- sr-users - kamailio.org

[Kamailio-Users] snmpstats invalid condition in executeInterprocessBufferCmd
by mayamatakeshi 27 May '09

27 May '09

Hello, we have kamailio 1.4 (r. 5728) running in production. We compiled kamailio with 4 times the default pkg memory pool size (#define PKG_MEM_POOL_SIZE 4*1024*1024 ). We use snmpstats to monitor registration and number of calls using Nagios and PRTG. Some days ago, after almost 30 days of continuous operation without problems, in the shell we executed "snmpwalk -v2c -c public 192.168.88.22 .1.3.6.1.4.1.27483" as we routinely do for a quick check of SNMP in the command line and it failed. Then we checked kamailio logs and we found this: May 21 16:25:35 ipx022 /usr/local/sbin/kamailio[8781]: ERROR:snmpstats:handle_openserSIPServiceStartTime: failed to read sysUpTime file at /tmp/openSER_SNMPAgent.txt I don't know what caused the above (maybe the file was mistakenly rm'd by someone. I failed to check it at that moment) but it was followed with this: May 21 16:25:35 ipx022 /usr/local/sbin/kamailio[8781]: ERROR:snmpstats:executeInterprocessBufferCmd: Received a request for contact: [protected]@[protected] for user: sip:[protected]@[protected] who doesn't exists May 21 16:25:35 ipx022 /usr/local/sbin/kamailio[8781]: ERROR:snmpstats:executeInterprocessBufferCmd: Received a request to delete contact: [protected]@[protected] for user: sip:[protected]@[protected] who doesn't exist ... then the same log "contact XXX who doesn't exist" happened several times per second till: May 21 16:36:30 ipx022 /usr/local/sbin/kamailio[8781]: ERROR:snmpstats:createRegUserRow: failed to create a row for openserSIPRegUserTable May 21 16:36:30 ipx022 /usr/local/sbin/kamailio[8781]: ERROR:snmpstats:updateUser: openserSIPRegUserTable ran out of memory. Not able to add user: [protected]@[protected] May 21 16:36:30 ipx022 /usr/local/sbin/kamailio[8781]: ERROR:snmpstats:executeInterprocessBufferCmd: Received a request for contact: [protected]@[protected] for user: sip:[protected]@[protected] who doesn't exists May 21 16:36:30 ipx022 /usr/local/sbin/kamailio[8781]: ERROR:snmpstats:insertContactRecord: no more pkg memory May 21 16:36:30 ipx022 /usr/local/sbin/kamailio[8781]: ERROR:snmpstats:executeInterprocessBufferCmd: openserSIPRegUserTable was unable to allocate memory for adding contact: [protected]@[protected] to user sip:[protected]@[protected]. During this time, snmpget queries issued by Nagios and PRTG were failing and the process spawned by module snmpstats was making heavy use of CPU. No other parts of kamailio were affected (calls and registration were OK), but snmpstats didn't normalize by itself so we decided to restart kamailio. I'm trying to recreate this in a lab machine but no luck so far. I am keeping the lab server busy with registration and calls while running snmpwalk in a loop. I've also ran "rm /tmp/openSER_SNMPAgent.txt" to check if could trigger the problem but the problem could not be recreated. If I push the lab server enough, I can see similar logs like this: May 27 20:16:54 ipx029 /usr/local/sbin/kamailio[14487]: ERROR:snmpstats:updateUser: openserSIPRegUserTable ran out of memory. Not able to add user: [protected]@[protected] May 27 20:16:54 ipx029 /usr/local/sbin/kamailio[14487]: ERROR:snmpstats:executeInterprocessBufferCmd: Received a request for contact: [protected]@[protected] for user: sip:[protected]@[protected] who doesn't exists May 27 20:17:05 ipx029 /usr/local/sbin/kamailio[14487]: ERROR:snmpstats:get_socket_list_from_proto: no more pkg memory May 27 20:17:05 ipx029 last message repeated 9 times May 27 20:17:59 ipx029 /usr/local/sbin/kamailio[14487]: ERROR:snmpstats:handle_openserSIPServiceStartTime: failed to read sysUpTime file at /tmp/openSER_SNMPAgent.txt May 27 20:40:25 ipx029 /usr/local/sbin/kamailio[14487]: ERROR:snmpstats:handle_openserSIPServiceStartTime: failed to read sysUpTime file at /tmp/openSER_SNMPAgent.txt But even with those messages, snmpget/snmpwalk doesn't stop to return responses. I was hoping someone could take a look at snmpstats code; the logs for snmpstats:executeInterprocessBufferCmd are marked with comments like : /* This should never happen. This is more of a sanity check. */ so probably my server hit some bug. regards, takeshi

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Re: [Kamailio-Users] ACKed dialog remains in state 3 instead of 4
by catalina oancea 27 May '09

27 May '09

---------- Forwarded message ---------- From: catalina oancea <catalina.oancea(a)gmail.com> Date: 2009/5/27 Subject: Re: [Kamailio-Users] ACKed dialog remains in state 3 instead of 4 To: Alex Balashov <abalashov(a)evaristesys.com> Hi Alex, thanks for answering >From what I know the record-route header is not compulsory, and dialog-matching can also be done using rfc dialog-matching instead of the did parameter in record-route (modparam("dialog", "dlg_match_mode", 2)). This is what I am trying to use, I don't want to use the record-route header at all. 2009/5/27 Alex Balashov <abalashov(a)evaristesys.com>: > Catalina, > > Unless I am missing something, dlg_manage() does not allow you to avoid > setting Record-Route; how else will the endpoints know to send sequential > messages through the proxy once the dialog is established? > > The purpose of dlg_manage() is to provide an alternative to setting a flag > for the purpose of indicating to Kamailio that the dialog the current INVITE > seeks to establish should be statefully tracked and exposed to additional > user-defined dialog module functionality (profiles, statistics, etc.) and > other module functionality relying on internal API callbacks from the dialog > module. > > modparam("dialog", "dlg_flag", 4) > > route { > ... > > # Process initial INVITE. > > if(is_method("INVITE")) { > ... > > setflag(4); # Track this dialog. > > } > > } > > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (678) 237-1775 >

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[Kamailio-Users] Dispatcher
by CSB 27 May '09

27 May '09

I wish to implement a kind of a failover situation, probably using the dispatcher module. We want to monitor our gateways by performing a range of tests (not just a ping or INFO but also waiting for a specific response to an INVITE). If the test fails then we want to mark a gateway as unavailable until the tests are successful again. Based on a bit of reading I think we could use the dispatcher module in database mode. We could run a script that performs the tests and marks the destination as inactive if the test fails. If that would work, can anyone advise how we mark a destination as inactive? It looks like the MI function ds_set_state does this but can I update the database directly? If so what field do I need to update? Thanks Cameron

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[Kamailio-Users] #sip-router irc channel
by Daniel-Constantin Mierla 27 May '09

27 May '09

Hello, the irc fans that want to do "a more real-time chat" about using/developing sip router project, can join the #sip-router channel on irc.freenode.net. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] Call Ended problem
by Chandrakant Solanki 27 May '09

27 May '09

Hi I am using kamailio 1.5.0 and asterisk 1.6.0.5 Phone is register successfully and incoming/outgoing call is working fine... But followings are problem I have faced... 1] I don't hear anything, the voice is not being transmitted ... however rtpproxy is running. 2] When I hang up the X-Lite call on the SIP phone, the regular phone does not get disconnected. Below is link for kamailio configuration file... *http://en.pastebin.ca/1435982* <http://en.pastebin.ca/1435982> I am calling on asterisk server and forward it to kamailio.. -- Regards, Chandrakant Solanki

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[Kamailio-Users] What is $ds (destination set)
by Iñaki Baz Castillo 27 May '09

27 May '09

Hi, what is the pseudo-variable $ds (destination set)? http://kamailio.org/dokuwiki/doku.php/pseudovariables:1.5.x#destination_set Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] kamailio multi-domain
by karhu 27 May '09

27 May '09

Ok, for User2 i added in Xlite Domain : 10.10.10.2 Proxy : 192.168.0.2 And i am able to call User1, and User1 is able to call User2. Now the problem is i there is no RTP packets :s. That's weird because i enable rtpproxy with bridge mode, with the command : rtpproxy -l 10.10.10.2 192.168.0.2 -s udp:127.0.0.1:7890. Obviously in my .conf file kamailio is listening on udp:127.0.0.1:7890. Cheers -- View this message in context: http://www.nabble.com/kamailio-multi-domain-tp23539467p23556076.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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Re: [Kamailio-Users] Auth problems with 1.5.1
by Daniel-Constantin Mierla 27 May '09

27 May '09

On 05/25/2009 11:58 AM, Raúl Alexis Betancor Santana wrote: > On Monday 25 May 2009 08:46:51 you wrote: > >> there were no radical changes to auth modules, afaik, do not know about >> postgres. Can you run in debug mode and see why second 401 is sent? Also >> return code $rc will show some hints if you print it with xlog? >> > > Here is a debug=6 output > > [...] > May 25 09:47:59 [13194] DBG:auth:check_response: our result > = '60045096c392b92f8c7a451fbb09e530' > May 25 09:47:59 [13194] DBG:auth:check_response: authorization faile could you check that the password is ok in your phone? Try to auth using other phone/tool (sipsak) ... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] Looking for a way to do routing from a database
by Bruce Ferrell 27 May '09

27 May '09

I've spent many hours looking over the module docs and examples. is there a way to do dialplans from a database? Thanks

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[Kamailio-Users] OpenBSD port updated to version 1.5.1
by Don Jackson 27 May '09

27 May '09

Hello, I have updated the OpenBSD port of Kamailio to support the 1.5.1 release. The port is available at: http://kamailio.org/pub/kamailio/1.5.1/packages/openbsd/ The posting to the OpenBSD ports mailing list is here: http://archives.neohapsis.com/archives/openbsd/2009-05/1702.html Thanks again to Henning Westerholt for putting this up on the kamailio site. Best regards, Don

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[Kamailio-Users] Can't get allow_trusted to work
by Saúl Ibarra 26 May '09

26 May '09

Hi all, I'm a little stuck here :-/ I can't get allow_trusted function from the premissions module to word. I'm receiving the following error: May 26 18:36:24 [18825] DBG:core:parse_msg: method: <PUBLISH> May 26 18:36:24 [18825] DBG:core:parse_msg: uri: <sip:saghul@10.69.69.1:5070> May 26 18:36:24 [18825] DBG:core:parse_msg: version: <SIP/2.0> May 26 18:36:24 [18825] DBG:core:parse_headers: flags=2 May 26 18:36:24 [18825] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK1f7d.e63e45e5.0>; state=16 May 26 18:36:24 [18825] DBG:core:parse_via: end of header reached, state=5 May 26 18:36:24 [18825] DBG:core:parse_headers: via found, flags=2 May 26 18:36:24 [18825] DBG:core:parse_headers: this is the first via May 26 18:36:24 [18825] DBG:core:receive_msg: After parse_msg... May 26 18:36:24 [18825] DBG:core:receive_msg: preparing to run routing scripts... May 26 18:36:24 [18825] DBG:core:parse_headers: flags=100 May 26 18:36:24 [18825] DBG:core:parse_via_param: found param type 234, <received> = <10.69.69.2>; state=6 May 26 18:36:24 [18825] DBG:core:parse_via_param: found param type 235, <rport> = <5060>; state=6 May 26 18:36:24 [18825] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKqtsklyyn>; state=16 May 26 18:36:24 [18825] DBG:core:parse_via: end of header reached, state=5 May 26 18:36:24 [18825] DBG:core:parse_headers: via found, flags=100 May 26 18:36:24 [18825] DBG:core:parse_headers: parse_headers: this is the second via May 26 18:36:24 [18825] DBG:maxfwd:is_maxfwd_present: value = 69 May 26 18:36:24 [18825] ERROR:permissions:allow_trusted_2: unknown protocol udp Configuration is realli simple: modparam("permissions", "db_url", "mysql://...") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") if (!allow_trusted("$si", "$proto")) { xlog("L_INFO","\n\n--NOT trusted! - $si - $proto"); sl_send_reply("403", "Forbidden"); exit; } The trusted table seems OK: mysql> select * from trusted; +----+------------+-------+--------------+------+ | id | src_ip | proto | from_pattern | tag | +----+------------+-------+--------------+------+ | 1 | 10.69.69.1 | any | NULL | NULL | +----+------------+-------+--------------+------+ What am I doing wrong? Thanks in advance!! Regards, -- Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes." ---------------------------------------------------------------- http://www.saghul.net/

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[Kamailio-Users] Mediaproxy relay error: 'module' object has no attribute 'Interface'
by jh＠tutpro.com 26 May '09

26 May '09

CSB writes: > When attempting to start the MediaProxy relay I get the following: > > May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: Starting MediaProxy > Relay 2.3.4 > > May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: fatal error: failed to > create MediaProxy Relay: 'module' object has no attribute 'Interface' i just tried with 2.3.4 (from debian packages) and it worked without problems. in config.ini try if using ip addresses instead of domain names would help. -- juha

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[Kamailio-Users] Mediaproxy relay error: 'module' object has no attribute 'Interface'
by CSB 26 May '09

26 May '09

When attempting to start the MediaProxy relay I get the following: May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: Starting MediaProxy Relay 2.3.4 May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: fatal error: failed to create MediaProxy Relay: 'module' object has no attribute 'Interface' May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: Traceback (most recent call last): May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: --- <exception caught here> --- May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: File "./media-relay", line 55, in ? May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: from mediaproxy.relay import MediaRelay May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: File "/usr/lib/python2.4/site-packages/mediaproxy/relay.py", line 30, in ? May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: from twisted.names import dns May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: File "/usr/lib/python2.4/site-packages/twisted/names/dns.py", line 116, in ? May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: class IRecord(components.Interface): May 26 16:45:00 ip-10-250-54-212 media-relay[18032]: exceptions.AttributeError: 'module' object has no attribute 'Interface' rpm -q python-twisted-core python-twisted-core-2.5.0-4.el5 rpm -q python-twisted-names python-twisted-names-0.3.0-3.el5 rpm -q python-twisted-web2 python-twisted-web2-8.1.0-1.el5.rf Any assistance appreciated. Cameron

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[Serusers] Nathelper doesn't support multipart MIME even though allowed by RFC 3261?
by Frank Durda IV 26 May '09

26 May '09

I have someone who is trying to send me SIP-T messages through a SER system that acts as a PSTN gateway. The destination PSTN switch can accept either SIP or SIP-T as can the originator of the INVITE. The SIP-T message appears to pass through SER in both directions with the SIP-T data apparently intact. However, there are problems with the SDP payload. The functions in modules/nathelper that locate the SDP payload (mainly in check_content_type() in nhelpr_funcs.c) appear to be coded to never work properly with multi-part MIME in the message headers, nor deal with section boundary markers in the message body. So when the headers contain a: Content-Type: multipart/mixed;boundary=tekBoundary instead of the expected*: Content-Type: application/sdp nathelper gets confused and says things like: May 22 21:25:56 ser1 ser[24905]: ERROR:check_content_type: invalid type for a message May 22 21:25:56 ser1 ser[24905]: ERROR: extract_body: content type mismatching May 22 21:25:56 ser1 ser[24905]: ERROR: force_rtp_proxy: can't extract body from the message * (nhelpr_funcs.c also tests for Content-Types beginning with "icat", "ion" and "appl" in addition to sdp but doesn't know how to deal with any others, including a "multipart/mixed" and the resulting nested message body. Subsequently, calls to the various force_rtp_proxy() and related functions when processing the INVITE, 183 Session Progress or 200 OK replies do not substitute in the new IP address provided by rtpproxy (because it can't find the start of the body or a known body type), and so the party receiving this message will get the uncorrected IP address and audio for the call fails due to an unreachable IP address. This limitation is noted in ser-2.0.0-rc1. Has anybody done any development to make check_content_type() and the related functions in nhelpr_funcs.c capable of dealing with multi-part MIME sections? Ideally, SER should allow the entire message body even with more than one section to pass without comment or complaint. Meanwhile, nathelper should be able to find the sdp section and allow parsing and editing of it as would have been done if the body contained only one section. In reading RFC 3261, it appears that multi-part MIME is allowed and so it appears that nathelper is violating the RFC by not coping with this. Here is the passage on this subject that caught my attention: 7.4.1 Message Body Type (RFC 3261) The Internet media type of the message body MUST be given by the Content-Type header field. If the body has undergone any encoding such as compression, then this MUST be indicated by the Content- Encoding header field; otherwise, Content-Encoding MUST be omitted. If applicable, the character set of the message body is indicated as part of the Content-Type header-field value. The "multipart" MIME type defined in RFC 2046 [11] MAY be used within the body of the message. Implementations that send requests containing multipart message bodies MUST send a session description as a non-multipart message body if the remote implementation requests this through an Accept header field that does not contain multipart. SIP messages MAY contain binary bodies or body parts. When no explicit charset parameter is provided by the sender, media subtypes of the "text" type are defined to have a default charset value of "UTF-8". Thanks in advance to any pointers to newer nathelper code that deals with this RFC-permitted message body format. Here is an example of the INVITE message that caused the problem: INVITE sip:9712345678@230.123.45.67:5060 SIP/2.0 Via: SIP/2.0/UDP 225.67.89.1:5060;branch=z9hG4bKfi0cjdk0qb8sv24f18ki1ddao4 To: <sip:9712345678@230.123.45.67:5060> From: <sip:8005551212@225.67.89.1>;tag=11775339 Call-ID: 1241812549-66096587@SIP CSeq: 1 INVITE Max-Forwards: 69 Contact: <sip:8005551212@225.67.89.1:5060;transport=udp> Supported: 100rel Expires: 330 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, PRACK, REFER, SUBSCRIBE, NOTIFY, UPDATE, REGISTER P-Asserted-Identity: <sip:8005551212@8.4.99.59> MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=tekBoundary Content-Length: 426 --tekBoundary Content-Type: application/sdp v=0 o=- 321533952 1243026447 IN IP4 225.67.89.1 s=- c=IN IP4 225.67.89.1 t=0 0 a=sendrecv m=audio 49168 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --tekBoundary Content-Type: application/isup; base=ansi92; version=ansi ..x ..x...x..x.Pxx ....x@xp.xx..xx.. --tekBoundary--

