorizing registering user hjkim... request: REGISTER sip:192.168.178.193 SIP/2.0 Authorization: Digest username="hjkim", uri="sip:192.168.178.193", algorithm=MD5, realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87", response="f58e11aff01e5123bcf5010405e91278" Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193 Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.1 Expires: 15 Contact: sip:hjkim@127.0.0.1:32860 response: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport=32860;received=192.168.178.193 From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193;tag=b27e1a1d33761e85846fc98f5f3a7e58.ad3a Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER WWW-Authenticate: Digest realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87" Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.193:5060 "Noisy feedback tells: pid=8397 req_src_ip=192.168.178.193 req_src_port=32860 in_uri=sip:192.168.178.193 out_uri=sip:192.168.178.193 via_cnt==1" error: authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above ============================================================================================ [root@cse3 ~]# serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0xf50b44c8 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0xf50b2218 d_ll { n : 2 first: 0xf50b4220 last : 0xf50b4348 } ...Record(0xf50b4220)... domain: 'location' aor : 'kim' ~~~Contact(0xf50b4260)~~~ domain : 'location' aor : 'kim' Contact : 'sip:hjkim@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50b4348)... domain: 'location' aor : 'mj' ~~~Contact(0xf50b4388)~~~ domain : 'location' aor : 'mj' Contact : 'sip:mj@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ===/Domain list=== ==================================================================================================================================== 4 ===================================== I received the message "received ICMP packet (type: 3, code: 10): from 192.168.178.194" What does means this message?? ============================================================================================ [root@cse3 ~]# sipsak -T -vv -s sip:iptel@192.168.178.194 warning: IP extract from warning activated to be more informational Max-Forwards set to 0 received ICMP packet (type: 3, code: 10): from 192.168.178.194 ###################################################################################### I’d appreciate it if you would give me some solutions. Thank you in advance..... mi-jin Song _________________________________________________________________ 행운의 주인공이 이번엔 나일꺼야, 진짜루... 인터넷 복권 http://www.msn.co.kr/money/interlotto/

2 1

[Serusers] Permanent aliases
by Igor Balk 23 Sep '05

23 Sep '05

Sorry if I'm asking stupid question but I'm trying to set up alias for example 1234<->john.doe and make it permanent. As far as I understand if I use serctl add alias command and then contact for the person will change (dynamic ip, login from other coputer etc) my alias would not work anymore. Please tell me if I'm worng and if there is a way to fix this problem. Thank you, Igor

3 2

[Serusers] MediaProxy Not Showing The Traffic
by Felipe Louback 23 Sep '05

23 Sep '05

I installed ser 8.14 and using mediaproxy 1.4.2. I am trying to use X-lite to make a call to the PSTN through a PSTN gateway (asterisk). The PSTN phone rings and get voice, but X-lite only pass voice. I can hear nothing. When I use the tool sessions.py available with mediaproxy, it shows me the call but shows that it is inactive, although the PSTN phone can hear audio. And after 60 seconds it drops the session, but the PSTN end still can hear the audio sent from X-lite. And sessions.py does not show the IP of asterisk.... [root@someplace mediaproxy]# ./sessions.py Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------------------------- 200.150.x.x:8000 - 127.0.0.1:35008 - ?.?.?.?:? inactive 0'10" Unknown Audio 0/0/0 If I run the rtpgenerator.py it shows the test rtp with traffic flowing. Here it is what I do in ser.cfg: if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!lookup("location")) { if ( ( uri=~"^sip:[0-9]{8}@.*" ) ) { t_relay_to_udp("X.X.X.X", "5060"); break; } sl_send_reply("404", "User not found"); break; }; If I turn off mediaproxy and disable the use_media_proxy, everything works fine. Any ideas? I read every possible document and found nothing about this. Thanks Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "For God so loved the world that he gave his one and only Son, that whoever believes in him shall not perish but have eternal life." John 3:16