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[Kamailio-Users] Error with table version
by Raúl Alexis Betancor Santana 25 May '09

25 May '09

This is and old issue that I still have around here ... -Kamailio 1.5.1 (svn) -PostgreSQL database -Database created with "kamdbctl create", so just clean - in kamailio.cfg: [...] loadmodule "uri_db.so" modparam("uri_db", "db_url", "postgres://openser:XXXXX@localhost:5435/voip3") modparam("uri_db", "db_table", "uri") modparam("uri_db", "user_column", "username") modparam("uri_db", "domain_column", "domain") modparam("uri_db", "uriuser_column", "uri_user") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) [...] May 24 10:22:27 [23631] INFO:core:init_mod: initializing module auth_db May 24 10:22:27 [23631] INFO:core:init_mod: initializing module uri_db May 24 10:22:27 [23631] ERROR:uri_db:mod_init: Invalid table version of the subscriber table May 24 10:22:27 [23631] ERROR:core:init_mod: failed to initialize module uri_db May 24 10:22:27 [23631] ERROR:core:main: error while initializing modules And I don't undestand why ... because subscriber table it's on version 6, that is what uri_db checks ... [uri_db_mod.c] /* Check table version */ ver = uridb_db_ver(&db_url, &db_table); if (ver < 0) { LM_ERR("Error while querying table version\n"); return -1; } else { if (use_uri_table) { if (ver != URI_TABLE_VERSION) { LM_ERR("Invalid table version of the uri table\n"); return -1; } } else { if (ver != SUBSCRIBER_TABLE_VERSION) { LM_ERR("Invalid table version of the subscriber table\n"); return -1; } } } [uri_db_mod.c] and SUBSCRIBER_TABLE_VERSION has value 6 is there something I'm doing wrong?, last time I compile myself this I was forced to comment that block of code on the uri_db sources. -- Raúl Alexis Betancor Santana Dimensión Virtual

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[Kamailio-Users] Auth problems with 1.5.1
by Raúl Alexis Betancor Santana 25 May '09

25 May '09

Hi all, I'm getting a lot of problems to migrate to 1.5.1, the last one is with auth. I'm using postgresql as DB and auth_db with the next params: [...] loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "realm_prefix", "sip.") modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "postgres://openser:XXXX@localhost:5435/voip3") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "use_domain", 1) modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=uuid") [...] With openser 1.3.2 and this: [...] if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed BI $T_branch_idx - M=$rm RURI=$ru F=$fu T=$tu IP=$si I D=$ci\n"); www_challenge("", "0"); exit; } [...] All is working ok .. but with kamailio 1.5.1 it doesn't run, It allway return 401 Authentication needed ... I don't found anything in the docs about any change on auth modules ... what could be happen ? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual

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[Kamailio-Users] Error runing svn version
by Raúl Alexis Betancor Santana 24 May '09

24 May '09

I'm geting this error when trying to run kamailio svn trunk version ... May 23 22:27:30 [17510] INFO:core:init_mod: initializing module usrloc May 23 22:27:30 [17510] INFO:usrloc:ul_init_locks: locks array size 512 May 23 22:27:30 [17510] ERROR:core:db_check_api: module db_postges does not export db_use_table function May 23 22:27:30 [17510] ERROR:usrloc:mod_init: failed to bind database module May 23 22:27:30 [17510] ERROR:core:init_mod: failed to initialize module usrloc May 23 22:27:30 [17510] ERROR:core:main: error while initializing modules ... Any clue? -- Raúl Alexis Betancor Santana Dimensión Virtual

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Re: [Kamailio-Users] Error runing svn version
by Daniel-Constantin Mierla 24 May '09

24 May '09

On 05/24/2009 10:50 AM, Raúl Alexis Betancor Santana wrote: > On Sunday 24 May 2009 08:38:42 you wrote: > >> Hello, >> >> On 05/24/2009 12:30 AM, Raúl Alexis Betancor Santana wrote: >> >>> I'm geting this error when trying to run kamailio svn trunk version >>> >>> ... >>> May 23 22:27:30 [17510] INFO:core:init_mod: initializing module usrloc >>> May 23 22:27:30 [17510] INFO:usrloc:ul_init_locks: locks array size 512 >>> May 23 22:27:30 [17510] ERROR:core:db_check_api: module db_postges >>> >> looks like there is a typo in the db_postgres name (you miss r or?!?). >> >> If you want to play with development version, then you should take the >> sources from GIT http://sip-router.org. It is the place where the >> development is done now. >> > > Umm, I want to use kamailio not sip-router by now, maybe I sould get .tar.gz > version of 1.5.1 instead of the svn > I thought you have Kamailio development version from svn trunk. If you have 1.5.x, use the SVN branch 1.5, it is recommended. There are some fixes since 1.5.1. Was the error the typo I the db_postgres module name? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Serusers] Memory allocation
by Jose Silva 22 May '09

22 May '09

Hi, I was trying to understand how SER manages its memory. From what I searched and read on the mailing list and SER documentation it seams to me that SER allocates all the necessary memory at start time and never allocates anymore memory. If the initial memory defined (PKG,SHM) is full no more will be allocated. Is this the correct Interpretation? Kind Regards, José Carlos Silva

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Re: [Kamailio-Users] Load Based Load Balancing
by Ovidiu Sas 22 May '09

22 May '09

Please keep the list in your replies. Check the dialog documentation carefully. It is all explained there: how to add dialogs into a profile and retrieve the them. The dialog profiles are stored into memory for fast access. Regards, Ovidiu Sas On Wed, May 20, 2009 at 4:26 PM, Kareem Hamdy <Kareem.Hamdy(a)trustvesta.com> wrote: > Thanks! > > I still don't know how to do it, but you've put me on the right track. I'm fairly new to Kamailio, so I'm trying to figure out all the syntaxing. I did an SQL query on my FS boxes, and I was able to see the number of dialogs per host. > > Is it possible to do something similar with Kamailio - where I store the dialogs in a DB and query it in the dispatcher section of my routes? > > If so, would you be able to give me an example syntax of how I'd go about doing so? > > Thanks, > Kareem > > > > > > -----Original Message----- > From: sip.nslu(a)gmail.com [mailto:sip.nslu@gmail.com] On Behalf Of Ovidiu Sas > Sent: Wednesday, May 20, 2009 12:10 PM > To: Kareem Hamdy > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] Load Based Load Balancing > > Hello Kareem, > > Use dialog profiling to track the number of calls per destination: > http://www.kamailio.net/docs/modules/1.5.x/dialog#id2531102 > In your script, before sending the INVITE out, check if the selected > destination has enough capacity to accept the call: > http://www.kamailio.net/docs/modules/1.5.x/dialog#id2532103 > > > Regards, > Ovidiu Sas > > On Wed, May 20, 2009 at 2:57 PM, Kareem Hamdy > <Kareem.Hamdy(a)trustvesta.com> wrote: >> Hello: >> >> I've been going through your lists and your documentation, and I'm unable to find how you set up load based load balancing. >> >> To further illustrate: >> >> >> >> For example, let's say we have the following live calls:1,2,3,4,5,6,7 >> >> Round Robin Works Well >> ---------------------- >> >> +-----+ >> | SER | >> +-----+ >> | >> | +--------------+ >> |-------| FreeSWITCH 1 | [Calls 1,3,5,7] >> | +--------------+ >> | >> | +--------------+ >> |-------| FreeSWITCH 2 | [Calls 2,4,6] >> +--------------+ >> >> >> >> But let's say some of the calls end significantly quicker than the others, how can I send calls by their load (current calls, cpu, etc) >> >> For example, let's say we have the following live calls:1,2,3,4,5,6,7 - Let's say calls 4 and 5 lasted about 30 seconds, while the others lasted about 5 minutes. How would I be able to route calls to the box that's least used - in this example, "FreeSWITCH 2" at any given time had the least number of calls, so calls 4,5, and 6 were dispatched to it. >> >> Load Based Load Balancing >> ------------------------- >> >> +-----+ >> | SER | >> +-----+ >> | >> | +--------------+ >> |-------| FreeSWITCH 1 | [Calls 1,3,7] >> | +--------------+ >> | >> | +--------------+ >> |-------| FreeSWITCH 2 | [Calls 2,6] >> +--------------+ >> >> >> >> Thanks, >> Kareem >> >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >

2 1

[Serusers] SER fail authentication using RADIUS
by Leon Li 22 May '09

22 May '09

Hi, I got the following error when using FreeRADIUS to authenticate SER registration. May 21 04:46:57 ser[3352]: REGISTER May 21 04:46:57 ser[3352]: REGISTER: challenging user May 21 04:46:57 ser[3354]: REGISTER May 21 04:46:57 ser[3354]: pre_auth(): Credentials received are not filled properly What does this mean? Also, I don't see any RADIUS traffic generated by SER. Please assist. Thanks Leon

1 0

[Kamailio-Users] Dialog profile size not resetting in 1.4.4.
by Alex Balashov 21 May '09

21 May '09

Greetings, I've run into an issue with 1.4.4 in which the following appears to be happening, and I'm yet to determine the cause: The total amount of open inbound calls (dialogs) is tracked with the dialog module by putting them in a global profile. Because of a segfault bug with profiles with "no values," a workaround is used wherein a profile "with" values is created and the hash key is always the same value (i.e. "1"). The dialog tracking flag is 2: modparam("dialog", "dlg_flag", 2) And profile looks like this: modparam("dialog", "profiles_with_value", "globalinbound;specificinbound") Anyway, inbound INVITEs get handed off to a route (route[1]) that performs some logic like this: route[1] { # Check if DID is assigned to an account with database # and store value in AVP, along with other values. if($avp(S:did_assigned) == "0") { sl_send_reply("404", "Not Found"); exit; } get_profile_size("specificinbound", "$avp(S:account_id)", "$var(a_concurrent_calls)"); if($var(a_concurrent_calls) >= $avp(S:port_limit)) { sl_send_reply("486", "Busy"); exit; } setflag(2); # Track dialog. set_dlg_profile("globalinbound", "1"); set_dlg_profile("specificinbound", "$avp(S:account_id)"); } What appears to be happening is that calls coming into unassigned numbers ($avp(S:did_assigned) == "0") increase the size of the "globalinbound" profile, which is something of a mystery to me since I am neither setting flag 2 nor calling set_dlg_profile() until further down in the logic! And after the 404 Not Found reply is sent, the profile size is not decremented. Is the use of statefully tracked replies (t_reply()) required in order to propagate dialog state through to the dialog module? My impression is this was not the case. More pressing concern is that profile size is increased and not decreased; in my interpretation, such calls should not be added to the 'globalinbound' profile at all. I have not had the opportunity to try to reproduce this with 1.5.x. Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Kamailio-Users] "Detached" timer messages.
by Alex Balashov 21 May '09

21 May '09

Greetings, I was wondering if someone fluent in the anatomy of the TM module can tell me what this condition actually implies in a commonsensical sort of way, from a user perspective -- it is on line 861 in modules/tm/timer.c: /* check first if we are on the "detached" timer_routine list, * if so do nothing, the timer is not valid anymore * (sideffect: reset_timer ; set_timer is not safe, a reseted timer * might be lost, depending on this race condition ) */ if (new_tl->timer_list==DETACHED_LIST){ LM_CRIT("set_timer for %d list called on a \"detached\" " "timer -- ignoring: %p\n", list_id, new_tl); goto end; } Reason I am wondering is that I got a few of these error messages in my logs: May 20 18:23:50 ser1 /usr/local/sbin/kamailio[25232]: CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x2a96cc0d50 And was not sure what to make of them. Thank you! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

2 3

[Kamailio-Users] Mailing list questions
by Kareem Hamdy 21 May '09

21 May '09

Is there an easier way to search through the mailing list other than going to http://lists.kamailio.org/pipermail/users/, and searching in my Firefox based on the subject name? Thanks

3 2

[Kamailio-Users] RLS - is rls-services supported ?
by Pascal Maugeri 21 May '09

21 May '09

Hi I'm doing an integration of the RLS module for Kamailio 1.5 and a third-party XCAP server. I see that when my client sends a SIP SUBSCRIBE to a list that is defined into the document rls-services, I only see the traffic from Kamailio/RLS to the XCAP server to retrieve a resource-lists document, nothing about the rls-services document ?! Does RLS support rls-services ? If not what lists should contain the resource-lists document ? Cheers Pascal

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Re: [Kamailio-Users] XCAP_CLIENT failing to fetch/parse resource-lists document
by Daniel-Constantin Mierla 21 May '09

21 May '09

On 05/20/2009 01:04 PM, Pascal Maugeri wrote: > > that is caused by CURLE_WRITE_ERROR. > > Do you get a pkg memory or so error before those pasted above? > > > Well I only get the following line before: > > May 20 13:59:52 sandbox /usr/sbin/kamailio[11424]: > WARNING:core:fm_free: free(0) called > > [...] > > Any idea of what could go wrong ? Do you get the error for all xcap queries? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Serusers] SER fail authentication using RADIUS
by Leon Li 21 May '09

21 May '09

Hi, I got the following error when using FreeRADIUS to authenticate SER registration. May 21 04:46:57 ser[3352]: REGISTER May 21 04:46:57 ser[3352]: REGISTER: challenging user May 21 04:46:57 ser[3354]: REGISTER May 21 04:46:57 ser[3354]: pre_auth(): Credentials received are not filled properly What does this mean? Also, I don't see any RADIUS traffic generated by SER. Please assist. Thanks Leon

1 0

[Kamailio-Users] DTMF and Kamailio
by Simon Muyal 20 May '09

20 May '09

Hello, I have the following topology: IPphones <-> IPBX-1 <-> <-> IPBX-A <-> IPphones IPphones <-> IPBX-2 <-> KAMAILIO <-> IPBX-B <-> IPphones ... IPphones <-> IPBX-N <-> IPBX-X <-> IPphones version: kamailio 1.4.1-notls (i386/linux) I experienced problems regarding DTMF digits transport. DTMF is transported through RTP data and it is not recognized at the end. Does anyone could recommend me how to force the negotiation between Kamailio and IPBXs to be sure that IPphones and IPBXs will use another method to transport DTMF (like OOB SIP INFO or RFC2833) Many thanks, Simon -- -------------------------------------------------------------- Simon Muyal Suivi des Services aux Usagers (SSU) Services operations & support Réseau National de Télécommunications pour la Technologie, l'Enseignement et la Recherche Tel: +33 (0) 1 53 94 20 40 | GIP RENATER Fax: +33 (0) 1 53 94 20 41 | c/o ENSAM E-mail: muyal(a)renater.fr | 151 Boulevard de l'Hôpital http://www.renater.fr | 75013 PARIS --------------------------------------------------------------

2 1

[Kamailio-Users] Load Based Load Balancing
by Kareem Hamdy 20 May '09

20 May '09

Hello: I've been going through your lists and your documentation, and I'm unable to find how you set up load based load balancing. To further illustrate: For example, let's say we have the following live calls:1,2,3,4,5,6,7 Round Robin Works Well ---------------------- +-----+ | SER | +-----+ | | +--------------+ |-------| FreeSWITCH 1 | [Calls 1,3,5,7] | +--------------+ | | +--------------+ |-------| FreeSWITCH 2 | [Calls 2,4,6] +--------------+ But let's say some of the calls end significantly quicker than the others, how can I send calls by their load (current calls, cpu, etc) For example, let's say we have the following live calls:1,2,3,4,5,6,7 - Let's say calls 4 and 5 lasted about 30 seconds, while the others lasted about 5 minutes. How would I be able to route calls to the box that's least used - in this example, "FreeSWITCH 2" at any given time had the least number of calls, so calls 4,5, and 6 were dispatched to it. Load Based Load Balancing ------------------------- +-----+ | SER | +-----+ | | +--------------+ |-------| FreeSWITCH 1 | [Calls 1,3,7] | +--------------+ | | +--------------+ |-------| FreeSWITCH 2 | [Calls 2,6] +--------------+ Thanks, Kareem

2 1

[Kamailio-Users] Fwd: Debian distributions
by David 20 May '09

20 May '09

To: users(a)listskamailio.org Debian used to pack with OpenSER, does Kamailio ?