2 2

RE: [Serusers] SIP/2.0 483 Too Many Hops
by Mike Williams 23 Sep '05

23 Sep '05

Here, try this script. I couldn't test it, but apart from minor errors it should work. # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias=server.sam.cse alias=sam.cse # debugging log_stderror=yes # (cmd line: -E) #mhomed=yes fifo="/tmp/ser_fifo" fifo_db_url="mysql://root:vlftmd@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:vlftmd@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { log(1, "ERROR: Too Many Hops\n"); sl_send_reply("483","Too Many Hops"); break; }; #keep this low to prevent possible DoS attacks and misconfigurations from overloading the server if (msg:len>=2048) { log(1, "ERROR: Message Too Big\n"); 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 (loose_route()) { route(1); break; }; if (method=="REGISTER") { log(1, "INFO: User Registered\n"); # save location before initiating welcome save("location"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "INFO: Initiating Welcome Message \n"); route(3); break; }; log(1, "record-route processing completed\n"); # 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=="INVITE") { log(1, "INFO: SIP Invite"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "ERROR: User Not Found\n"); break; }; }; record_route(); }; route(1); } route[1] { #send the call outward if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:mijin@sam.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.194" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "ERROR: Welcome Message Script Failed\n"); } log(1, "INFO: Welcome Message Sent\n"); } #----------------End Script--------------------- ---Mike -----Original Message----- From: serusers-bounces(a)lists.iptel.org on behalf of ? ?? Sent: Fri 9/23/2005 6:30 AM To: serusers(a)lists.iptel.org Subject: [Serusers] SIP/2.0 483 Too Many Hops confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear.... I inserted command like this. What I want is that when I loaded test.cfg file, what I check SIP messages using sipp client mode I want 200 OK. But, as you see, I received "SIP/2.0 483 Too Many Hops" error message. So I was advanced about this. - alias="sam.cse"//SIP_ DOMAIN alias="server.sam.cse"//my_computer.SIP_DOMAIN alias="192.168.178.194"//my_ip - add my hostname in /etc/host - change to : if (!mf_process_maxfwd_header("100") but, I couldn't solve this problem. I would really appreciate for giving some advise. Thank you in advance............................. mi-jin Song ===================================================== [root@cse4 ~]# ser -f test.cfg Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.178.194 [192.168.178.194]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.178.194 [192.168.178.194]:5060 Aliases: tcp: server:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: cse4:5060 udp: server:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: cse4:5060 *: 192.168.178.194:* *: sam.cse:* *: server.sam.cse:* stateless - initializing 0(0) Maxfwd module- initializing exec - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) ERROR: tcp_init: bind(6, 0x810dafc, 16) on 127.0.0.1: Address already in use ===================================================== [root@cse4 sipp]# ./sipp -sn uac 192.168.178.194 Resolving 192.168.178.194... ------------------------------ Scenario Screen -------- [1-4]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 5061 2.54 s 25 192.168.178.194:5060(UDP) 5 new calls during 0.538 s period 2 ms scheduler resolution 0 concurrent calls (limit 30) Peak was 1 calls, after 0 s 0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 25 0 0 100 <---------- 25 0 25 180 <---------- 0 0 0 200 <---------- E-RTD 0 0 0 ACK ----------> 0 0 [ 0 ms] BYE ----------> 0 0 0 200 <---------- 0 0 0 ------------------------------ Test Terminated --------------------------------- ----------------------------- Statistics Screen ------- [1-4]: Change Screen -- Start Time | 2005-09-23 20:03:13 Last Reset Time | 2005-09-23 20:03:15 Current Time | 2005-09-23 20:03:16 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:536 | 00:00:02:543 Call Rate | 9.328 cps | 9.831 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 5 | 25 Total Call created | | 25 Current Call | 0 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 0 Failed call | 6 | 25 -------------------------+---------------------------+-------------------------- Response Time | 00:00:00:000 | 00:00:00:000 Call Length | 00:00:00:004 | 00:00:00:004 ------------------------------ Test Terminated -------------------------------- 2005-09-23 20:03:16: Unexpected message for Call-ID '25.4743.127.0.0.1(a)sipp.call.id': while expecting '100' response, received 'SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 127.0.0.1:5061;received=192.168.178.194 From: sipp <sip:sipp@127.0.0.1:5061>;tag=25 To: sut <sip:service@192.168.178.194:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6e68 Call-ID: 25.4743.127.0.0.1(a)sipp.call.id CSeq: 1 INVITE Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.194:5060 "Noisy feedback tells: pid=4688 req_src_ip=192.168.178.194 req_src_port=5060 in_uri=sip:service@192.168.178.194:5060 out_uri=sip:service@192.168.178.194:5060 via_cnt==18" ' . sipp: There were more errors, enable -trace_err to log them. ======================================================= << test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="server.sam.cse" alias="sam.cse" alias="192.168.178.194" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes 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/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:vlftmd@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len>=204822) { 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=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save ??\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3?? ?? \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DB?? ?? ? ?? \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:mijin@sam.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.194" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 ? \n"); } _________________________________________________________________ ???? ?? ??? ???? ???... MSN ??? ??? http://vod.msn.co.kr _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] $from in avpops different?
by sip 23 Sep '05

23 Sep '05

I have this avpops test going... avp_db_load("$ruri/username", "$urire/$urire_scheme"); # Load the urire column for this and check to see if # it matches the $from address if(avp_check("$urire", "eq/$from/I")) { do stuff } I'm calling from the number number(a)domain.com. If I use the above, the avp_check test fails. If I put if(avp_check("$urire", "eq/number(a)domain.com/I")) instead, the check works exactly the way it's supposed to. It's almost as if the $from address isn't right. Is there some way I can check what it's doing at this point other than this painful trial and error? N.

1 0

[Serusers] need help
by chenhao 23 Sep '05

23 Sep '05

Hi! I realy curious about the details of how midiaproxy system works.I mean you know there three parts proxy sever, proxy dispather and mediaproxy module as i know.I got some general ideas from the readme file.But I dont think thats enough.Please send me some detail files or maybe one example or something.Thanks very much!