2 1

Re: [Kamailio-Users] [Kamailio-Devel] Bugs on "kamctl" related to CR??? - Moving from Devel to User List
by Edson - Lists 20 May '09

20 May '09

Hi, Ovidiu I'm replicating this trade on User list, as You suggested, but take look on what I had done: Kamailio-151:/etc/kamailio # kamdbctl reinit kamailio MySQL password for root: INFO: Database kamailio deleted INFO: test server charset INFO: creating database kamailio ... INFO: Core Kamailio tables succesfully created. Install presence related tables? (y/n): y INFO: creating presence tables into kamailio ... INFO: Presence tables succesfully created. Install tables for imc cpl siptrace domainpolicy carrierroute userblacklist htable purple? (y/n): y INFO: creating extra tables into kamailio ... INFO: Extra tables succesfully created. Kamailio-151:/etc/kamailio # kamctl cr addrt 0 'XXXX' ERROR 1146 (42S02) at line 1: Table 'kamailio.route_tree' doesn't exist ERROR: cr - SQL Error Kamailio-151:/etc/kamailio # and the error/bug persist. Even with a clean DB definition... Where could the problem be? I understand that it's on the "kamctl" script, but I can be wrong... Edson. Ovidiu Sas escreveu: > This is a question for users, not for devel. > Please install the extra tables via kamdbctl. > Take a look also at /etc/kamailio/kamctlrc where the extra tables are defined. > > > Regards, > Ovidiu Sas > > On Tue, May 19, 2009 at 4:25 PM, Edson - Lists <4lists(a)gmail.com> wrote: >> Hi, Guys... >> >> After some tests, I realize that CarrierRoute will feet better on we >> want to do, but when trying to include a carrier, I got: >> >> # kamctl cr addrt 1 XXXX >> ERROR 1146 (42S02) at line 1: Table 'kamailio.route_tree' doesn't exist >> ERROR: cr - SQL Error >> >> Then I try to include a route by hand with the commands: >> >> INSERT INTO carrier_name (carrier) VALUES ("default"); >> INSERT INTO domain_name (domain) VALUES ("default"); >> INSERT INTO carrierroute (carrier,domain,scan_prefix,flags,mask,prob,strip,rewrite_host,rewrite_prefix,rewrite_suffix,description) >> VALUES("1","1","1000","0","0","1","0","PEER-01","","","FRANCE"); >> >> And try to see the results, but I got: >> >> # kamctl cr show >> cr routing tree >> ERROR 1146 (42S02) at line 1: Table 'kamailio.route_tree' doesn't exist >> cr routes >> >> Now is this resulting from bug(s) in the "kamctl" script or from wrong >> definitions on the "kamctlrc" file? >> >> Edson. >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Devel mailing list >> Devel(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/devel >> >

3 3

[Kamailio-Users] rtpproxy error
by Chandrakant Solanki 20 May '09

20 May '09

Hi I have found following errors... Anybody how it occurs... Is there any special way to install rtpproxy..?? May 20 05:01:33 asterisk4 /sbin/kamailio[24792]: ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy May 20 05:01:33 asterisk4 /sbin/kamailio[24792]: ERROR:nathelper:send_rtpp_command: proxy <unix:/var/run/rtpproxy.sock> does not respond, disable it May 20 04:11:40 asterisk4 /sbin/kamailio[24525]: WARNING:nathelper:rtpp_test: can't get version of the RTP proxy May 20 04:11:40 asterisk4 /sbin/kamailio[24525]: WARNING:nathelper:rtpp_test: support for RTP proxy <unix:/var/run/rtpproxy.sock> has been disabled temporarily -- Regards, Chandrakant Solanki

1 0

[Kamailio-Users] new feature: event_route
by Daniel-Constantin Mierla 20 May '09

20 May '09

Hello, I just introduced support for a new type of route in configuration file - event_route. The prototype for it is: event_route[groupid:eventid] { [actions] } The main purpose for it is to allow modules (and core) to be able to execute code written in configuration file when a specific event happens, without altering the config grammar. Also, in short term, the modules that fire events should be able to impose what type of actions can be used in the respective event_route - now functions allowed in request route can be used. First module that uses this feature is htable, executing an event route only once, when all modules were initialized. A typical use case is to initialize some items in a hash table. Example: modparam("htable", "htable", "a=>size=4;") event_route[htable:mod-init] { $sht(a=>calls-to::10.10.10.10) = 0; $sht(a=>max-calls-to::10.10.10.10) = 100; } route { if(is_method("INVITE") && !has_totag()) { switch($rd) { case "10.10.10.10": lock("calls-to::10.10.10.10"); $sht(a=>calls-to::10.10.10.10) = $sht(a=>calls-to::10.10.10.10) + 1; unlock("calls-to::10.10.10.10"); if($sht(a=>calls-to::10.10.10.10)>$sht(a=>max-calls-to::10.10.10.10)) { sl_send_reply("500", "To many calls to .10"); exit; } break; ... } } } This system will be used to get kamailio's error_route and local_route functionalities. Other cases I have in mind now are: - auto-expired dialog route introduced in kamailio 1.5.0 to become event_route[dialog:auto-expired] - call event route when a location record has expired on timer: event_route[usrloc:auto-expired] making available via PV details of expired contact - rtimer module routes to become event routes Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Kamailio-Users] XCAP_CLIENT failing to fetch/parse resource-lists document
by Pascal Maugeri 20 May '09

20 May '09

I see that xcap_client is failing to parse a document from the XCAP server: May 20 12:41:47 sandbox /usr/sbin/kamailio[11280]: ERROR:xcap_client:send_http_get: while performing curl option May 20 12:41:47 sandbox /usr/sbin/kamailio[11280]: ERROR:rls:get_resource_list: while fetching data from xcap server May 20 12:41:47 sandbox /usr/sbin/kamailio[11280]: ERROR:rls:rls_handle_subscribe: while attepmting to get a resource list May 20 12:41:47 sandbox /usr/sbin/kamailio[11280]: ERROR:rls:rls_handle_subscribe: occured in rls_handle_subscribe I know the document is retrieved by the XCAP client (I see the HTTP traffic). The document content is very simple: <?xml version="1.0" encoding="UTF-8"?> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> </resource-lists> >From that point I'm a bit confused on how to troubleshoot this issue. How could I get more log from curl ? Regards, Pascal

2 1

[Kamailio-Users] auth to a provider ?
by BERGANZ François 20 May '09

20 May '09

Hello all, I need that Kamailio auth to my provider how to auth kamailio to my provider? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 1

[Kamailio-Users] Kamailio Callgroup
by Andre Buchwald 19 May '09

19 May '09

Hi, is it possible to configure a callgroup by Kamailio? UA_1 calls UA_2 that is a callgroup-member. In this case all members receive "invite" from UA_1 and if one member accepts the "invite" all other invites should be cancelled. Thanks

3 2

[Kamailio-Users] Realtime Integration Of Asterisk 1.4 With Kamailio 1.5.x
by Gavin Henry 19 May '09

19 May '09

This has just appeared on voip-info via my Google Alerts. It will answer a *lot* of questions that appear on the list: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+Wit… (Written by the Daniel so you know it works!) Cheers.

2 1

Re: [Kamailio-Users] VoiceMail MWI
by jh＠tutpro.com 19 May '09

19 May '09

Chandrakant Solanki writes: > But how can i use... can i have example or .cfg file for same... you send publish requests to kamailio using mi/xmlrpc commands. there is nothing to configure there. in addition, you of course need to process mwi subscribe requests from your users. -- juha

1 0

[Serusers] VoiceMail MWI
by Chandrakant Solanki 19 May '09

19 May '09

Hi all, I am newbie to kamailio and asterisk... I am using ... *Asterisk 1.6.0.5 on CentOS 5* (172.18.100.65) *kamailio 1.5.0-notls on CentOS 5* (172.16.0.10) I am successfully registered my sip phone with kamailio server and call were flow from one number to another. I have multiple Asterisk Server and one Kamailio server... when user dial number, but unable to reach he/she can put voicemail for that user on asterisk server where he/she can called... After a few minutes (say 10 minutes) i will execute one script in cron (every asterisk server) that will find voice mail from and put it on centeralized voicemail server, which is same as above configuration... How can i use MWI ... to indicate user that he/she has voicemaili in his/her mailbox... What changes should be requiered at kamailio site as well as asterisk side... Plz help me out. Thanks in advance -- Regards, Chandrakant Solanki

3 3

[Serusers] FreeRADIUS authentication via radiusclient-ng
by Leon Li 19 May '09

19 May '09

Hi, I installed SER and FreeRADIUS, radiusclient-ng. However, SER seems never talk with FreeRADIUS, as I found no RADIUS auth logs. After digging radiusclient.conf, I found it calls FreeRADIUS using a script (/usr/local/sbin/login.radius), but this script does nothing only output as below. $ sudo /usr/local/sbin/login.radius This is the dummy login.radius script. If you want that this script does something useful you'll have to replace it. The following RADIUS environment variables are set: Bye, bye. How can I configure radiusclient-ng to call FreeRADIUS when there is an authentication request? Thanks Leon

1 0

[Kamailio-Users] RLS with XCAP server - document changes notification
by Pascal Maugeri 18 May '09

18 May '09

Hi I am planning to use the RLS module with an external XCAP server (integrated_xcap_server=0) to store users resource-lists documents. My question is related to the changes of a such resource-lists document and how the RLS server behaves. I've read carrefully the module documentation<http://kamailio.org/docs/modules/1.5.x/rls.html#id2469762>but this point remains unclear. Let's say that user U1 has subscribed to her resource-lists, stored in a XCAP server, that contains contacts U2 and U3. U1 receives notifications of status updates of U2 and U3. If U1 adds a new contact U4 to the resource-lists (eg. using XCAP), does the RLS detects this change and subscribes to U4 status updates ? Regards, Pascal

2 3

[Serusers] Warning: run_failure_handlers - ser rel2_0_0
by Tomasz Zieleniewski 18 May '09

18 May '09

Hi I quite often receive such warning: ser: Warning: run_failure_handlers: no negative handler (0, 166) What does it mean? Should I bother at all? Thank You in advance. Best Regards, - Tomasz Zieleniewski

2 1

[Kamailio-Users] error : CRITICAL **: pecan_stream_write_full: assertion `stream' failed in kamailio
by rachid 15 May '09

15 May '09

Hello, I use kamailio to communicate between a sip client and a MSN client but when a sip client send a message to MSN client, the kamailio print me this message:CRITICAL **: pecan_stream_write_full: assertion `stream' failed someone can tell me where is the probleme? thank you -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

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[Kamailio-Users] early media and first SIP/Kamailio Configuration
by bahanfar＠hotmail.de 15 May '09

15 May '09

Hi, I am newcomer in SIP and Kamailio Wold an have a question. I have two IP Telefones and would like to realize with Kamailio an IP Communication. The Problem: a. The UA 1 calls UA 2 b. UA 2 start recording the video form UA 1, before a SIP call is established (early media) How can I do that? Can anybody help me or can give me any Info about example application. Thank you very much. Bahanfar _________________________________________________________________ http://redirect.gimas.net/?n=M0905xIE8_MSN4 Internet Explorer 8 jetzt für MSN - kostenlos!

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Re: [Kamailio-Users] kamailio multi-domain
by karhu 15 May '09

15 May '09

>You mean you've added the users to the subscriber table? Yes i added User1 and User2 in the subscriber's mysql table. Xlite conf : usr : User1 psw : xxxx Domain : 10.10.10.2 usr: User2 psw : xxxx Domain : 192.168.0.2 In mysql table i added : User2 with xxx psw and 192.168.0.2 domain. (don't really know if it is the righ way...!?) It seems the registering request are fine, below are the wireshark captures for user1 and user2 : http://www.nabble.com/file/p23555745/User1_Register.jpg http://www.nabble.com/file/p23555745/User2_Register.jpg Etheir i try to call from User2 to User1 or User1 to User2 i got nothing. On XliteUser2 i tried for the call : User1(a)10.10.10.2 >You've enabled auth in the cfg given below, so this is the expected behaviour. >Your UA should reply with the proper credentials, then the call should be >established. Take a look to the log file of the kamailio, and do a sip trace >(e.g. with ngrep -W byline port 5060) to get more information about the >eventual problem. Below ngrep -W byline port 5060 capture, User1 call User2 INVITE sip:boby@10.10.10.2 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.3:41394;branch=z9hG4bK-d87543-18094a3f30346a6c-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1000@10.10.10.3:41394>. To: "boby"<sip:boby@10.10.10.2>. From: "toto"<sip:1000@10.10.10.2>;tag=89767174. Call-ID: MzE3NDc5YzMwNDI2YjdjMWE4Zjg2Y2RlYTQ3YTZiMzY.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="1000",realm="10.10.10.2",nonce="4a0d276f000000535a5a47a45038d6cbfab9392dd8d576cf",uri="sip:boby@10.10.10.2",response="87afe587be968300f615006a989c9c7f",algorithm=MD5. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 534. . v=0. o=- 2 2 IN IP4 10.10.10.3. s=CounterPath X-Lite 3.0. c=IN IP4 10.10.10.3. t=0 0. m=audio 20562 RTP/AVP 107 119 0 98 8 3 101. a=alt:1 1 : wbrcVqeO gWxBJcbu 10.10.10.3 20562. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. m=video 31202 RTP/AVP 115 34. a=alt:1 1 : Ph3NZX4n 9x8Vo2nU 10.10.10.3 31202. a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960. a=fmtp:34 QCIF=1 MAXBR=1960. a=rtpmap:115 H263-1998/90000. a=rtpmap:34 H263/90000. a=sendrecv. # U 10.10.10.2:5060 -> 10.10.10.3:41394 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 10.10.10.3:41394;branch=z9hG4bK-d87543-18094a3f30346a6c-1--d87543-;rport=41394. To: "boby"<sip:boby@10.10.10.2>;tag=0c6c077910af0e0627388e55f0ad5733-3061. From: "toto"<sip:1000@10.10.10.2>;tag=89767174. Call-ID: MzE3NDc5YzMwNDI2YjdjMWE4Zjg2Y2RlYTQ3YTZiMzY.. CSeq: 2 INVITE. Server: Kamailio (1.5.0-notls (x86_64/linux)). Content-Length: 0. . # U 10.10.10.3:41394 -> 10.10.10.2:5060 ACK sip:boby@10.10.10.2 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.3:41394;branch=z9hG4bK-d87543-18094a3f30346a6c-1--d87543-;rport. To: "boby"<sip:boby@10.10.10.2>;tag=0c6c077910af0e0627388e55f0ad5733-3061. From: "toto"<sip:1000@10.10.10.2>;tag=89767174. Call-ID: MzE3NDc5YzMwNDI2YjdjMWE4Zjg2Y2RlYTQ3YTZiMzY.. CSeq: 2 ACK. Content-Length: 0. . Cheers, Karhu -- View this message in context: http://www.nabble.com/kamailio-multi-domain-tp23539467p23555745.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Error when connecting to mysql
by Alex R.S.M 15 May '09

15 May '09

Hi, I am getting the following error when kamailio wants to connect to mysql within a simple perl program. "mysql" module is already loaded in the config file. Is there any modparam for mysql that has to be set? ERROR:core:XS_OpenSER__Message_log: perl error: Can't locate object method "connect" via package "DBI" at /etc/openser/test.pl line 17. simple perl program: #!/usr/bin/perl use OpenSER; use OpenSER::Constants; sub run { ??? } 1 Thanks, A. R. _________________________________________________________________ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutor…

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[Kamailio-Users] xmpp:cmd_send_message: invalid content-type 0x100ff
by Joao Gomes Pereira 15 May '09

15 May '09

Hello to all Im trying to use Kamailio with xmpp module in Component Mode. So, I have Jabberd2 installed and running in the same machine, and this is my config: loadmodule "xmpp.so" modparam("xmpp", "domain_separator", "*") modparam("xmpp", "gateway_domain", "jabber.servebeer.com") modparam("xmpp", "xmpp_domain", "jabber.servebeer.com") modparam("xmpp", "xmpp_host", "my_server_IP") modparam("xmpp", "xmpp_port", 5347) if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } but I'm receiving this errors: kamailio[8218]: ERROR:xmpp:cmd_send_message: invalid content-type 0x300ff What could be the problem? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

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[Kamailio-Users] sr-users mailing list
by Daniel-Constantin Mierla 15 May '09

15 May '09

Hello, the users mailing list for sip router project is available as: sr-users(a)lists.sip-router.org. you can subscribe to it at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users People starting testing sip router and willing to help with documentation are invited to join. The other mailing lists dedicated to sip router project are: - sr-dev - for development related discussions http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev - sr-users-es - sip router Spanish community mailing list http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] kamailio multi-domain
by karhu 14 May '09

14 May '09

Hello, First of all i would like to say i am a newbee in Kamailio :). Here is my network configuration : User 1 User2 10.10.10.3 10.10.10.2 192.168.0.2 192.168.0.10 -----------------*KAMAILIO*----------------- User1 and User2 are registered in mysql. When i installed Kamailio, i edited kmctlrc to have SIP_DOMAIN = 10.10.10.2. I am using XLite Siphone for both users. When User 1 is registering to Kamailio there is no problem, but when User2 try to do the same i get 407 Proxy Authentication Required.. What i would like to do is User1 call User2 trough kamailio. Obviously i activated nathelper module and started rtpproxy. Do i need to configure multi-domain module ? Here is my .conf file : ############################################# ####### Global Parameters ######### debug=9 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" #port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.168.0.2 listen=udp:10.10.10.2 ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:10.10.10.2:5060") # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7890") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } ############################################## Thanks -- View this message in context: http://www.nabble.com/kamailio-multi-domain-tp23539467p23539467.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] x-lite 3.0 on linux
by Daniel-Constantin Mierla 14 May '09