2 1

[Serusers] SIP/2.0 483 Too Many Hops
by 송 미진 23 Sep '05

23 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear.... I inserted command like this. What I want is that when I loaded test.cfg file, what I check SIP messages using sipp client mode I want 200 OK. But, as you see, I received "SIP/2.0 483 Too Many Hops" error message. So I was advanced about this. - alias="sam.cse"//SIP_ DOMAIN alias="server.sam.cse"//my_computer.SIP_DOMAIN alias="192.168.178.194"//my_ip - add my hostname in /etc/host - change to : if (!mf_process_maxfwd_header("100") but, I couldn't solve this problem. I would really appreciate for giving some advise. Thank you in advance............................. mi-jin Song ===================================================== [root@cse4 ~]# ser -f test.cfg Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.178.194 [192.168.178.194]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.178.194 [192.168.178.194]:5060 Aliases: tcp: server:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: cse4:5060 udp: server:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: cse4:5060 *: 192.168.178.194:* *: sam.cse:* *: server.sam.cse:* stateless - initializing 0(0) Maxfwd module- initializing exec - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) ERROR: tcp_init: bind(6, 0x810dafc, 16) on 127.0.0.1: Address already in use ===================================================== [root@cse4 sipp]# ./sipp -sn uac 192.168.178.194 Resolving 192.168.178.194... ------------------------------ Scenario Screen -------- [1-4]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 5061 2.54 s 25 192.168.178.194:5060(UDP) 5 new calls during 0.538 s period 2 ms scheduler resolution 0 concurrent calls (limit 30) Peak was 1 calls, after 0 s 0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE ----------> 25 0 0 100 <---------- 25 0 25 180 <---------- 0 0 0 200 <---------- E-RTD 0 0 0 ACK ----------> 0 0 [ 0 ms] BYE ----------> 0 0 0 200 <---------- 0 0 0 ------------------------------ Test Terminated --------------------------------- ----------------------------- Statistics Screen ------- [1-4]: Change Screen -- Start Time | 2005-09-23 20:03:13 Last Reset Time | 2005-09-23 20:03:15 Current Time | 2005-09-23 20:03:16 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:536 | 00:00:02:543 Call Rate | 9.328 cps | 9.831 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 5 | 25 Total Call created | | 25 Current Call | 0 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 0 Failed call | 6 | 25 -------------------------+---------------------------+-------------------------- Response Time | 00:00:00:000 | 00:00:00:000 Call Length | 00:00:00:004 | 00:00:00:004 ------------------------------ Test Terminated -------------------------------- 2005-09-23 20:03:16: Unexpected message for Call-ID '25.4743.127.0.0.1(a)sipp.call.id': while expecting '100' response, received 'SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 127.0.0.1:5061;received=192.168.178.194 From: sipp <sip:sipp@127.0.0.1:5061>;tag=25 To: sut <sip:service@192.168.178.194:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6e68 Call-ID: 25.4743.127.0.0.1(a)sipp.call.id CSeq: 1 INVITE Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.194:5060 "Noisy feedback tells: pid=4688 req_src_ip=192.168.178.194 req_src_port=5060 in_uri=sip:service@192.168.178.194:5060 out_uri=sip:service@192.168.178.194:5060 via_cnt==18" ' . sipp: There were more errors, enable -trace_err to log them. ======================================================= << test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="server.sam.cse" alias="sam.cse" alias="192.168.178.194" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes 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/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:vlftmd@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len>=204822) { 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=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save 성공\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3으로 이동 \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DB에서 찾을 수 없음 \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:mijin@sam.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.194" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 끝 \n"); } _________________________________________________________________ 책상위에 다리 올리고 느긋하게 즐긴다... MSN 온라인 상영관 http://vod.msn.co.kr

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[Serusers] error 513 when sendin a message
by Ángel Gómez 23 Sep '05

23 Sep '05

Hi, I has send a MESSAGE to my SER server and it allways send me the response "513 message to big". I send it with sipsak -s sip:ninguna@10.22.1.249 -vvv -l 5060 -M, when I send a REGISTER or INVITE it works whell. I have alias for various domains, an it can´t be te error, have somo one had this error ever? How can I get RIP of it? please help.

1 0

[Serusers] What is a difference between Class4 and Class 5 Softswitch ( VOIP)
by Manoj Ghag 23 Sep '05

23 Sep '05

hi, All Good morning Can anybody tell me what is a exact difference between Class4 and Class5 softswitch ? Because many people or company have there own defination Regards --Manoj --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

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[Serusers] SER and sipsak problem;;;;;;;
by 송 미진 23 Sep '05

23 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear. Alll Hi, I have some questions............. My study object is "communication among many computers(three computers)based SIP". So I have used SER with sipsak. (operating system : Linux Fedora 3) but SER was not runned well. In frankly, I don't know what to do. Following is my try, my SIP_DOMAIN is "sam.cse" and ip is 192.168.178.194. [root@cse4 ~]# sipsak -s sip:mijin@sam.cse -M warning: ignoring -i option when in usrloc mode received: SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 127.0.0.1:32778;received=192.168.178.194;branch=z9hG4bK.01b7ca3a;rport=32778 From: sip:sipsak@127.0.0.1:32778;tag=141698a0 To: sip:mijin@sam.cse;tag=b27e1a1d33761e85846fc98f5f3a7e58.5de2 Call-ID: 337025184(a)127.0.0.1 CSeq: 1 MESSAGE Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.194:5060 "Noisy feedback tells: pid=3626 req_src_ip=192.168.178.194 req_src_port=5060 in_uri=sip:mijin@sam.cse out_uri=sip:mijin@sam.cse via_cnt==18" error: did not received the 'MESSAGE' that was sent (see above). aborting =============================================== I want to communicate another computer has ip:192.168.178.193. [root@cse4 ~]# ser -f /usr/local/etc/ser/ser.cfg -l 192.168.178.193 0(3965) WARNING: fix_socket_list: could not rev. resolve 192.168.178.193 0(3965) WARNING: fix_socket_list: could not rev. resolve 192.168.178.193 Listening on udp: 192.168.178.193 [192.168.178.193]:5060 tcp: 192.168.178.193 [192.168.178.193]:5060 Aliases: stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) ERROR: udp_init: bind(4, 0x810ab6c, 16) on 192.168.178.193: Cannot assign requested address =============================================== I setted my hostname and ip in /etc/host. and inserted alias="sam.cse" alias="server.sam.cse" alias="192.168.178.194" but, as you see, I received "483 Too Many Hops" message;;;;; I don't know what to do................ I appreciate if you give me some solutions, please ^^ Thank you in advance.... mi-jin Song _________________________________________________________________ 증권 정보 가장 빠르고 편하게 보실 수 있습니다. MSN 증권/투자 http://www.msn.co.kr/stock/

1 0

[Serusers] LCR usage (How to enter data in gw table?)
by Amos Nungu 23 Sep '05

23 Sep '05

Hi, I have a question on how to enter values specifically ip_address in the gw table? I would like to give a go this LCR module. I'm facing a problem where by: - 1. I created the tables (gw, lcr, gw_grp) 2. Inserting values into the tables ready for use. The gw:ip_addr field is int data type, not able to store the decimal IP format!! How to manage that? Thanks in advance.

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[Serusers] One path of RTP traffic
by Alberto 23 Sep '05

23 Sep '05

Hi, I have a SER + Mediaproxy. I have not any problem the call between SIP clients (behind or not the NATs) but when I try to call to PSTN (via cisco) I only have RTP traffic from SIP client to PSTN. Summarizing, the path of rtp traffic would have to be from: up: SIP Client ----> SER ---> GW-PSTN down: SIP Client <---- SER <--- GW-PSTN but, really is: up: SIP Client ----> SER ---> GW-PSTN down: SIP Client <------------- GW-PSTN I use the command: rewritehostport("212.xxx.xxx.xxx:5060"); when I match a geographic number. The complete scheme is: SIP Client ---- NAT --------- SER+Mediaproxy -------- SIP Server --- GWPSTN Some idea? Thanks, -- Alberto

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[Serusers] serweb configuration
by Elad Dotan 23 Sep '05

23 Sep '05

Hi, I've installed 4 RPM files on a Fedora CORE 3. The RPM are: ser-0.9.3-0.i386.rpm php-pecl-classkit-0.4-2.i386.rpm ser-mysql-0.9.3-0.i386.rpm serweb-0.9.3-0.i386.rpm I enabled the httpd and mysql service. I'm not sure what to do next. All the Documentation I have is related to source flies and not RPM. Ser is working but I don't know how to configure the serweb in order for it to work. Could you Please help? Thank you, Elad.

1 0

[Serusers] Problems with radius_proxy_authorize and others
by Bruno Machado 23 Sep '05

23 Sep '05

Hello We are running SER 0.8.14 and FreeRadius 1.0.4 on Slackware 10 kernel 2.6.12.3. Yesterday, we had problems with some functions of SER. radius_www_authorize & www_challenge radius_proxy_authorize & proxy_challenge if(method=="REGISTER") { Make sure that user's dont register infinite loops # (note: does not match with folded lines) if(search("^(Contact|m):.*@(domain\.com\.br)")) { log(1, "LOG\n"); sl_send_reply("476","No Server Address in Contacts Allowed" ); break; }; #if (!radius_www_authorize("")) #{ #www_challenge("","0"); # break; #}; #if (!radius_proxy_authorize( "")) { #proxy_challenge("sipproxy.taho.com.br", "0" ); #break; #}; save("location"); #if (!save("location")) { # sl_reply_error(); #}; #m_dump(); break; }; This block ( if(method=="REGISTER"){block} ) has these functions. When "radius_www_authorize & www_challenge" or "radius_proxy_authorize & proxy_challenge" are uncommented, the SER becomes too slow: it receives around 5 consecutive invites in 7 seconds. After the fifth INVITE, SER answers with a 487 to the User Agent. The call to PSTN is established after around 30 seconds . When we comment these functions, the system runs perfectly, fast and fast. But is strange, because these functions are called normally in the routes we the INVITE is received: route[5] ............. if (!radius_proxy_authorize("")) { proxy_challenge( "domain.com.br", "0" ); break; }; ............. In the route, they work perfectly. I'd like to know if I'm doing wrong things in this scripts. Thanks a lot, friends. Bruno Machado __________________________________________________ Faça ligações para outros computadores com o novo Yahoo! Messenger http://br.beta.messenger.yahoo.com/

2 1

[Serusers] How to use "serctl file <file_name" command?
by Bhaskar Bhumkar 23 Sep '05

23 Sep '05

Hi, I am planning to use SER to send some SIP requests. Is it possible to initiate requests like INVITE from SER? If yes, how can I do that? I saw the description of "serctl file <file_name" and it is supposed to be used to read commands from a file. Does it refer to rest of the serctl commands or SIP requests like INVITE, REGISTER? Appreciate your help, Regards, Bhaskar...