14 May '09

Hello, during last weeks while testing the kamailio (openser) modules with sip-router core I got x-lite 3.0 working on linux (ubuntu) with wine 1.1.20. I posted guidelines at: http://appdb.winehq.org/objectManager.php?sClass=version&iId=16413 Voice, IM&Presence seem to work just fine. Video was not tested as I couldn't make any wine application see the video cam, although it is working fine on linux with skype or ekiga. For me it is a great tool to test the presence, hope it will be useful for some of you as well. Apart of x-lite, I got other sip softphone working with wine: - pangolin - works fine, but the presence status changes are not sent to server (no PUBLISH) -- might be a bug in the software, maybe someone running it on windows can confirm it or not - zoiper communicator rc1 free edition for windows - linux version seems to be available now, but requires to register with their site Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] kamailio and/or ser 3.0 with sip router core
by Daniel-Constantin Mierla 14 May '09

14 May '09

Hello, long awaited period of SIP Router integration phase is more and more approaching the end. Apart of 4 modules that are not integrated yet from Kamailio - seas, siptrace, uac_redirect and nat_traversal (work being undertaken) -- the rest of 88 modules are ready for http://sip-router.org core. I have created a page to collect guidelines for the default Kamailio configuration file used with sip-router. I invite all of you to test it, not only this one, but your private configs as well. http://sip-router.org/wiki/migration/kamailio-3.0-config Feel free to add to the wiki page or post questions about it to sr-users(a)lists.sip-router.org. This is a new list created for users of sip-router project, you can subscribe to it at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users At this stage, sip router can be used with modules from both Kamailio (OpenSER) and SER projects, mixed in same instance at the same time. Therefore you can choose the best ones that fit your needs. For Kamailio (OpenSER) users there is a page that tries to collect new features they got from SER side (still a lot to add there, hope ser developers will contribute what they find missing). Another page collects links for ser and kamailio users to find what the other project brings: http://sip-router.org/wiki/migration/new-features-for-k-users http://sip-router.org/wiki/features/inherited-features It was not only integration work in the past months, sip router has quite a bunch of brand new feature. The wiki page to collect the summary is available at: http://sip-router.org/wiki/features/new-in-devel Jan just set online documentation portal, available for web browsing: http://sip-router.org/docbook/sip-router/branch/master/ Common modules are located in directory 'modules' while kamailio and ser specific modules can be found under module_k and modules_s. Project's wiki portal got also good content and it is under heavy work -- note that everyone can contribute to wiki: http://sip-router.org/wiki Couple of core and tm features are not yet integrated or not exactly as they were in Kamailio. Please report the ones you are missing in your config at: http://sip-router.org/wiki/devel/kamailio-integration#tm_and_core_extensions It is the place that should set the priority rank for integration. I hope that in 1-2 months time frame to get into beta stage and prepare a stable release. Many thanks for those that allocated resources to make things evolve to this stage and many thanks in advance for those that will start testing and help improving the sip router project. Beware that is still very early stage and not recommended for production. Cheers, Daniel PS. Codename 3.0 refers to Kamailio (OpenSER) or SER using sip router core, there is not a stable release yet. -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] dialplan subst issue
by Andreas Granig 14 May '09

14 May '09

Hi all, I was playing around with dialplan module (kam 1.4) for the first time and it seems that the repl_exp doesn't handle back references correctly: mysql root@kamailio> select * from dialplan\G *************************** 1. row *************************** id: 1 dpid: 1 pr: 1 match_op: 1 match_exp: @voicebox\.local$ match_len: 0 subst_exp: ^sip:(.*)$ repl_exp: sip:xx_\1_yy attrs: voicebox 1 row in set (0.00 sec) In my config, I do: dp_translate("1", "$ru/$var(dptest)"); xlog("L_INFO", "DP result='$var(dptest)', attr='$avp(s:dialplan_attr)\n"); and the result is: DP result='sip:xx_user@voicebox.local', attr='voicebox' So the string after the back reference ("_yy" in this case) is not inserted. Any ideas? Andreas

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[Kamailio-Users] presence pres-rules and database watchers table
by Dubravko Caric 13 May '09

13 May '09

Hi all, is there any way to automatically change "reason" column from "rejected" back to "NULL" or "allow" in watchers table after UAC#1 moves the UAC#2 from its blacklist to whitelist? i see that kamailio (1.5.0) correctly reads index.xml file from database, but "rejected" reason is still in the watcher table until I delete it manually and restart UAC. ...and is this even suppose to work like that or should I use different approach? Thanks Dubravko

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[Kamailio-Users] kamailio
by Abou NDIAYE 13 May '09

13 May '09

Hello, In fact I started with kamailio. The calls go smoothly on my router (with NAT) kamailio that the proxy is started or not, the calls pass. I put what in the conf file for the proxy intercepts all calls sip and then forward. Thank you in advance. _________________________________________________________________ Découvrez Windows Live Spaces et créez votre site Web perso en quelques clics ! http://spaces.live.com/signup.aspx

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[Kamailio-Users] Get send_socket if I have no new branch
by catalina oancea 13 May '09

13 May '09

Hello all In my kamailio.cfg I don't use append_branch at all and I listen on multiple addresses. Before running t_relay I need to know on which socket will the request be sent. I don't want to change the socket, but I want to know which socket is chosen. I tried $(branch(send_socket)[0]) and $(branch(send_socket)) but I got error "ERROR:pv:pv_get_branchx: error accessing branch [0]" Does anybody know how I can get this info? Thanks Catalina

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Re: [Kamailio-Users] two UACs behind same NAT
by Dubravko Caric 13 May '09

13 May '09

Hi, ----- Original Message ---- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Iñaki Baz Castillo <ibc(a)aliax.net> Cc: users(a)lists.kamailio.org Sent: Monday, May 11, 2009 10:04:49 AM Subject: Re: [Kamailio-Users] two UACs behind same NAT >> >> Bad luck then... :( >> SIP ALG is the worst enemy for SIP. >> > workarounds could be: > - run sip server also on a different port than 5060 (say 5070) - > kamailio is just fine doing it. Point the users behind such ALGs to this > port. Still alg can detect it, but most of them do the detection by port > 5060 > - use TCP if the phone supports it - most of algs do not touch TCP > connections, but ... > - use TLS if the phone supports it - safest - the alg cannot touch it yes, we are planning to use TLS and I hope we won't have that kind of problems then > - recommended - send back the router and ask for money return well, I could do that but the problem is not just with this router. I saw a list of this kind of routers (small home routers) that are not recommended to use with SIP, and I think that TLS is only universal solution. Best regards Dubravko Cheers, Daniel

2 1

[Serusers] SER with RADIUS
by Leon Li 13 May '09

13 May '09

Hi, I am trying to make SER working with FreeRADIUS. However, I got an error using www_challenge, " 0(3184) parse error (119,25-26): unknown command, missing loadmodule?" The 119 line is www_challenge("", "0"); I did load auth module. Any idea? Here is my config: # # $Id: ser-basic.cfg,v 1.1.2.2 2008/06/12 13:15:06 alfredh Exp $ # # This a very basic config file w aliases and anamed route but # w/o authentication, accounting, database, multi-domain support etc. # Please refer to ser.cfg for a more complete example # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) #memdbg=10 # memory debug message level #memlog=10 # memory statistics log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* 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) listen=202.158.197.134 port=5060 children=4 #user=ser #group=ser #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # # ------------------ module loading ---------------------------------- 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/textops.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/ctl.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ctl params # by default ctl listens on unixs:/tmp/ser_ctl if no other address is # specified in modparams; this is also the default for sercmd modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") # listen on the "standard" fifo for backward compatibility modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") # listen on tcp, localhost #modparam("ctl", "binrpc", "tcp:localhost:2046") # -- auth_radius params -- modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_reply("483","Too Many Hops"); break; } if (msg:len >= max_len ) { sl_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(FORWARD); break; } if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(FORWARD); 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") { # authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save_contacts("location"); break; } # native SIP destinations are handled using our USRLOC DB if (!lookup_contacts("location")) { sl_reply("404", "Not Found"); break; } append_hf("P-hint: usrloc applied\r\n"); } route(FORWARD); } route[FORWARD] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); } } Thanks, Leon

2 7

[Kamailio-Users] BAN a scanner
by BERGANZ François 12 May '09

12 May '09

Hello, Is it possible to ban an IP which scan my servers ? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

5 5

[Kamailio-Users] WARNING:core:fm_free: free(0) called in kamailio
by rachid 12 May '09

12 May '09

Hello, when i start a sip client (x-lite) i have in kamailio debug message this error: WARNING:core:fm_free: free(0) called someone can tell me where is the problem?? thank you ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

2 1

[Kamailio-Users] RTP timeout and nathelper
by mayamatakeshi 11 May '09

11 May '09

Hello, RTPProxy 1.2.0 can be instructed to notify and external app when RTP timeout occurs. It seems this notification must be enabled on a per session basis. So, does module nathelper have the means to request this notification? regards, takeshi

1 0

[Kamailio-Users] two UACs behind same NAT
by dubravko caric 11 May '09

11 May '09

Hi all, I have a problem when using two UACs behind same NAT on Linksys WRT54GL router. Registrations, Invites, ... are handled correctly, but after 30sec in the call UACs hangs up the call (RTP goes directly between clients). This happens because UAC#1 (which answerd the call with OK) didn't recive ACK from UAC#2. This isn't happening when I have this same clinets behind NAT on PIX525 and everything is working fine. Everithing is also fine when UAC#1 calls someone on PSTN (RTP is handled by RTPProxy) or someone behind other NAT. Strange thing is that UAC#1 sends Ringing response and OK message with this information in Contact header: Contact: <sip:bob@public_ip_address:31479;rinstance=e0f8548cf68b5b6e> but proxy forwards this Ringing response and OK message to UAC#2 with this information in Contact header: Contact: <sip:bob@public_ip_address:0;rinstance=b18efeaa8a0e3f0f> so port is "0" and with ngrep on proxy I saw that this port is replaced with port "5060" and ACK is sent to the UAC#1 with this parameters public_ip_address:5060 which isn't correct because UAC#1 is using 31478-31488 port range and expects traffic on these ports. Some of debug messages that i see on proxy are: 1. sl_filter_ACK: to late to be a local ACK!, 2. e2e proxy ACK found building branch for end2end ACK totag for e2e ACK found: 1 I'm using Kamailio 1.5.0 with standard kamailio.cfg and newest XLite (with STUN and ICE options turn on). I would be greatfull if someone can give me any guidelines what to look next in order to solve this issue. Best regards Dubravko

3 6

[Kamailio-Users] creating openser db on an external mysql server
by Alex R.S.M 11 May '09

11 May '09

Hi, Creating openser database on the localhost works fine using "kamdbctl create". However, I need to create openser database on an external mysql server. I tried the following command but it's not working: kamdbctl create openser@<external_mysql_server_ip> What's the correct command? Thanks, A R _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_BR_life_in_synch_052009

4 11

[Kamailio-Users] call routing problem
by bhrugu mehta 08 May '09

08 May '09

Hi all, I am new to openser. I have register two sip user in openser (as register server) and call handling in asterisk. when 1001 user do a call to 1002 nothing happen. call rejected. If posible give a sip.conf and extension.conf snap of this scenario. any suggestion? Bhrugu Mehta

3 2

[Kamailio-Users] H.323 / SIP signalling gateway for OpenSER
by Bilbo 07 May '09

07 May '09

Hi, Has anyone tried connecting the 'Cisco Unified Border Element' to OpenSER as a H.323 /SIP signalling gateway to allow H.323 video calls to be placed over OpenSER? Can you see any problems trying to connect these two systems together? Thanks -- View this message in context: http://www.nabble.com/H.323---SIP-signalling-gateway-for-OpenSER-tp23401501… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 3

[Kamailio-Users] sip router presentation at amoocon 2009
by Daniel-Constantin Mierla 06 May '09

06 May '09

Hello, those interested in seeing the slides I presented at Amoocon 2009, can get the link to them at: http://sip-router.org/2009/05/05/amoocon-2009-presentation/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

2 1

[Kamailio-Users] H.323 signalling gateway for OpenSER
by Bilbo 06 May '09

06 May '09

Hi, Has anyone tried connecting the 'Cisco Unified Border Element' to OpenSER as a H.323 /SIP signalling gateway to allow H.323 video calls to be placed over OpenSER? Can you see any problems trying to connect these two systems together? Thanks -- View this message in context: http://www.nabble.com/H.323-signalling-gateway-for-OpenSER-tp23401428p23401… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Kamailio-Users] web interface to add new users
by Joao Gomes Pereira 04 May '09

04 May '09

Hello to all Does someone knows of a Web application to allow users to add an account in Kamailio? SIREMIS is only for administrator use. Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

2 1

[Kamailio-Users] Arbitrary contact added in 302 generated responses
by Francisco Javier Lizaran Vilches 04 May '09

04 May '09

Hi all: Have running this script on Openser 1.2.3-notls version managing forwards this way: route[3] { ... t_on_failure("1"); if avp_db_load("$ru", "*") { if (is_avp_set("$avp(s:fwdbusy)/s")) { setflag(23); }; ... failure_route[1] { ... if ((isflagset(23)) && (t_check_status("486"))) { if (avp_pushto("$ru", "$avp(s:fwdbusy)")) { t_reply("302","Moved Temporarily"); return; }; }; ... User A calls user B and B has fwdbusy parameter set in user preferences; if B is busy, proxy sends 302 message back to A with contact set to $avp(s:fwdbusy) value. This setup usually works fine. However, sometimes the proxy shows a strange behaviour. It starts to append an arbitrary contact to the contact header in every forwarding it does. The contact appended has nothing to do with A or B or the uri set in the avp. The same uri is appended in all call forwardings performed in the system. If the proxy is restarted, the problem disappears. Example: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.100.2.254:5060;branch=z9hG4bK51461DD5 From: <sip:30132@10.100.2.254 <sip%3A30132(a)10.100.2.254>>;tag=92DAB388-EE7 To: <sip:20050@domain.com <sip%3A20050(a)domain.com> >;tag=880a5593aeb097bc75600b31d6e17107-78ac Call-ID: C0C84A1C-33CE11DE-BEBEEAA9-C0D323DF(a)192.168.2.40 CSeq: 101 INVITE Contact: sip:30050@domain.com <sip%3A30050(a)domain.com>, <sip:030410@10.172.0.254:5060;transport=udp>;q=0 Server: OpenSER (1.2.3-notls (i386/linux)) Content-Length: 0 <sip:030410@10.172.0.254:5060;transport=udp>;q=0 is added in all forwardings done the system. Unfortunately I have no debug info cause it happens in production environment. I couldn't reproduce the problem in test environment. Have you got any idea on what could make it happen? Thanks a lot: Fran Lizaran

2 3

[Kamailio-Users] Http_query
by sasirekha jaganathan 04 May '09

04 May '09

Hi, Is it possible to get the XML response(i.e content other than text) in http_query function? Thanks Now surf faster and smarter ! Check out the new Firefox 3 - Yahoo! Edition http://downloads.yahoo.com/in/firefox/?fr=om_email_firefox

3 2

[Kamailio-Users] Callback functionality
by sasirekha jaganathan 04 May '09

04 May '09

Hi, Is it possible to implement a custom callback in kamailio modules? Thanks Now surf faster and smarter ! Check out the new Firefox 3 - Yahoo! Edition http://downloads.yahoo.com/in/firefox/?fr=om_email_firefox

3 2

[Serusers] ser problem
by Akhmad Dhany 04 May '09

04 May '09

Hi,, I send my problem in attach files. Thanks very much. Regards, Dhany

2 1

[Kamailio-Users] communication between SIP and MSN users without using purplemap table
by rachid 04 May '09

04 May '09

Hello , I use kamailio to have communication between sip client(bob) and MSN client (alice) and i added msn address in bob budy liste(in x-lite) so i want to have bidirectional communication without using purplemap !! someone can help me or tel me witch "file.c" to modifie thank you -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

3 2

[Kamailio-Users] changelog page not found
by ram 03 May '09

03 May '09

http://www.kamailio.org/mos/view/News/NewsItem/Kamailio-v1.5.1-Released//pu… The page you are trying to access does not exist. Please select a page from the main menu. This is from page404.php Ram

2 1

[Kamailio-Users] Multiple SIP Proxy Environment / Socket Information
by Brandon Armstead 02 May '09

02 May '09

Hello guys, Is there a method upon using lookup("location") to also pull out the "socket" information for the original location the UAC registered to, for scenarios of this example: P1 & P2 share same usrloc database. UA1 registers to P1 UA2 registers to P2 UA1 calls UA2 UA1 invites -> P1 -> INVITES -> UA2 (bypassing P2 -- where the actual nat binding is). Now upon P1 looking up usrloc for UA2, I would like to recognize that P1 is not the Proxy to deliver the call, and forward the request to P2 to send to UA2. So currently I have: UA1 INVITE -> P1 INVITE -> UA2 I wish to have: UA1 INVITE -> P1 INVITE -> P2 INVITE -> UA2 Is there an easy method to do this? I have been looking at the new nat traversal module it looks like it is doable with this (any further input as far as that?). Also is it possible with the classic Nat Helper module? Any input is appreciated, thanks!