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[Serusers] JABBER and SER
by Saleh, Saleh 22 Sep '05

22 Sep '05

Hello Everyone, I am new to this list. I would greatly appreciate anyone who can tell me what I need to do next in order for SER version 0.9.3 to use the Jabber Module correctly? This is what I have done so far (on a Red Hat Linux 9 machine): Installed Jabberd 1.4.4 Installed GNU pth-2.0.4 Installed Expat 1.95.8 Installed Libidn -I have Modified the config file for SER to load the "Jabber.so" module after the "pa.so" and "tm.so" modules have been loaded -(I'm sure this is where I am stuck, what to do with parameters in config file for SER but I have started to place those in as well, I am just confused about making the documentation match my settings/variable names) -I have partially created a database in MySQL but I have not put tables or data in it because I have never done this before. This is what I can/cannot do: -I can run the jabber server -I cannot run the SER with the config file which includes the Jabber.so module load command. What other files do I need to modify or download? What information do I need? I'm not very familiar with linux, so please assume I know basic commands/knowledge... Any comments/suggestions/websites/help is GREATLY appreciated! Thanks, Saleh

1 0

[Serusers] Avpops info....?
by sip 22 Sep '05

22 Sep '05

Is there any good, DETAILED information on AVPops usage that, say, doesn't read like a developers API reference page? Currently, I'm cleaning up portions of my ser.cfg which have horrid, ugly hacks in them that I put in to handle this case or that case... For instance, I use the calls_forwarding table to handle call blocking and call forwarding info (since it seems a little more reasonable than usr_preferences as it simply has more relevant tables). For call blocking, I can say give a purpose of 'blocking' and an action of relay or reply... with parameters set to what kind of reply or where to relay (voicemail server? Alternate server? etc). For forwarding, I can give an action of forwarding with a parameter of where to forward and another to see if he has permission to forward to PSTN numbers. Seems like a very handy table for it. Now, my current method of doing this is with this horrid mixture of external scripts, stripping portions off the SIP_HF_FROM header (since it comes with additional info like tags and brackets and comments), and feeding the new stuff back into an SQL query to determine whether or not the call needs to be blocked, forwarded, and if either, where it goes. It's a mess, I'll admit, but it's because I've been poring over AVPops stuff and have yet to figure out what I'm actually DOING with any of it. I get that it uses little i:XX constructs as variables (which is bizarre, but whatever -- I imagine I could alias them to something human-readable). And... well... that's about as far as I've gotten deciphering the API docs and making them into something usable. I imagine I would what... set the avp_table to be calls_forwarding, and then create some schemes for it (?)... but after that, actually doing things like taking just the sip:bob@fred.com portion of an incoming uri (which would look along the lines of "Ima Doofus" <sip:bob@fred.com:5060;user=tel>tag=something;tag2=something else ) and then trying to determine if that user is on the list of blocked users and, if so, what should be the response action... or taking the callee info, and if the callee isn't available, checking to see if he has a forwarding number, checking to see where it goes, etc, etc. It all SEEMS like it should be elementary stuff, but I'm simply not getting it. Anyone have any good references, well-detailed documentation, long, in-depth books on the topic they could send me toward? I'd be your friend for, like, ever. :) N.

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Re: [Serusers] Avpops info....?
by Paul Hazlett 22 Sep '05

22 Sep '05

You could look at the Getting Started document on www.onsip.org because we use the AVPOPS module in several examples and we give a much deeper description of what the functions do. However, we only use a subset of the AVPOPS library. Regards, Paul On 9/22/05, sip <sip(a)arcdiv.com> wrote: > Yeah... the problem with this documentation is that it was clearly written by > someone who doesn't WRITE documentation as a matter of course, but needed > something to be out there just to say there's documentation. > > It's a little like a developer writing docs for his own code or API. VERY > understandable to the developer himself, but not so crystal to anyone else. > This is one of the primary reasons I never let my developers write their own > API docs. They get with a tech writer, and the tech writer asks questions > about what the API does, and then writes documentation in such a way that > monkeys (such as myself) can understand. > > I was looking for something a little more... lucid. > > If it doesn't exist, cool. I'll keep muddling away and perhaps I'll hit that > eureka point of understanding somewhere, but I was hoping. > > N. > > > On Thu, 22 Sep 2005 09:29:31 -0400, Paul Hazlett wrote > > The offical documentation for AVPOPS (for SER-0.9.x) can be found here: > > > > http://www.voice-system.ro/docs/avpops/0.9.0/ > > > > Regards, > > Paul > > > > On 9/22/05, sip <sip(a)arcdiv.com> wrote: > > > Is there any good, DETAILED information on AVPops usage that, say, doesn't > > > read like a developers API reference page? Currently, I'm cleaning up portions > > > of my ser.cfg which have horrid, ugly hacks in them that I put in to handle > > > this case or that case... > > > > > > For instance, I use the calls_forwarding table to handle call blocking and > > > call forwarding info (since it seems a little more reasonable than > > > usr_preferences as it simply has more relevant tables). For call blocking, I > > > can say give a purpose of 'blocking' and an action of relay or reply... with > > > parameters set to what kind of reply or where to relay (voicemail server? > > > Alternate server? etc). For forwarding, I can give an action of forwarding > > > with a parameter of where to forward and another to see if he has permission > > > to forward to PSTN numbers. Seems like a very handy table for it. > > > > > > Now, my current method of doing this is with this horrid mixture of external > > > scripts, stripping portions off the SIP_HF_FROM header (since it comes with > > > additional info like tags and brackets and comments), and feeding the new > > > stuff back into an SQL query to determine whether or not the call needs to be > > > blocked, forwarded, and if either, where it goes. > > > > > > It's a mess, I'll admit, but it's because I've been poring over AVPops stuff > > > and have yet to figure out what I'm actually DOING with any of it. I get that > > > it uses little i:XX constructs as variables (which is bizarre, but whatever -- > > > I imagine I could alias them to something human-readable). And... well... > > > that's about as far as I've gotten deciphering the API docs and making them > > > into something usable. > > > > > > I imagine I would what... set the avp_table to be calls_forwarding, and then > > > create some schemes for it (?)... but after that, actually doing things like > > > taking just the sip:bob@fred.com portion of an incoming uri (which would look > > > along the lines of "Ima Doofus" > > > <sip:bob@fred.com:5060;user=tel>tag=something;tag2=something else ) and then > > > trying to determine if that user is on the list of blocked users and, if so, > > > what should be the response action... or taking the callee info, and if the > > > callee isn't available, checking to see if he has a forwarding number, > > > checking to see where it goes, etc, etc. > > > > > > It all SEEMS like it should be elementary stuff, but I'm simply not getting > > > it. Anyone have any good references, well-detailed documentation, long, > > > in-depth books on the topic they could send me toward? > > > > > > I'd be your friend for, like, ever. :) > > > > > > N. > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

1 0

[Serusers] Audiocodes + SER
by GT 22 Sep '05

22 Sep '05

Hello All, Immediately I start SER, I get the following errors using SER with Audiocodes. Is it SER working with Audiocodes, if not which devices are recommended ? MP-104 errors Version ID: 4.60A.016.003 1d:23h:23m:11s ( sip_stack)(653 ) !! [ERROR] SIPStackMngr: Message was not Parsed correctly ! 1d:23h:23m:11s ( sip_stack)(654 ) !! [ERROR] Last parsed line was : 1d:23h:23m:11s ( sip_stack)(655 ) !! [ERROR] The error was = Unexpected input 1d:23h:23m:11s ( sip_stack)(656 ) !! [ERROR] The Line of the error was = 1 1d:23h:23m:11s ( sip_stack)(657 ) !! [ERROR] The Column of the error was = 2 Thanks, Ggtel

1 0

[Serusers] Remote Party ID
by Sebastian Kühner 22 Sep '05

22 Sep '05

Hi, I'm using the append_rpid_hf() command to add the remote party id. I've implemented it like this: modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@xxx.xxx.xxx.xxx;user=phone>;party=calling;id-type=subscriber;screen=yes;privacy=off") [...] # first the caller needs to be authenticated if (uri=~"^sip:[0-9]*@(xxx.xxx.xxx.xxx|(voip1\.)?test\.com)") { if (!(src_ip==xxx.xxx.xxx.xxx | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("xxx.xxx.xxx.xxx", "subscriber")) { proxy_challenge( "xxx.xxx.xxx.xxx","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; }; append_rpid_hf(); What I noticed is that sometimes ser sends it and sometimes not. I have a value in the rpid column of the user... Has anyone of you an idea why that happens?? Thanks! Sebastian

1 0

[Serusers] Can I put network or subnet in trusted table?
by Ryan Pagquil 22 Sep '05

22 Sep '05

Hi, Will SER able to determine if I put a whole network or subnet in the trusted table? So that I will not need to put all the ip addresses of my network in the trusted table. 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] dbtext module.
by Skimmer 22 Sep '05

22 Sep '05

Hello! Could you help me? I try to use ser with dbtext module. In ser.cfg I have... ------------------------------------------------ ... loadmodule "/opt/sfw/lib/ser/modules/dbtext.so" ... modparam("usrloc", "db_mode", 2) modparam("usrloc|auth_db", "db_url", "/var/sertdb" ... Where /var/sertdb - the path to text files (the tables) I have the following tables: ----------------------------------------------- root@Solaris>ls -l /var/sertdb/ total 6 -rw-rw-rw- 1 root other 90 Sep 22 13:04 location -rw-rw-rw- 1 root other 59 Sep 22 12:26 subscriber -rw-rw-rw- 1 root other 59 Sep 22 13:07 version The "Location" file ----------------------------------------------- username(str) domain(str,null) contact(str) expires(int) q(double) callid(str) cseq(int) But when I start ser I got the following messages in log: ----------------------------------------------- Sep 22 13:32:19 Solaris ser[8611]: [ID 767465 daemon.warning] Warning: uac does not spread accross the whole hash table Sep 22 13:32:19 Solaris ser[8611]: [ID 736066 daemon.error] preload_udomain(): Error while doing db_query Sep 22 13:32:19 Solaris ser[8611]: [ID 247783 daemon.error] register_udomain(): Error while preloading domain 'location' Sep 22 13:32:19 Solaris ser[8611]: [ID 551419 daemon.error] domain_fixup(): Error while registering domain What is the problem? What I do wrong? Thank you. Best regards. Simon. -----------------ser.cfg----------------------------- ----------------------------------------------------- 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 "/opt/sfw/lib/ser/modules/mysql.so" loadmodule "/opt/sfw/lib/ser/modules/dbtext.so" loadmodule "/opt/sfw/lib/ser/modules/sl.so" loadmodule "/opt/sfw/lib/ser/modules/tm.so" loadmodule "/opt/sfw/lib/ser/modules/rr.so" loadmodule "/opt/sfw/lib/ser/modules/maxfwd.so" loadmodule "/opt/sfw/lib/ser/modules/usrloc.so" loadmodule "/opt/sfw/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/opt/sfw/lib/ser/modules/auth.so" #loadmodule "/opt/sfw/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) modparam("usrloc|auth_db", "db_url", "/var/sertdb" modparam("usrloc", "user_column", "username" # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) #