4 23

[Kamailio-Users] CDR / EDR Accounting With Kamailio
by Brandon Armstead 02 May '09

02 May '09

Hello, First off going to thank you guys ahead of time, I'm currently running into the following issue: I'm trying to account for internal calls, i.e. example scenarios such as: UA1 and UA2 are both on P1 UA1 calls UA2 so, effectively: UA1 -> P1 -> UA2 Now the problem I'm running into is with the direction of the calls, I am having trouble accounting for both sides of the call, i.e. I store an avp such as: $avp(s:account-id) Obviously the AVP is going to be set to one account_id, and my problem is that the Callee is what ends up "being accounted" for not, the "Caller". Another scenario I would like some insight on and having problems with is: UA1 is on P1, UA2 is on P2 UA1 calls P1 -> P2 -> UA2 same type scenario, only one end of the call (Callee is being accounted for) -- how can I accurately account using kamailio acc module for both sides of the call for both users? Any thought / input on this is very much appreciated, thanks!

3 2

[Serusers] SER can't connect to mysql
by Paul Osier 01 May '09

01 May '09

Hi. When starting SER, I get the following line in the logs: new_connection(): Access denied for user 'ser'@'localhost' (using password: YES) I have the following line in ser.cfg: modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") along with the required modules loaded. I can log into mysql with mysql -h localhost -u ser -pheslo ser , so I think all the mysql stuff is set right and the parameter has the correct info. Any ideas why it's not connecting with the ser.cfg file setup? Thanks.

1 0

[Kamailio-Users] LCR Private Memory
by Stagg Shelton 01 May '09

01 May '09

I am working to get LCR implemented on my migration system, and when I try to start kamailio i find the below information in the log file, and kamailio fails to start. My lcr table has 30667 rows in it. I have read about changing the config.h file to increase the private memory, but my question is how much should I allocate. I forsee going up to more than 200,000 routes within the next 12 month period. I am currently working towards migrating up from openser 1.3.3 to kamailio-1.4.4-notls. I am currently not using LCR in my production openser 1.3.3 system. This is new functionality that we are implementing as part of the upgrade. Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_convert_row: no private memory left Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_convert_rows: error while converting row #2808 Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:db_mysql:db_mysql_store_result: error while converting result Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:core:db_do_query: error while storing result Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:lcr:reload_gws: Failed to query lcr data Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: CRITICAL:lcr:mod_init: Failed to reload gateways and routes Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:core:init_mod: failed to initialize module lcr Apr 29 17:13:49 n01-sip-c61-56-atl /sbin/kamailio[5372]: ERROR:core:main: error while initializing modules Thanks Stagg

5 11

[Kamailio-Users] SIREMIS installation
by Nikolaos Markolefas 30 Apr '09

30 Apr '09

Hi all, just installed SIREMIS on a Ubuntu server, on top of kamailio 1.5. It seems to work alright, but for the charts sections. What I see is : """ Open Flash Chart JSON Parse Error [Syntax Error] Error at character 2, line 2: 0: 1: <div style='font-size: 12px; color: blue; font-family:Arial; font-weight:bold;'>[2009-04-09 12:40:1 """ Apache produces the following log (error.log): *[Thu Apr 09 15:39:58 2009] [notice] Apache/2.2.9 (Ubuntu) PHP/5.2.6-2ubuntu4 with Suhosin-Patch configured -- resuming normal operations [Thu Apr 09 15:40:07 2009] [error] [client 192.168.110.25] File does not exist: /var/www/siremis-0.9.2/siremis-web/js/sound.js, referer: http://v-srv-obama/siremis/bin/controller.php?view=main.CrtMainView [Thu Apr 09 15:40:09 2009] [error] [client 192.168.110.25] File does not exist: /var/www/siremis-0.9.2/siremis-web/js/sound.js, referer: http://v-srv-obama/siremis/bin/charts.php?cg=system * The file indeed does not exist. I'd also like to note that while i downloaded the 0.9.2 version, SIREMIS homepage shows v0.9.1 . Any ideas?

3 9

[Kamailio-Users] from replace
by Gentrice's kaiser 30 Apr '09

30 Apr '09

Dear Sir: I wrote a failure route for serial call mechanism, it help for no answer call ... but all these serial calls need to have a new CallerID, so I use replace_from for this purpose. modparam("uac","from_restore_mode","auto") failure_route[3] { log(1,"******* I'm failure_route[3] *******\n"); if (avp_pushto("$ru", "$avp(forward)")) { append_branch(); avp_delete("$avp(forward)"); uac_replace_from("sip:1234@$fd"); # change from header t_on_failure("3"); # try next , recursive t_relay(); } } If only one forward number in AVP array , this code working fine, the issue will happen if multiple forward number trying, ( forward 2nd,3rd ...number) the UAC module will restore from header in 200 OK of its following message when a UA answered, we will have a duplicated "from header" in 200 OK (sending out opensips). It means we will have 2 "from" headers (in 200 OK) to caller, if 2nd forwarded callee answer. Caller ---> Invite ---> opensips ----> UA 0, no answer ----> UA 1, no answer ---->UA 2, answer <----200 OK <---200OK<--- this 200ok will have 2 "from" header. Most of UA just ignore redundant from header, but some can not. I think it should be UAC modules issue, anyone can help me? best regards KK

3 7

[Serusers] SIP Diversion for SER 0.9.6
by inge 30 Apr '09

30 Apr '09

Hi all, I look for a way to use SIP DIVERSION. In case of a call transfer, I have to conserve the original called number and send it to SS7 gateway. But for a good interoperability between SIP and SS7 in this kind of application I just see the way of SIP DIVERSION support. Can someone help me ? Regards, A.L

3 5

[Kamailio-Users] Kamailio v1.5.1 Released
by Daniel-Constantin Mierla 30 Apr '09

30 Apr '09

Kamailio v1.5.1 is out - a minor release of the branch 1.5, including fixes since v1.5.0 - configuration file and database compatibility is preserved. Kamailio 1.5.1 is based on the latest version of branch 1.5, including many fixes in code and documentation, therefore those running 1.5.0 are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.5.1/src/ <http://www.kamailio.org/pub/kamailio/1.5.1/src> Detailed changelog: http://www.kamailio.org/pub/kamailio/1.5.1/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.5.1/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Binaries and packages will be uploaded at: http://www.kamailio.org/pub/kamailio/1.5.1/ Modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ <http://www.kamailio.org/docs/modules/1.5.x> What is new in 1.5.x release series is summarized in the announcement of v1.5.0: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes /Note: Kamailio is the new name of OpenSER project. First version under Kamailio name was 1.4.0. Older versions will continue to use OpenSER name. / -- Daniel-Constantin Mierla http://www.asipto.com/

3 2

[Kamailio-Users] insert_hf not persistant
by David Aldworth 30 Apr '09

30 Apr '09

I think I hijacked a thread when I posted this previously. Sorry 'bout that. The issue we are having is where Kamailio relays an INVITE with the correct To header field domain, however when it relays the subsequent CANCEL it seems like it does not correctly change the To header field domain. Here is what we use to try and put in the correct To header in the CANCEL: remove_hf("To"); if ($tt == null) insert_hf("To: <sip:$rU@1.2.3.4>\r\n", "From"); else insert_hf("To: <sip:$rU@1.2.3.4>;tag=$tt\r\n", "From"); xlog("DBG1: Trying to fix CANCEL: td=$td ci=$ci rm=$rm ru=$ru ou=$ou\n"); Changes to the left side of the at-symbol persist, however changes to the domain are reverted and changed back to what was received from adjacent proxy. Any ideas why it reverts?

2 1

[Kamailio-Users] openser CDR Diversion calls
by Ilya Pichugin 29 Apr '09

29 Apr '09

Hi all! I'm new in openSER and I'm trying to get correct CDRs when the call is forwarded on busy, noanswer and uncond., but i have some troubles :( I use accounting flags by acc module and radius - so, I can get cdrs from acc table or raddact table When I do any type of FWD call by AVP module, for example, A (pstn) -> B (ser user, fwd to GSM) -> C (GSM) i get cdrs for A -> C, but I want get cdrs for A - B and B - C Example: mysql> SELECT * FROM openser.usr_preferences u; +----+---------+----------+------------+------------+------+-------------+---------------------+ | id | uuid | username | domain | attribute | type | value | last_modified | +----+---------+----------+------------+------------+------+-------------+---------------------+ | 12 | 3097999 | 3097999 | xx.xxx.com | fwdall | 0 | 89112820419 | 0000-00-00 00:01:00 | Make call 8123364000 (pstn) - 3097999 (SERuser fwdall to GSM) - 89112820419 (GSM) and CDRs from raddact table: mysql> SELECT callingstationid,canonicalURI, acctstarttime,acctstoptime, acctsessiontime FROM radacct r where acctstarttime>'2009-04-23' and callingstationid like '8123364000'; +------------------+--------------+---------------------+---------------------+-----------------+ | callingstationid | canonicalURI | acctstarttime | acctstoptime | acctsessiontime | +------------------+--------------+---------------------+---------------------+-----------------+ | 8123364000 | 89112820419 | 2009-04-24 10:29:10 | 2009-04-24 10:29:12 | 2 | +------------------+--------------+---------------------+---------------------+-----------------+ from acc table: mysql> SELECT id,method,sip_code,sip_reason,time,src_user,dst_user FROM openser.acc a; +------+--------+----------+------------+---------------------+------------+-------------+ | id | method | sip_code | sip_reason | time | src_user | dst_user | +------+--------+----------+------------+---------------------+------------+-------------+ | 2420 | INVITE | 200 | OK | 2009-04-24 10:29:10 | 8123364000 | 89112820419 | | 2421 | ACK | 200 | OK | 2009-04-24 10:29:10 | 8123364000 | 89112820419 | | 2422 | BYE | 200 | OK | 2009-04-24 10:29:12 | 3097999 | 8123364000 | +------+--------+----------+------------+---------------------+------------+-------------+ So, how do you bill diversion calls? Help, pls. -- Best Regards, Ilya Pichugin i.pichugin(a)ptl.ru

3 2

[Kamailio-Users] \0 in SIP-Messsages - uac_replac_from - Lumps - etc...
by Carsten Bock 29 Apr '09

29 Apr '09

Hi, short question: I've noticed, when the UAC-Module (from 1.3) replaces the From-Headerin subsequential requests, the From-Header is filled up with "zeros". For most clients this is absolutely fine (tested with Snom's, AVM's, Cisco's and many more). I've noticed, that some Zyxel devices however do not like this and ignore the request in this case... Is this somehow defined in the RFC? Can somebody give me hint on a chapter or something? Or does it break the RFC? Thanks in advance, Carsten

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[Kamailio-Users] insert_hf not persistant
by David Aldworth 29 Apr '09

29 Apr '09

The issue we are having is where Kamailio relays an INVITE with the correct To header field domain, however when it relays the subsequent CANCEL it seems like it does not correctly change the To header field domain. Here is what we use to try and put in the correct To header in the CANCEL: remove_hf("To"); if ($tt == null) insert_hf("To: <sip:$rU@1.2.3.4>\r\n", "From"); else insert_hf("To: <sip:$rU@1.2.3.4>;tag=$tt\r\n", "From"); xlog("DBG1: Trying to fix CANCEL: td=$td ci=$ci rm= $rm ru=$ru ou=$ou\n"); Changes to the left side of the at-symbol persist, however changes to the domain are reverted and changed back to what was received from adjacent proxy. Any ideas why it reverts?

1 0

[Kamailio-Users] Amoocon 2009
by Daniel-Constantin Mierla 29 Apr '09

29 Apr '09

Amoocon (former AsteriskTag) 2009 will take place in Rostock, Germany, May 04-05, 2009. Kamailio (OpenSER) and SIP-Router.org projects will be presented at the event by talks and case studies of: * Daniel-Constatin Mierla: http://www.amoocon.de/users/14 * Olle E. Johansson: http://www.amoocon.de/users/61 If you are around Berlin the days before the event or in Rostock during the event and want to meet, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Serusers] SER proxy compatibility with Red Hat Enterprise Linux 5
by inge 28 Apr '09

28 Apr '09

Hi, I want to buy a RHEL5 to install a SER Proxy 0.9.6. I want to know if this version is compatible whith the RHEL 5. Thanks.

2 1

[Kamailio-Users] sharing vedio between sip (x-lite) and MSN
by rachid 28 Apr '09

28 Apr '09

Hello all, I instaled kamailio and i want to have sharing video between a sip user (X-lite) and MSN user someone can help me please? -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

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Re: [Kamailio-Users] Load Balancing Asterisk using Kamailio
by Daniel-Constantin Mierla 27 Apr '09

27 Apr '09

Hello, not cc-ing to mailing list could lead to no answer... private messages from mailing list threads are ignored. Other benefits with mailing lists is that others can answer quicker or better. Thanks, Daniel On 04/27/2009 04:52 PM, Kurt Weasel wrote: > Hi Daniel, thank you for the reply. > I have some more questions to ask. > > 1. Since the dial plan is configured exactly on Asterisk (at > etc/asterisk/sip.conf and /etc/asterisk/extension.conf), do we need to > enable MySQL for Kamailio installation? Or the default installation > configuration will do (without additional dialplan or MySQL setting)? > My idea is only using Kamailio as pure load balancer so we hope to do > this as simple as possible without degrading the function. > > 2. I assume using Kamailio dispatcher module will forward some of the > users to asterik server 1 and some of them to asterisk server 2. Then > if user A is registered with Asterisk server 1 and user B is > registered with Asterisk server 2, can user A call user B? Without > Kamailio, the answer is no since they are in different Asterisk > server. With Kamailio, I am expecting both user A and B can call each > other although they are registered with different servers. Is my > configuration enough to achieve this function? Or do we need some > additional setup for this? > > Thanks again. > > Best Regards, > Kurt > > On Mon, Apr 27, 2009 at 8:02 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 04/27/2009 07:10 AM, Kurt Weasel wrote: > > Hi, > > I searched around the web to load balance asterisk servers and > found Kamailio for possible solution. Let's say I have two > identical asterisk servers with same dialplan and > configuration and I want both servers look like they have same > IP address from clients. Based on my understanding, we need 1 > Kamailio as load balancer and 2 Asterisk servers as the real > servers. Let's say the setup is : > > Kamailio load balancer 192.168.2.1 > Asterisk Server #1 192.168.2.2 > Asterisk Server #2 192.168.2.3 > > My question is, X-Lite softphone Configuration should be set > to domain 192.168.2.1, right? > > I also want to know the step by step configuration to set > kamailio as load balancer. I have not used Kamailio before. > However after searching the documentation, the step (based on > my understanding) is somewhat like this : > > 1. Install Kamailio. I will use the step by step here : > http://kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn > > 2. Then, using dipatcher module, I will configure it using > this guide here : > http://kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha > Then modify the dispatcher.list file to match the IP address > of my asterisk servers : > > /1 sip:192.168.2.2:5060 <http://192.168.2.2:5060> > <http://192.168.2.2:5060> > 1 sip:192.168.2.3:5060 <http://192.168.2.3:5060> > <http://192.168.2.3:5060>/ > > > Am I missing some steps? > Do I also need to configure dialplan or any other file at > Kamailio load balancer? Or those two steps basically done it > all for simple load balancing configuration? Thanks for your > responses. > > these steps are ok if you do not deal with NAT. If yes, then > things get a bit more complex, you need PATH support on load > balancer, but AFAIK, Asterisk does not support it -- maybe I am wrong. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Kamailio-Users] LCR gateway failthrough..
by Nick Warr 27 Apr '09

27 Apr '09

I'm using the LCR module to send traffic based on just the prefix of the number at the moment. I have two gateway groups, one with a local provider to connect locally, one with a larger one to do international calls (they can also connect locally but with a higher price).. When I call next_gw() will LCR use all possible gateway groups, or just the gateway(s) in the first matching group? The second question is tangential to the first, if it does send the traffic to other gateway group after failing through the first, does it regenerate the RURI, because the local provider wants a numerical prefix on the calls I send, and the other one no (basically do I need to account for this in my config?)