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[Serusers] 513 message too big
by Ángel Gómez 22 Sep '05

22 Sep '05

hi people, I have instaled SIP Express Router in al PC with debian sarge and trying to test it. I have tested the REGISTER and INVITE request, and it works, but when I send a MESSAGE with sipsak (sispak - M) the server gave me a response of "513 message too big". I have search the web for a solution but I did got it. The message arive to te server one time (it don`t enter into a bucle). Thank for helping, and sorry for mi bad english.

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[Serusers] how to log in softphone?
by zhangshuai 22 Sep '05

22 Sep '05

Dear all, I'm new to SER. I wish to use softphone X-lite on SER and radius in db_mode. I have installed SER 0.8.14 (by tarball) and Mysql 3.23.54 in Redhat 9. I have added some subscribers into DB "ser". And after I added some users in the file: /usr/local/etc/raddb/users as the document "iptel.org SIP Express Router v0.11.0 -- Admin's Guide"(http://www.iptel.org/ser/doc/seruser/seruser.html#DEFAULTSCRIPT) guided in Chap 3.3: joe(a)iptel.org Auth-Type := Digest, User-Password == "heslo" Reply-Message = "Authenticated", Sip-Rpid = "1234" I could log in X-lite and make calls. Now, how could I log in X-lite by using the users that DB contains rather than those in the file: /usr/local/etc/raddb/users? Many thanks. Shuai http://www.godentek.biz

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Re: RE: [Serusers] need help
by zhangshuai 22 Sep '05

22 Sep '05

hi, Rodrigue ESTEVE�� I think the counter will increase every 2 or 3 seconds. Hi, Is the counter increasing when you start ?nbsp;serctl monitor ?nbsp;? Are there any errors in your /var/log/syslog for ser ? Do you still receive icmp port unreachable error ? Can you do the following: # nc -u your_sip_ip 5060 Type something, if you receive an error message like this : read(net): Connection refused, your sip port is not reachable. De : julien [mailto:julientoser@gmail.com] Envoy?nbsp;: jeudi 22 septembre 2005 07:31 ?nbsp;: Iqbal Cc : Rodrigue ESTEVE; serusers(a)iptel.org Objet : Re: [Serusers] need help sorry... i want to say that: i can not see port 5060 when i try netstat -an. Cos i test wit fedora, so i am using ser version 0.8.12 On 9/22/05, julien <julientoser(a)gmail.com> wrote: Now, ser is running. But i can see port 5060 when i try netstat -an. I force the line listen=10.0.0.6 (ip of server linux) in ser.cfg something is wrong? On 9/22/05, julien <julientoser(a)gmail.com > wrote: Hi all, At least i see in the console the line error in ser.cfg, so i am trying to correct it. On 9/22/05, julien bossart <julientoser(a)gmail.com > wrote: Thank you for your instruction. To:Rodrigue Actually, i don't see that my server linux open port 5060 when i use netstat -an To:iqbal I can ping to other IP address. Today, i make the same example, but with a line ADSL so my server now have ip 10.0.0.6 (ip private). Then i use ip 10.0.0.5 for sip client. I try to connect to server but i get the same IMCP. Today, i tried to do: /etc/init.d/ser start -c -E to check the config file, it notice that: ERROR: bad config file (3 errors). (I don't understand cos i use ser.cfg default and uncomment only some line for mysql) What will you do when you meet this error? how can i active mode debug to find the error in ser.cfg? Thank you so much for your instruction. Julien On 9/21/05, Iqbal <iqbal(a)gigo.co.uk> wrote: and find the IP address of the client, do a traceroute to it, although this will fail it might let you see how far it is getting. Also can you ping any other IP addresses like ping www.yahoo.com iqbal Rodrigue ESTEVE wrote: > Hi Julien, > > Can you do this in command line on your linux : > > Netstat �an > > And check that your ser is listening on udp port 5060. > > rodrigue > > ------------------------------------------------------------------------ > > *De :* serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] > *De la part de* julien bossart > *Envoy?:* mercredi 21 septembre 2005 12:02 > *?:* serusers(a)iptel.org > *Objet :* [Serusers] need help > > Hi all, > > I am new to this forum. > > Today, i download ser package, then i install in my fedora core 4. > > Everything is ok. > > Next, i start ser and use serctl add to add account to ser server. > > Then i use sjphone to connect to ser server. > > Sjphone try to send REGISTER to ser server. I use ethereal to get > > the package. I see that ser server responds ICMP host unreacheable > > all the time. So, REGISTER is failed. > > Can you give me some suggestion? > > Thank you for that. > >------------------------------------------------------------------------ > >_______________________________________________ >Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > = = = = = = = = = = = = = = = = = = = = = = �� Shuai ��zhangshuai(a)goldentek.biz ��2005-09-22