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[Serusers] SER with RADIUS
by Leon Li 27 Apr '09

27 Apr '09

Hi, Does anyone still have the instruction of SER RADIUS Howto? The link http://iptel.org/ser/ser_radius.html Found in an old thread doesn't work now. (see below for the old thread) [Serusers] ser RADIUS Howto Jan Janak jan at iptel.org <mailto:serusers%40lists.iptel.org?Subject=%5BSerusers%5D%20ser%20RADIUS %20Howto&In-Reply-To=> Fri Sep 12 02:11:33 UTC 2003 * Previous message: [Serusers] serweb main page display <http://lists.iptel.org/pipermail/serusers/2003-September/002595.html> * Next message: [Serusers] Implementing Voicemail <http://lists.iptel.org/pipermail/serusers/2003-September/002596.html> * Messages sorted by: [ date ] <http://lists.iptel.org/pipermail/serusers/2003-September/date.html#2591 > [ thread ] <http://lists.iptel.org/pipermail/serusers/2003-September/thread.html#25 91> [ subject ] <http://lists.iptel.org/pipermail/serusers/2003-September/subject.html#2 591> [ author ] <http://lists.iptel.org/pipermail/serusers/2003-September/author.html#25 91> ________________________________ Hello, I finished SER RADIUS Howto. The document thoroughly describes installation and configuration of RADIUS server and client libraries for authentication and accounting with ser. RADIUS related questions appear quite often on the mailing lists so I hope the document helps. I would appreciate if anyone using another RADIUS server than FreeRADIUS could describe the setup and send it to me (plaintext is OK, I will convert it to docbook). You can get the document from here: http://iptel.org/ser/ser_radius.html Any feedback is appreciated. Jan. Thanks Leon

1 0

[Kamailio-Users] switch to TCP when UDP message is bigger than MTU
by Sandro Bordacchini 26 Apr '09

26 Apr '09

Hi all. I am doing some test about tcp/udp transport protocols in SIP and IP fragmentation. I would like to test this behaviour, as stated in par. 18.1.1 of rfc3261: If a request is within 200 bytes of the path MTU, or if it is larger than 1300 bytes and the path MTU is unknown, the request MUST be sent using an RFC 2914 [43] congestion controlled transport protocol, such as TCP. If this causes a change in the transport protocol from the one indicated in the top Via, the value in the top Via MUST be changed. This prevents fragmentation of messages over UDP and provides congestion control for larger messages. However, implementations MUST be able to handle messages up to the maximum datagram packet size. To do this, I am running a OpenSer 1.3.2 with a "default" configuration (and very simple: no accounting, no auth, no DB) and I have two SIP phones registered on that server (for example, ext. 100 and ext. 101). When 100 calls 101, the INVITE reaches the OpenSer that appends some "dummy" headers (I have added some append_hf in the route), just to let the message be bigger than the MTU. The INVITE is routed towards 101 with UDP protocol and I have IP fragmentation. Do you have some hint to get the OpenSer work as described in the RFC (i.e. switch automatically to TCP)? Or is this only possible with UACs and not with proxies? TIA -- Ing. Sandro Bordacchini Dexgate Sr. Tech. Engineer P.zza Fermi 1 - Fornacette IT-56012 Calcinaia (Pisa) Tel. +390587424145 ext 580

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[Serusers] Error in CANCEL generation while forking - ser 2.0
by Tomasz Zieleniewski 24 Apr '09

24 Apr '09

Hi, I found an error in the SER proxy behavior. I use SER v_2_0_0 from CVS. When SER forks requests because there is a number of registered users for particular AOR, and then receives first final response, it in the standard way cancels all other transaction. Problems is that it is not compliant with RFC3261 because it doesn't add Max-Forwards header to the CANCEL requests. RFC3261 says that Max-Forwards must be present in any SIP method. CANCEL sip:16@212.180.179.42:9360 SIP/2.0 Via: SIP/2.0/UDP 212.180.179.42:7060;branch=z9hG4bKb721.ceeb7aa5.1 From: <sip:pgr@sip.rd.touk.pl <sip%3Apgr(a)sip.rd.touk.pl>>;tag=374595 Call-ID: 0da793c0ff2e7f2fc81b23e4166bbab3(a)212.180.179.42 To: <sip:tzl@sip.rd.touk.pl <sip%3Atzl(a)sip.rd.touk.pl>> CSeq: 479638 CANCEL Route: <sip:sip.rd.touk.pl:7060;lr> User-Agent: Sip EXpress router(2.0.0 (x86_64/linux)) Content-Length: 0 Kind regards, Tomasz Zieleniewski

1 0

[Kamailio-Users] textops usage
by Juan Perez 24 Apr '09

24 Apr '09

Hello I am receiving the SDP from a provider with this format: g729/8000/1, which I interpret as codec g729, sampling rate 8000 and the number of channels is 1. The calling party [linksys pa2p] apparently does not like the "1" and terminates the call. I want to strip it using textops module and pass back the SDP w/o the "1" and do this in the On_Reply and the Route sections: if (search_body("g729/8000/1")) { xlog("L_INFO","mylog: On Reply 1 section. Found the string.\n"); replace_body("g729/8000/1","g729/8000"); } but it can not even find it. Is leaving me with several questions: 1- is the "re" that I am using right? 2- Do I have to escape the "/"? 3- does the replace and search function go "deep" enough in the SDP? cheers jp

3 5

[Kamailio-Users] Problem with LCR
by Antonio Reale 24 Apr '09

24 Apr '09

Hi all, I'm using kamailio 1.4.4. In kamailio.cfg I have call forward on busy and call forward no answer that works good when I redirect the call to a local user. If I redirect the call to a PSTN number using lcr module I receive the error: ERROR:lcr:next_gw: No ruri_user AVP If I call directly the PSTN number all is OK and I see in the debug: DBG:lcr:next_gw: Added ruri_user_avp <0255XXXXXX> Where I need to set the ruri_user_avp? I just have modparam("lcr", "ruri_user_avp", "$avp(i:500)") in the script. With a very similar configuration and openser 1.2.3 the problem doesn't occur. Thanks. Regards. Antonio.

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[Kamailio-Users] bidirectionnal chating and presence between sip and MSN
by rachid 23 Apr '09

23 Apr '09

Hello all , I want to have a bidirectionnal chating and presence between sip and MSN ,but i dont know how to configure kamailio thank you for your help -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

3 2

Re: [Serusers] SER Support for SIP Diversion header
by Stefan Sayer 23 Apr '09

23 Apr '09

Hello, o Randy Jackson [04/22/09 19:49]: > Stefan, > > > > For a SER/SEMS configuration using the voicemail app, 4605 calls 4610. > 4610 doesn’t answer and call is forwarded to 9200. If the forwarded > call is routed to SER/SEMS (i.e. see example below) when the SER > (ser-2.0.0-rc1) receives the below INVITE which contains the Diversion > header in the message header portion of the request, what module and > methods do you use to retrieve the parameters from the Diversion header, > like the URI? 9200 does not have a vm account. The vm app will not > answer the call for account 4610 based on the below INVITE. Thanks in > advance. The question is: where do you get the email address from? If it's in DB I think you could get the user from the diversion header into avp with some clever textops and then do the DB query with it to get email address and user into P-App-Param header. If you are only using mode=box (only voicebox, no mail) though, it may be simpler to get the user in SEMS like with this patch to voicemail app: Index: AnswerMachine.cpp =================================================================== --- AnswerMachine.cpp (revision 1354) +++ AnswerMachine.cpp (working copy) @@ -531,10 +531,16 @@ throw AmSession::Exception(500, "voicemail: no email address"); } - user = get_header_keyvalue(iptel_app_param,"usr", "User"); + // user = get_header_keyvalue(iptel_app_param,"usr", "User"); + AmUriParser div_parser; + if (!div_parser.parse_contact(getHeader(req.hdrs, "Diversion"), 0, end)) { + ERROR("could not parse Diversion header"); + throw AmSession::Exception(500, "voicemail: no user found"); + } + user = div_parser.uri_user; if (!user.length()) user = req.user; - + sender = get_header_keyvalue(iptel_app_param, "snd", "Sender"); if (!sender.length()) sender = req.from; btw, isn't Diversion supersedet by History-Info? BR Stefan > > > > > > INVITE sip:9200@192.168.1.1;user=phone > <mailto:9200@192.168.1.1;user=phone> SIP/2.0 > > Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKac816850588;alias > > Max-Forwards: 70 > > From: <sip:4605@192.168.1.2>;tag=1c816838653 > > To: <sip:9200@_192.168.1.1_;user=phone> > > Call-ID: 816837947271200020365(a)192.168.1.2 > > CSeq: 1 INVITE > > Contact: <sip:4605@192.168.1.2;transport=tcp> > > Supported: em,100rel,timer,replaces,path,resource-priority > > Allow: > REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE > > *Diversion: <sip:4610@192.168.1.1>;reason=no-answer;screen=no;privacy=off* > > User-Agent: > > Content-Type: application/sdp > > Content-Length: 270 > > > -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

1 1

[Kamailio-Users] SR-Users-ES maillist
by Iñaki Baz Castillo 23 Apr '09

23 Apr '09

Hi, for the spanish speakers community there is a maillist about Sip-Router: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es This maillist has been merged with the previous Kamailio Users-es maillist, so the maillist history remains. Best regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

[Kamailio-Users] New listen IP without restart
by catalina oancea 23 Apr '09

23 Apr '09

Hello everybody, Does anybody know if there is a way to add a new listen line in kamailio.cfg and make kamailio start listening on that IP without restarting the server? Thanks a lot Catalina

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[Serusers] is serusers@lists.iptel.org mailing list still in use?
by Leon Li 23 Apr '09

23 Apr '09

Hi, I am just wondering whether serusers(a)lists.iptel.org is still the one to request support? I didn't received any emails in last several days. Cheers Leon

2 1

Re: [Kamailio-Users] Multiple SIP Proxy Environment / Socket Information
by Iñaki Baz Castillo 21 Apr '09

21 Apr '09

El Martes, 21 de Abril de 2009, Brandon Armstead escribió: > Inaki, Please, mantain the thread in the maillist. > Thanks for the input -- that does look similar to what I'm looking to > do, however not 100%. From looking at the RFC it looks as if there is > always a certain path being traversed, however in my case, its more of a > "this is not the server that holds this user location socket binding, lets > forward to the server that does". > > So in essence: > > UA1 can register to P1, P2, P3, and so on..... > > However it does not register to P3 via the means of UA1 REGISTER -> P1 -> > P2 -> P3 (REGISTRATION). > > It simply registers directly to P1, or P2, or P3, etc.... > > In which case upon user lookup: lookup("location"), I want to be able to > say "Hey I do not hold the socket for this registration, forward to server > that does to do the actual lookup(location) and t_relay() to the UA. Yes, it's a different scenario. You could do some trick: - When P1 receives a REGISTER it sets a bflag for it (bflag 1). - When P2 receives a REGISTER it sets a bflag for it (bflag 2). - When P3receives a REGISTER it sets a bflag for it (bflag 3). When P1 receives a request and performs location for that AoR, it will also extract the bflags, so P1 checks if bflag 1 is on. If not and bflag 2 is on, then it routes (by setting $du => without changing the resolved RURI) the request to P2. :) -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

[Kamailio-Users] Disable auto ACK
by kévin l'heveder 21 Apr '09

21 Apr '09

Hi, i want to generate my own ACK after a OK ( After ringing). but even if i drop the OK in my reply_route a ACK is auto generated by Kamailio and sent How could i prevent this behavior ?? I found some clues in the sources : *#define no_autoack(_t_) ((_t_)->flags&T_NO_AUTOACK_FLAG)* But i didn't see how i could flag a message to disable AUTOACK If someone have an idea or THE response, that would be great!! Thanks. Koon

2 1

[Kamailio-Users] configuration
by shiva prasad 21 Apr '09

21 Apr '09

Hi All, I've successfully installed kamailio 1.5x version on fedora 9 os, could you please let me know is it possible to configure Ekiga(client) to connect kamailio server,. if possible, could you please let me know how to configure the client? Thanks in advance. -- Regards, Shivaprasad BS

2 1

[Kamailio-Users] help using rewrite prefix
by Carlos A. Alvarez 20 Apr '09

20 Apr '09

I need help setting up carrierroute using the rewrite prefix and the strip function. It seem like every time I use the Rewrite Prfix function the carrierroute module will rewrite the prefix, but then I don't get certain reply messages on kamailio. At least from what I can see on the log. INFO:carrierroute:cr_do_route: uri 5555550001 was rewritten to sip:0115555550001@aslab.commx.net, carrier 1, domain 10 INFO:carrierroute:rewrite_uri_recursor: URI or prefix tree nodes empty, empty rule list INFO:carrierroute:cr_do_route: rewrite_uri_recursor doesn't complete, uri 0115555550001, carrier 1, domain 10 Is this normal behavior? I also noticed that some messages coming back from the host are not being processed by kamailio. For example on this particular call I purposely get a 604 from the server. The thing is I can't process a failure route for this message, because kamailio doesn't see is, even though it is being sent back to the server. Also when I use the Strip function is looks like carrierroute does a second lookup on the route table. In the example below I am calling a number using the prefix 551... The route table is below. +----+---------+--------+--------------+-------+------+------+-------+-----------------+----------------+----------------+------------------------+ | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description | +----+---------+--------+--------------+-------+------+------+-------+-----------------+----------------+----------------+------------------------+ | 30 | 1 | 1 | 551 | 0 | 0 | 1 | 2 | 10.10.10.1 | | | LD-1 | 31 | 1 | 1 | 1 | 0 | 0 | 1 | 0 | 10.10.10.1 | | | LD | Log for that call INFO:carrierroute:cr_do_route: uri 5515555550004 was rewritten to sip:15555550004@10.10.10.1, carrier 1, domain 1 INFO:carrierroute:cr_do_route: uri 15555550004 was rewritten to sip:15555550004@10.10.10.1, carrier 1, domain 1 Carrier found!! ("1", "131.226.0.144", "15555550004", "15555550004", "COMMX-LD"); Now please note that if I remove route id 31 then the call log looks like it is below: INFO:carrierroute:cr_do_route: uri 5515555550004 was rewritten to sip:15555550004@10.10.10.1, carrier 1, domain 1 INFO:carrierroute:rewrite_uri_recursor: URI or prefix tree nodes empty, empty rule list INFO:carrierroute:cr_do_route: rewrite_uri_recursor doesn't complete, uri 15555550004, carrier 1, domain 1 Please help!!! Carlos It will also help if there was complete routing example using carrierroute v 1.5.0, since the one in the wiki is outdated. Thanks.

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Re: [Kamailio-Users] bidirectionnal chat and presence sharing between a SIP and a jabber user
by Daniel-Constantin Mierla 20 Apr '09

20 Apr '09

Please cc all the time to mailing lists. Emailing privately will make the message dropped. Also, mailing list has the benefit of providing access to other people that might know better the answer. Thanks, Daniel On 04/20/2009 01:18 PM, rachid wrote: > *Hello, > I use the purplemap table to mapping sip uri and jabber client,so i > want to have a bidirectionnal chat and sharing between a dip and > jabber users > > I tried with purple modules* > > 2009/4/20 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 04/17/2009 11:22 AM, rachid wrote: > > Hello all, > > someone can help me to configure kamailio to i was able to have a > bidirectionnal chat and presence sharing between a SIP and a > jabber user ? > thank you > > what is the problem? Did you get or not any communication between > sip and xmpp? Have you tried with xmpp or purple modules? > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > > > > -- > ------------------------------------------------------------ > BAHA Rachid > Elève ingénieur en 3ème année à l'INPT, Rabat > GSM:(+33) 0619787609 > > -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Kamailio-Users] Question about route[4] for NAT
by x-kamailio＠sidell.org 20 Apr '09

20 Apr '09

Sorry for the potentially stupid question... I'm trying to make sense of the standard kamailio.cfg file. At the top, it says to remove the #n# comments to enable nat processing. Looking down in the code, I see that one of the sections to be enabled is route[4]. But, no where else in the file do I see that route(4) is ever called. Is the cfg file missing something, or is route(4) called in some way that is not apparent from reading the file? Thanks! -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

2 1

[Kamailio-Users] NAT & Branching in Parallel Forking Scenarios
by Brandon Armstead 20 Apr '09

20 Apr '09

Hello Guys, Going to say thanks for any input ahead of time. My question is currently we may have a scenario where we have a client using several UAC's registered to the same username. Now what I've seen is that sometimes one endpoint of User A may be NATed, another may be public. This seems to be causing some call flow control issues in a sense for when we handle for NAT / vs NOT NAT, at the time of a request and all of this is in parallel being handled. Example Scenario. User A calls User B -- (both internal). User B has end points X (NAT), Y(PUBLIC), and Z(NAT). So a call is being branched to X, Y, and Z in parallel. Now we fix_nated_contact for endpoint X, next we go to Y, however we've already fixed_nated_contact, thus this call is deemed bad, and then Z continues. Any thoughts / examples / suggestions for these types of scenarios? This is just an example one, one particular case where issues we have seen is more like follows: PSTN Calls User B (external to internal) User B has the same previous end points (X,Y,Z) + (PSTN End Points that we are adding in parallel to fork to). Same type issue, same type scenario. The end result of the issues are lack of audio, mis routed contacts, etc. So any help and input is appreciated, thanks!

3 2

[Kamailio-Users] run openser and asterisks in one box
by Metshet Seleshi 20 Apr '09

20 Apr '09

hi all, is it possible to run openser and asterisks in one box, I have googled but all the result I got have rewritehostport("X.X.X.X:5060) with a diffrent IP than the server IP, I am assuming either they have 2 network card or refering to another host . but will it be possible to put them togather? if so how, thanks, please help

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[Kamailio-Users] Profile takes some time to be cleared
by mayamatakeshi 20 Apr '09

20 Apr '09

Hello, I'm using the set_dlg_profile function from the dialog module (kamailio 1.4). I use it to control limit of simultaneous calls to clients. I'm noticing that after this function is called, even if the call terminates immediately, it takes some 3 to 4 seconds for the profile to be cleared. Is there any reason for this? I can see even if I call unset_dlg_profile on failure_route, the profile will take that time span to be cleared. regards, mayama

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[Kamailio-Users] shift/reduce conflict
by priz＠libero.it 20 Apr '09

20 Apr '09

Hi, I'm a newbie of kamailio. I'm an Asterisk expert, and I want to explore the "*SER" world, because I need a real SIP proxy. I planned kamailio project to begin. I want the radius auth. I have a question about a compile message. If I try to compile kamailio- 1.5.0-tls, I always receive the message "cfg.y: conflicts: 1 shift/reduce" Before build: # make distclean # make mantainer- clean and then: # make all bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex The rest of build is OK, without errors. Is this a problem or is normal? And if is not normal, how to avoid this problem? I have a Debian Lenny # uname -a Linux voicelab 2.6.26-1-686 #1 SMP Sat Jan 10 18:29: 31 UTC 2009 i686 GNU/Linux Uncomment: Makefile: TLS=1 FREERADIUS=1 Makefile. vars MODS_MYSQL=on and MODS_RADIUS=on /usr/local/src/kamailio/kamailio-1.5.0- tls/modules/acc/Makefile ENABLE_RADIUS_ACC=true Many thanks Priz

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[Kamailio-Users] opening 2 and more channels on 1 SIP account
by Tamer Higazi 18 Apr '09

18 Apr '09

Hi! I am new to opensips, and I am looking for a powerfull sip gateway sollution, because asterisk isn't able to handle such thing. My case: I have a Grandstream VoIP Device, at which a DECT base with 2 cordless phones are connected at 1 FX1 Port. If a call is placed and made through one cordless phone the other cordless phone appears as busy. What I want: I want to place and receive as many calls at the same time through 1 1 FXS Port with one single SIP Account, and through this where the base station is connected through. Question: Does kamilio have the ability for such things?! For any help and advise I would thank you kindly Tamer

2 1

[Kamailio-Users] More Complex Example Config File
by Alberto Listas 17 Apr '09

17 Apr '09

Hi, I am having a difficult time setting up a config file that works properly for a simple dispatcher with failover, database authentication and persistence. Could someone send me a more extensive example than the one that comes by default? Thanks, Alberto Furtado

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[Kamailio-Users] Lumps again
by Alex Hermann 17 Apr '09

17 Apr '09

Hello all, i got bitten by the lumps mess again. I can understand the lump concept makes a bit of sense if requests are just passing though a proxy, but nowadays Kamailio obtains more and more UA functionality where it is completely unappropriate. How do I store a modified PUBLISH body? I change it with subst_body() before calling handle_publish() but ofcourse the original body is stored by the presence UA. Is there currently a method to store a modified body? (apart from looping the message back to myself) Would implementing an apply_lumps() function or something like that be very difficult? -- Greetings, Alex Hermann

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[Serusers] switch
by hanane zahidi 17 Apr '09

17 Apr '09

Hi All; I need to use switch in my ser.cfg file .But ser give me some errors .so i need the correct syntax to do it here is my ser.cfg if ( method=="REGISTER"){ i = %fu % 2; switch(i){ case"1": route(1); break; case"0": route(2); break; } } thank tou for your help best regards

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[Kamailio-Users] get sdp audio port in a script variable
by antonroman 17 Apr '09

17 Apr '09

Hi all, I need to get the INVITE sdp audio port in a script variable to be used in an external command. Is this possible without modifying the source code? I think it can't be done with the TEXTOP module, can be? Thanks in advance Regards -- View this message in context: http://www.nabble.com/get-sdp-audio-port-in-a-script-variable-tp23061909p23… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] SER authentication with LDAP
by Leon Li 17 Apr '09

17 Apr '09

Hi all, I just would like to know if we can link SER to LDAP server (Microsoft AD) for user authentication, rather than using MySQL. If yes, how can we achieve it? Thanks Leon

1 0

[Kamailio-Users] prefix() cannot take pseudovariable argument?
by Alex Balashov 16 Apr '09

16 Apr '09

Why can't prefix() accept a pseudovariable argument, e.g. prefix("$avp(S:the_prefix)"); ? It is interpreted literally. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Serusers] Looking for commercial support
by Jean-Paul Smets 16 Apr '09

16 Apr '09

Hi, We are looking for commercial support for SER and SEMS. More precisely, we are looking for an engineer or a company who could help us to improve our existing SER based infrastructure. We would also be happy to get support from someone who has good experience with SER, SEMS and python. We do not need much in terms of features (registrar + proxy + voice2mail), but we want to be able to solve any little issue related to specific softphone or hardware, and to learn how to write new scripts for SER and SEMS. Do you have any advice? Regards, JPS. -- Jean-Paul Smets-Solanes, Nexedi CEO - Tel. +33(0)6 29 02 44 25 ERP5 Enterprise: Free / Open Source ERP for Critical Applications http://www.erp5.com ERP5 Express: Hosted Open Source ERP for small companies http://www.myerp5.com Nexedi: Consulting and Development of Free / Open Source Software http://www.nexedi.com

1 0

[Kamailio-Users] reducing T1 timer per route
by Juan Perez 16 Apr '09

16 Apr '09

hello I am using lcr and have 3 gws that I send the calls to, so far I am handling error responses using "on_failure_route" and calling the next_gw function and it is working fine. But in cases when 1 of those gws is disconnnected or super busy and a 100 Trying is not received in the first 3 - 5 seconds I would like to drop that call and use the next_gw. Right now I have to wait until all retries fail to try the next gw, but that happens after 64*T1 = 32 seconds. How can I accomplish that? cheers jp

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[Kamailio-Users] starting kamailio using kamailio binary
by Juan Perez 16 Apr '09

16 Apr '09

hello when using kamctl to start/stop kamailio it also provides a lot more "functions" like "kamctl lcr show", "kamctl ul show", etc. If I start kamailio using kamailio -f "path to my config" etc how can I have the same functionalities? regards jp

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mailing list description
by Daniel-Constantin Mierla 16 Apr '09

16 Apr '09

SIP Router Project - Users mailing list

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[Serusers] new installation
by Leon Li 16 Apr '09

16 Apr '09

Hi, I need to install SER for my company, which is pretty new for me. If we only need it for SIP registrar and proxy, do I need install any dependencies? And is there a list to explain what each dependency does? If I install the basic without dependencies, can I add them later and how? Thanks Leon

2 4

[Serusers] (no subject)
by Cristina Martínez Jimenez 15 Apr '09

15 Apr '09

which are the system requirements for SER? thanks, Cristina Martínez _________________________________________________________________ ¿Quieres crear tus propios emoticonos gratis? Descubre cómo hacerlo en el Club Oficial de Messenger http://vivelive.com/ilovemessenger/

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[Serusers] Connect SER with a SOFTSWITCH
by hanane zahidi 15 Apr '09

15 Apr '09

Hi All ; i need to connect SER with a softswitch but i found nothing in google. please if someone has some interesting links. give them to me thank you best regards

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[Kamailio-Users] 513 Message too big
by Joao Gomes Pereira 15 Apr '09

15 Apr '09

Hello to all I have Asterisk forwarding calls through Kamailio. If calls are placed by an Asterisk extension (like X-Lite) everything works fine. ( ex: X-Lite -> Asterisk -> Kamailio ) If calls are placed through a Cisco Router (connected to Asterisk) Kamailio answers: "513 Message too big" ( ex: Cisco VoIP client -> Cisco Router -> Asterisk -> Kamailio ) Also, the destination phone rings, but the calling party is disconnected because of this 513 message. This message if probably being originated here: if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; What could be the problem? Is the problem in Kamailio configuration or Asterisk configuration? And why does the call works when placed by an Asterisk extension, but doesn't work when placed by an Asterisk connected equipment? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

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Re: [Kamailio-Users] ERROR:core:receive_msg: no via found in reply
by Ovidiu Sas 15 Apr '09

15 Apr '09

Please open a bug with respect to this issue. This error should end up in the error_route. Try to run tcpdump with buffer rotation and see if you can spot the bogus reply. Regards, Ovidiu Sas On Mon, Apr 13, 2009 at 4:30 PM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: > OK, the error has occurred again today, and the error_route didn't > catch the condition. Any other ideas? > > thanks, > Geoff > > On Sun, Apr 12, 2009 at 6:42 PM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: >> Ovidiu, >> Thanks for your continued input. I currently have the error_route >> block configured with some logging. I am hoping this will be >> sufficient to capture the condition. The error hasn't happened today, >> traffic has been very light as it is Eastern and Passover, and I >> suppose people have better things to do than be on the phone :) >> >> If I get the log message and nothing from error_route, I will >> definitely do a tcpdump of the SIP port and try to correlate the >> timing. >> >> thanks! >> Geoff >> >> On Sun, Apr 12, 2009 at 8:46 AM, Ovidiu Sas <osas(a)voipembedded.com> wrote: >>> Hello Geoffrey, >>> >>> With tcpdump, you can capture data in buffers and rotate the buffers >>> and keep only the latest 2-3 buffers (check the man page). >>> As soon as the error arise, you stop the traffic and your bogus >>> message should be in your last buffer. >>> Just an idea ... >>> >>> Also, if your cps is not very high, you can print the callid of each >>> request/reply, and when the error arise, you can correlate it with the >>> callid. >>> >>> >>> Regards, >>> Ovidiu Sas >>> >>> On Sun, Apr 12, 2009 at 8:06 AM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: >>>> Ovidiu, >>>> Thank you for your pointer. I could capture all the SIP traffic, but >>>> it would be a LOT of data to sift through. This machine processes >>>> hundreds of thousands of calls per day. I have tried the error_route >>>> handler to see if it shows up there... fingers crossed. I was unaware >>>> of that handler before now. >>>> >>>> Thanks for the info! >>>> Geoff >>>> >>>> On Sat, Apr 11, 2009 at 8:38 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: >>>>> How often do you get this error? >>>>> IIRC, last time when I got this type of error it was related to a >>>>> missing Via header into a SIP reply. >>>>> Can you capture/sniff the traffic and check if all the received >>>>> replies are sane (via header is present)? >>>>> >>>>> Another idea would be to capture the bogus message into the error_route: >>>>> http://www.kamailio.net/dokuwiki/doku.php/core-cookbook:1.5.x#error_route >>>>> and then print the message via $mb: >>>>> http://www.kamailio.net/dokuwiki/doku.php/pseudovariables:1.5.x#sip_message… >>>>> >>>>> I haven't checked the code to see if this type of error will end up in >>>>> the error_route, but is it worth a try. >>>>> If it's not caught, please open a bug, because it should be. >>>>> >>>>> >>>>> Regards, >>>>> Ovidiu Sas >>>>> >>>>> On Sat, Apr 11, 2009 at 5:58 PM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: >>>>>> Tried adding that log into my onreply_route[1] block, but it didn't >>>>>> work. I can't seem to figure out I can possibly capture the SIP >>>>>> message which is generating this error. Any help would be much >>>>>> appreciated. >>>>>> >>>>>> On Fri, Apr 10, 2009 at 10:23 PM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: >>>>>>> Perhaps I should clarify what I am looking for a bit. I am guessing >>>>>>> based on the log "no via found in reply" that I would need to check >>>>>>> for this condition in my onreply_route[1] block. >>>>>>> >>>>>>> Based on my knowledge of SIP, all reply responses are REQUIRED to have >>>>>>> a VIA header to ensure the response traverses the same path as the >>>>>>> request... correct? >>>>>>> >>>>>>> Would the following effectively deal with this condition, or would >>>>>>> sending a 500 error in response to a reply cause problems down stream? >>>>>>> >>>>>>> onreply_route[1]{ >>>>>>> if(!is_present_hf("Via")){ >>>>>>> sl_reply_error("500","No Via found in response."); >>>>>>> } >>>>>>> } >>>>>>> >>>>>>> Thanks for anyone's assistance. >>>>>>> Geoff >>>>>>> >>>>>>> On Fri, Apr 10, 2009 at 12:02 PM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: >>>>>>>> Hello, >>>>>>>> I am seeing the following message repeatedly in my log file. I am >>>>>>>> guessing I have a server or servers misbehaving. Where might I be >>>>>>>> able to check for this error condition and log appropriately? I have >>>>>>>> attached my current configuration file for reference. >>>>>>>> >>>>>>>> Thanks! >>>>>>>> Geoff >>>>>>>> >>>>>>>> >>>>>>>> Apr 10 11:45:40 atl-sipgateway2 /usr/local/sbin/kamailio[15487]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:45:42 atl-sipgateway2 /usr/local/sbin/kamailio[15490]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:45:46 atl-sipgateway2 /usr/local/sbin/kamailio[15490]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:45:50 atl-sipgateway2 /usr/local/sbin/kamailio[15507]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:45:54 atl-sipgateway2 /usr/local/sbin/kamailio[15497]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:45:58 atl-sipgateway2 /usr/local/sbin/kamailio[15491]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:46:02 atl-sipgateway2 /usr/local/sbin/kamailio[15490]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:46:06 atl-sipgateway2 /usr/local/sbin/kamailio[15485]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:47:46 atl-sipgateway2 /usr/local/sbin/kamailio[15485]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:47:47 atl-sipgateway2 /usr/local/sbin/kamailio[15486]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:47:48 atl-sipgateway2 /usr/local/sbin/kamailio[15507]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:47:50 atl-sipgateway2 /usr/local/sbin/kamailio[15503]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:47:54 atl-sipgateway2 /usr/local/sbin/kamailio[15492]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:47:58 atl-sipgateway2 /usr/local/sbin/kamailio[15485]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:48:02 atl-sipgateway2 /usr/local/sbin/kamailio[15490]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:48:06 atl-sipgateway2 /usr/local/sbin/kamailio[15491]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> Apr 10 11:48:10 atl-sipgateway2 /usr/local/sbin/kamailio[15492]: >>>>>>>> ERROR:core:receive_msg: no via found in reply >>>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (OpenSER) - Users mailing list >>>>>> Users(a)lists.kamailio.org >>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>> >>>> >>> >> >

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[Kamailio-Users] Carrierroute and or LCR
by Stagg Shelton 14 Apr '09

14 Apr '09

I am preparing to add a new upstream SIP provider to my network, and now have a requirement to route based on the number that the user is calling. From what I have read it appears that LCR is the better method for doing this. I have one small issue in that I have a few customers who switch through my network in order to get to a specific carrier. Currently I am using carrierroute with all of my customers in order to support the ones that need to switch to specific carriers, and everyone else takes the default carrierroute. Can carrierroute and LCR be used together? If so can you give me a few pointers so that I can get a lab system working. If not can you suggest whether or not carrierroute could or should be used to route based on rate center. The rate center list from my new carrier is about 31,000 rows. If carrier route cannot or should not be used for this purpose, can you recommend a way that LCR can be used, and still have a few customers switch out to a specific carrier regardless of the number dialed? Thank You Stagg Shelton

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[Kamailio-Users] FW: LCR
by Carlos A. Alvarez 14 Apr '09

14 Apr '09

-----Original Message----- From: Carlos A. Alvarez Sent: Tuesday, April 14, 2009 11:32 AM To: 'tushar(a)eurus.net' Subject: RE: [Kamailio-Users] LCR I have a similar problem, with lcr and mysql modules. I am running lenny. The deb packages install fine. Bet when compiling the lcr and mysql are excluded. I do have installed all the dev libraries. Any ideas? -----Original Message----- From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of tushar(a)eurus.net Sent: Tuesday, April 14, 2009 10:06 AM To: juha heinanen; tushar(a)eurus.net; users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] LCR Thanks Jaha, but I dont think they have added LCR in excluded list, I myself complied the ver, I also tried to complied just the lcr module but no use. I wanted to know if anyone else is able to make it work? What make file you are using for Kamailio ? Tushar Original Message: ----------------- From: jh(a)tutpro.com (Juha Heinanen) Date: Tue, 14 Apr 2009 16:55:12 +0300 To: tushar(a)eurus.net, users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] LCR tushar(a)eurus.net writes: > Recently i installed Kamailio 1.5 everything working fine except the LCR > module, found theer was no lcr.so module file in the module file. if you compiled yourself, make sura that lcr module is not excluded. it uses pcre library and the may have been added to some exclude list. i'm not using standard kamailio makefile myself and don't know the details. -- juha -------------------------------------------------------------------- mail2web LIVE - Free email based on Microsoft(r) Exchange technology - http://link.mail2web.com/LIVE _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.287 / Virus Database: 270.11.54/2056 - Release Date: 04/14/09 06:17:00

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Re: [Kamailio-Users] LCR
by tushar＠eurus.net 14 Apr '09

14 Apr '09

Thanks Juha for instant replies, will update you after make giving a try on it. Original Message: ----------------- From: jh(a)tutpro.com (Juha Heinanen) Date: Tue, 14 Apr 2009 17:18:41 +0300 To: tushar(a)eurus.net, users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] LCR Juha Heinanen writes: > exclude_modules = list of module you use > export exclude_modules > make sorry, should have read exclude_modules = list of module you do NOT use -- juha -------------------------------------------------------------------- mail2web.com Enhanced email for the mobile individual based on Microsoft® Exchange - http://link.mail2web.com/Personal/EnhancedEmail

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Re: [Kamailio-Users] LCR
by tushar＠eurus.net 14 Apr '09

14 Apr '09

Thanks Jaha, but I dont think they have added LCR in excluded list, I myself complied the ver, I also tried to complied just the lcr module but no use. I wanted to know if anyone else is able to make it work? What make file you are using for Kamailio ? Tushar Original Message: ----------------- From: jh(a)tutpro.com (Juha Heinanen) Date: Tue, 14 Apr 2009 16:55:12 +0300 To: tushar(a)eurus.net, users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] LCR tushar(a)eurus.net writes: > Recently i installed Kamailio 1.5 everything working fine except the LCR > module, found theer was no lcr.so module file in the module file. if you compiled yourself, make sura that lcr module is not excluded. it uses pcre library and the may have been added to some exclude list. i'm not using standard kamailio makefile myself and don't know the details. -- juha -------------------------------------------------------------------- mail2web LIVE Free email based on Microsoft® Exchange technology - http://link.mail2web.com/LIVE