1 0

[Serusers] ser + cpl-script timeuot question!
by Oskar Hartikainen 22 Sep '05

22 Sep '05

Hi. Can some one explain or have a hint on, why my ser 0.9.3 will not use timeout specified in a cpl-script (<proxy timeout="15">)? ser uses the modparam("tm", "fr_inv_timer", 20) specified in ser.cfg. how do i get ser to use the timeout from the cpl-script proxy tag. the script is working properly otherwise. Thanks, Oskar _________________________________________________________________________ <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE cpl SYSTEM "cpl-06.dtd"> <cpl> <subaction id="totest"> <location url="sip:test@7017ws12.haninge.kth.se"> <proxy/> </location> </subaction> <incoming> <location url="sip:oskar@7017ws07.haninge.kth.se"> <proxy timeout="15"> <busy> <sub ref="totest"/> </busy> <noanswer> <sub ref="totest"/> </noanswer> <failure> <sub ref="totest"/> </failure> </proxy> </location> </incoming> </cpl> ________________________________________________________________________

1 0

RE: [Serusers] need help
by Rodrigue ESTEVE 22 Sep '05

22 Sep '05

Hi, Is the counter increasing when you start « serctl monitor » ? Are there any errors in your /var/log/syslog for ser ? Do you still receive icmp port unreachable error ? Can you do the following: # nc -u your_sip_ip 5060 Type something, if you receive an error message like this : read(net): Connection refused, your sip port is not reachable. ________________________________ De : julien [mailto:julientoser@gmail.com] Envoyé : jeudi 22 septembre 2005 07:31 À : Iqbal Cc : Rodrigue ESTEVE; serusers(a)lists.iptel.org Objet : Re: [Serusers] need help sorry... i want to say that: i can not see port 5060 when i try netstat -an. Cos i test wit fedora, so i am using ser version 0.8.12 On 9/22/05, julien <julientoser(a)gmail.com> wrote: Now, ser is running. But i can see port 5060 when i try netstat -an. I force the line listen=10.0.0.6 <http://10.0.0.6/> (ip of server linux) in ser.cfg something is wrong? On 9/22/05, julien <julientoser(a)gmail.com > wrote: Hi all, At least i see in the console the line error in ser.cfg, so i am trying to correct it. On 9/22/05, julien bossart <julientoser(a)gmail.com > wrote: Thank you for your instruction. To:Rodrigue Actually, i don't see that my server linux open port 5060 when i use netstat -an To:iqbal I can ping to other IP address. Today, i make the same example, but with a line ADSL so my server now have ip 10.0.0.6 <http://10.0.0.6/> (ip private). Then i use ip 10.0.0.5 <http://10.0.0.5/> for sip client. I try to connect to server but i get the same IMCP. Today, i tried to do: /etc/init.d/ser start -c -E to check the config file, it notice that: ERROR: bad config file (3 errors). (I don't understand cos i use ser.cfg default and uncomment only some line for mysql) What will you do when you meet this error? how can i active mode debug to find the error in ser.cfg? Thank you so much for your instruction. Julien On 9/21/05, Iqbal <iqbal(a)gigo.co.uk> wrote: and find the IP address of the client, do a traceroute to it, although this will fail it might let you see how far it is getting. Also can you ping any other IP addresses like ping www.yahoo.com <http://www.yahoo.com/> iqbal Rodrigue ESTEVE wrote: > Hi Julien, > > Can you do this in command line on your linux : > > Netstat -an > > And check that your ser is listening on udp port 5060. > > rodrigue > > ------------------------------------------------------------------------ > > *De :* serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] > *De la part de* julien bossart > *Envoyé :* mercredi 21 septembre 2005 12:02 > *À :* serusers(a)lists.iptel.org > *Objet :* [Serusers] need help > > Hi all, > > I am new to this forum. > > Today, i download ser package, then i install in my fedora core 4. > > Everything is ok. > > Next, i start ser and use serctl add to add account to ser server. > > Then i use sjphone to connect to ser server. > > Sjphone try to send REGISTER to ser server. I use ethereal to get > > the package. I see that ser server responds ICMP host unreacheable > > all the time. So, REGISTER is failed. > > Can you give me some suggestion? > > Thank you for that. > >------------------------------------------------------------------------ > >_______________________________________________ >Serusers mailing list > serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers <http://lists.iptel.org/mailman/listinfo/serusers> > >

1 0

[Serusers] iptel.org setup
by Ryan Pagquil 22 Sep '05

22 Sep '05

Hi, I'm interested in iptel.org ser configuration. The security setup is nice, where those iptel.org users that are not registered will not be able to call anywhere, and also those users that are not belong to iptel.org will also not permitted to make a call. But registered users from other domains can call registered iptel.org users. What should be the ser.cfg for me to achieve this? 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

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