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Re: [Kamailio-Users] LCR
by tushar＠eurus.net 14 Apr '09

14 Apr '09

Thanks Jaha, but I dont think they have added LCR in excluded list, I myself complied the ver, I also tried to complied just the lcr module but no use. I wanted to know if anyone else is able to make it work? What make file you are using for Kamailio ? Tushar Original Message: ----------------- From: jh(a)tutpro.com (Juha Heinanen) Date: Tue, 14 Apr 2009 16:55:12 +0300 To: tushar(a)eurus.net, users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] LCR tushar(a)eurus.net writes: > Recently i installed Kamailio 1.5 everything working fine except the LCR > module, found theer was no lcr.so module file in the module file. if you compiled yourself, make sura that lcr module is not excluded. it uses pcre library and the may have been added to some exclude list. i'm not using standard kamailio makefile myself and don't know the details. -- juha -------------------------------------------------------------------- mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint

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Re: [Kamailio-Users] LCR
by tushar＠eurus.net 14 Apr '09

14 Apr '09

HI All, Recently i installed Kamailio 1.5 everything working fine except the LCR module, found theer was no lcr.so module file in the module file. I tried to rebuild the module separtly also but didnt work. Is there any changes in this module in ver 1.5? Did anyone also had same problem, I tried on debian 5 as well as on Centos 5 with same result. Though there is no issue in insatallation of ver 1.4 in either of the machines. Tushar Original Message: ----------------- From: Henning Westerholt henning.westerholt(a)1und1.de Date: Tue, 14 Apr 2009 10:59:03 +0200 To: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] carrier route vs LCR -------------------------------------------------------------------- mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint

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[Kamailio-Users] How to calculate hash_index on DB carrierroute
by Uriel Rozenbaum 14 Apr '09

14 Apr '09

Hi gentlemen, I'm configuring some balance and failure routes with CarrierRoute module. I have the following topology: -----------> Carrier1 --------> GW1 -----------> Carrier2 Kamailio -----------> Carrier1 --------> GW2 -----------> Carrier2 The idea will be to select a carrier using carrierroute going out through GW1. If GW1 fails (timeout) i want the call to go to the same carrier through the other GW. I assume the hash value calculated on cr_route will be the same for the failure route, but I need to verify the fields used on the DB to calculate a hash_index and the way that they are calculated so I can assure the previous behavior. Thanks in advance! Uriel

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[Kamailio-Users] reply cannot be parsed
by sasirekha jaganathan 14 Apr '09

14 Apr '09

Hi, When printing the message in function receive_msg() in core the buffer seems to be corrupt with CSeq header name. It shows as Seq and naturally the parsing fails throwing error as reply cannot be parsed and also no 2nd via found in reply. SIP/2.0 487 Request Terminated#015#012 Via: SIP/2.0/UDP xxxx;branch=z9hG4bK158a.61909fe2.0#015#012 From:xxxxx Call-ID: B110CD4E_CE5E_4F70_9472_439E0796F926#015#012 To: xxxxxx#015#012 Seq: 180 INVITE#015#012 Max-Forwards: 70#015#012 User-Agent: Kamailio (1.4.3-tls (i386/linux))#015#012 Content-Length: 0#015#012#015#012 Did anyone face this situation earlier? Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

2 1

[Kamailio-Users] ERROR:core:receive_msg: no via found in reply
by Geoffrey Mina 12 Apr '09

12 Apr '09

Hello, I am seeing the following message repeatedly in my log file. I am guessing I have a server or servers misbehaving. Where might I be able to check for this error condition and log appropriately? I have attached my current configuration file for reference. Thanks! Geoff Apr 10 11:45:40 atl-sipgateway2 /usr/local/sbin/kamailio[15487]: ERROR:core:receive_msg: no via found in reply Apr 10 11:45:42 atl-sipgateway2 /usr/local/sbin/kamailio[15490]: ERROR:core:receive_msg: no via found in reply Apr 10 11:45:46 atl-sipgateway2 /usr/local/sbin/kamailio[15490]: ERROR:core:receive_msg: no via found in reply Apr 10 11:45:50 atl-sipgateway2 /usr/local/sbin/kamailio[15507]: ERROR:core:receive_msg: no via found in reply Apr 10 11:45:54 atl-sipgateway2 /usr/local/sbin/kamailio[15497]: ERROR:core:receive_msg: no via found in reply Apr 10 11:45:58 atl-sipgateway2 /usr/local/sbin/kamailio[15491]: ERROR:core:receive_msg: no via found in reply Apr 10 11:46:02 atl-sipgateway2 /usr/local/sbin/kamailio[15490]: ERROR:core:receive_msg: no via found in reply Apr 10 11:46:06 atl-sipgateway2 /usr/local/sbin/kamailio[15485]: ERROR:core:receive_msg: no via found in reply Apr 10 11:47:46 atl-sipgateway2 /usr/local/sbin/kamailio[15485]: ERROR:core:receive_msg: no via found in reply Apr 10 11:47:47 atl-sipgateway2 /usr/local/sbin/kamailio[15486]: ERROR:core:receive_msg: no via found in reply Apr 10 11:47:48 atl-sipgateway2 /usr/local/sbin/kamailio[15507]: ERROR:core:receive_msg: no via found in reply Apr 10 11:47:50 atl-sipgateway2 /usr/local/sbin/kamailio[15503]: ERROR:core:receive_msg: no via found in reply Apr 10 11:47:54 atl-sipgateway2 /usr/local/sbin/kamailio[15492]: ERROR:core:receive_msg: no via found in reply Apr 10 11:47:58 atl-sipgateway2 /usr/local/sbin/kamailio[15485]: ERROR:core:receive_msg: no via found in reply Apr 10 11:48:02 atl-sipgateway2 /usr/local/sbin/kamailio[15490]: ERROR:core:receive_msg: no via found in reply Apr 10 11:48:06 atl-sipgateway2 /usr/local/sbin/kamailio[15491]: ERROR:core:receive_msg: no via found in reply Apr 10 11:48:10 atl-sipgateway2 /usr/local/sbin/kamailio[15492]: ERROR:core:receive_msg: no via found in reply

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[Kamailio-Users] correct usage of t_reply
by David Aldworth 11 Apr '09

11 Apr '09

I'm am attempting to return a 503 "Service Unavailable" after my kamailio server sends the provisional response to customer UA if we cannot find a route for the call. I am using: t_reply("503", "Service Unavailable"); However this results in: ERROR:tm:w_t_reply: failed to send a t_reply to a message for which no transaction-state has been established Reading up on t_reply, it clearly states: "Sends a stateful reply after a transaction has been established." So, I guess thats means even though we have received the invite, and send 100 Trying back, a transaction does not actually exist. So, my question is, how do I tell the sending UAC that we are 5XX so that it can stop sitting there waiting for session progress? Thanks for any help. David

3 3

Re: [Kamailio-Users] correct usage of t_reply
by mayamatakeshi 11 Apr '09

11 Apr '09

On Sat, Apr 11, 2009 at 1:19 PM, David Aldworth <daldworth(a)teliax.com> wrote: > Is there a way to tell if a transaction already exists before executing > t_newtran(); ? I suppose you could use t_check_trans(). But I haven't tried it. > If the route query scripts executes resulted in a route, but then we get a > 5XX from upstream, a transactions then exists. When it tries to execute > t_newtran(); I get the following: > > ERROR:tm:t_newtran: transaction already in process 0x2873dd08 > > If the query did not get a route, and no transaction exists, then executing > t_newtran(); before the t_reply works perfectly. > > Using sl_send_reply works, but if the transaction does exist then it sends > *two* 5XX messages back to the UAC. This is actually not harmful as far as I > can tell. But it's not clean either. I think at any point in the cfg it should be clear to the reader/writer where a transaction exists or not depending of which functions were called. I mean, you should know it. If you must recourse to some function to tell if a transaction exists or not, then I think the cfg is not clear. regards, takeshi P.S.: please, always CC to the list. > On Apr 10, 2009, at 10:06 PM, mayamatakeshi wrote: > >> On Sat, Apr 11, 2009 at 12:53 PM, David Aldworth <daldworth(a)teliax.com> >> wrote: >>> >>> I'm am attempting to return a 503 "Service Unavailable" after my >>> kamailio server sends the provisional response to customer UA if we >>> cannot find a route for the call. I am using: >>> >>> t_reply("503", "Service Unavailable"); >>> >>> However this results in: >>> >>> ERROR:tm:w_t_reply: failed to send a t_reply to a message for which no >>> transaction-state has been established >>> >>> Reading up on t_reply, it clearly states: "Sends a stateful reply >>> after a transaction has been established." So, I guess thats means >>> even though we have received the invite, and send 100 Trying back, a >>> transaction does not actually exist. >>> >>> So, my question is, how do I tell the sending UAC that we are 5XX so >>> that it can stop sitting there waiting for session progress? >> >> Hello, >> you can send the reply statelessly: >> sl_send_reply("503", "Service Unavailable" ); >> >> or statefully: >> t_newtran(); >> t_reply("503", "Service Unavailable"); >> >> Before you can use t_reply, a transaction must be established. This >> can be done explicitly calling t_newtran() or it will be do internally >> by other functions like t_relay. >> >> regards, >> takeshi. > >

1 0

[Kamailio-Users] Serious regressions in 1.5.0 dialog module - or just me?
by Alex Balashov 09 Apr '09

09 Apr '09

Greetings, I recently deployed 1.5.0 and was thrilled to have the opportunity to use the new sqlops module for a purpose that required a lot of custom database logic, but unfortunately had to roll back to 1.4.4 because a lot of stuff seemed to be broken in the 'dialog' module. I refuse to believe that the 'dialog' module was released in such a broken state, so I am thinking I am problem is that I am neglecting something very important or that there are some omissions in the documentation. System is CentOS 64-bit on a dual-CPU quad-core setup with 8 GB of RAM. Two fundamental problems: 1) get_profile_size() on profiles with no "values" (i.e. inside the enumerated "profiles_no_value" modparam) crashes. This problem also exists on 1.4.4. I have not tried other 1.4.x releases. So, I have: modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 65536) modparam("dialog", "rr_param", "dlgcor") modparam("dialog", "dlg_flag", 2) modparam("dialog", "default_timeout", 28800) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 0) modparam("dialog", "profiles_no_value", "global") modparam("dialog", "profiles_with_value", "user") And then, in my INVITE handler for inbound routes: # Set dialog construction flag. setflag(2); # Add this dialog to the global call count profile. set_dlg_profile("global"); # Add this dialog to a per-customer dialog profile. set_dlg_profile("user", "$avp(S:user_id)"); # Put out some debugging values for global and customer profiles. get_profile_size("global", "$var(global_dialog_count)"); get_profile_size("user", "$avp(S:user_id)", "$var(customer_dialog_count)"); The first get_profile_size() statement crashes--the one that refers to the profile without values. According to the GDB backtrace, the crash occurs at modules/dialog/dialog.c:701: if(sp_dest->setf(msg, &sp_dest->pvp, (int)EQ_T, &val)<0) { LM_ERR("setting profile PV failed\n"); return -1; } The cause is that sp_dest is NULL here. I have determined that the value of 'result' in w_get_profile_size() as passed into this prototype: static int w_get_profile_size(struct sip_msg *msg, char *profile, char *value, char *result) is NULL. I would conclude from this - not being a developer - that the root cause is that the config file/route script lexer does not properly distinguish between the value-less and with-value argument combination of get_profile_size() that can occur. A workaround is possible by setting my 'global' profile to be "with values" as well and using a bogus value key like "1" that is always the same. However, I do not think this should be happening. 2) The second problem only occurs in 1.5.0 and not in 1.4. It is much more serious. It would appear that the dialogs are not actually being inserted into the profiles - either the global or the per-user ones. I can see from the debug logs that the dialogs are being tracked because I set the dialog construction flag (2). I have also tried using dlg_manage(). However, get_profile_size(...) always returns 0 no matter what I do. --- I find it hard to believe that the 'dialog' module in 1.5.0 was released with such regressions, and cannot find anything in the devel list archives or the tracker suggesting that this behaviour has been elsewhere. My only conclusion is that either I am doing something wrong or that there is a problem with my environment. Admittedly, I have not tried reproducing these issues in 32-bit library environment and perhaps should. Please help! -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

4 19

[Serusers] Modified Display Name
by inge 09 Apr '09

09 Apr '09

Hi All, I know how to use subst function to modify the display name. But I try to use it with an usr_table like by provisioning in it the username and the new display name. Nevertheless it seems that subst doesn't support avp in the substitution string resulting that it print out "s:displayname" instead of the value. Is there any workaround to do something like that ? Regards,

2 2

[Serusers] To uri
by hanane zahidi 09 Apr '09

09 Apr '09

hi all please can you tell me how can i refer to "TO URI" field in my ser.cfg in order to test the sip destination best regards

2 1

[Kamailio-Users] failure_route Pseudo Variable
by Geoffrey Mina 09 Apr '09

09 Apr '09

Hello, I am using dispatcher and LCR. In my failure route I want to log the IP address of the machine which responded with an error. What is the pseudo I would use for this? I tried $si, but that appears to always contain the source IP of the server which issued the first INVITE. thanks, Geoff

2 2

[Kamailio-Users] No PID file found
by Alex R.S.M 09 Apr '09

09 Apr '09

Hi, I've just installed openser 3.1. When I try to start the service I am getting the following error. Is this a bug or I am missing something in the setting: /etc/openser$ sudo openserctl restart INFO: Stopping OpenSER : ERROR: No PID file found (/var/run/openser.pid)! OpenSER probably not running INFO: check with 'ps axw | /bin/egrep openser' Thanks, A S _________________________________________________________________ Rediscover Hotmail®: Get quick friend updates right in your inbox. http://windowslive.com/RediscoverHotmail?ocid=TXT_TAGLM_WL_HM_Rediscover_Up…

4 3

[Kamailio-Users] svn trunk frozen for sr integration
by Daniel-Constantin Mierla 08 Apr '09

08 Apr '09

Hello, Wednesday, April 15, is scheduled to import kamailio modules in the git repository of sip-router.org. That will become the working place for new features as Kamailio developers agreed to switch to git. From now until migration is done, developers should hold any commit for new features or cosmetic changes to the SVN trunk. The sources will be imported with entire commit log history plus lot of new patches done during last weeks to make them compile with sip-router - status at: http://sip-router.org/wiki/devel/kamailio-integration#module_integration_st… Even for bug fixes (if it is the case considering Easter holidays) we should work on 1.5 branch after a bit more private testing before committing. If some developer has important patches to commit in very short time, please reply. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Serusers] SIP message generator
by Steven C. Blair 08 Apr '09

08 Apr '09

I need to place a load on our server by generating a large number of SUBSCRIBE messages with a specific Event header type. I've tried sipsak without success. We have PacketGen on a laptop but that is a confusing application. Does anyone have any suggestions for how to generate 50-100 SUBSCRIBE messages per second? Thanks,steve

5 9

[Kamailio-Users] NONCE_REUSED return code
by catalina oancea 08 Apr '09

08 Apr '09

Hello, Does anybody know in which situation the NONCE_REUSED return code for www_authenticate would appear? I understand the usage of the STALE_NONCE code, this is when the nonce expires and the servers sends a new nonce to the phone. But why is the NONCE_REUSED used and why does it occur sometimes? Should I reject or accept the registration when this code appears? typedef enum auth_result { AUTH_ERROR = -5, /* Error occurred, a reply has not been sent out */ NO_CREDENTIALS, /* Credentials missing */ STALE_NONCE, /* Stale nonce */ INVALID_PASSWORD, /* Invalid password */ USER_UNKNOWN, /* User non existant */ ERROR, /* Error occurred, a reply has been sent out -> */ /* return 0 to the openser core */ AUTHORIZED, /* Authorized. If returned by pre_auth, */ /* no digest authorization necessary */ DO_AUTHORIZATION, /* Can only be returned by pre_auth. */ NONCE_REUSED /* Returned if nonce is used more than once */ /* Means to continue doing authorization */ } auth_result_t; Many thanks, Catalina Oancea

2 5

[Serusers] Use of @IP source in ser.cfg
by inge 08 Apr '09

08 Apr '09

Hi all, I know that it's possible to use a variable in XLOG for print out the IP Source. But is it possible to work on @IP_SRC (use if, avp, etc.) in ser.cfg ? I can work with search() but it's not very flexible to have dynamic test with database for example. My SER version is 0.9.6. Regards, Adrien

1 1

[Kamailio-Users] Path and received parameter
by Klaus Darilion 07 Apr '09

07 Apr '09

Hi! During REGISTER processing I call add_path_received() in the outbound proxy. Now if the INVITE is routed from main proxy to the client (via the OBP), the OBP as to read the received parameter from the Ruote header and use this value as destination URI. Is there a function available to perfomr this action or do I have to do it manually (parsing Route header to the get parameter)? thanks klaus

3 4

[Kamailio-Users] LCR Error
by Geoffrey Mina 07 Apr '09

07 Apr '09

Hello, I just upgraded a server from 1.4 to 1.5. I am getting this error when loading LCR. Did something change, or am I losing my mind? n Apr 7 12:09:27 atl-sipgateway1 kamailio: ERROR:core:main: bad config file (2 errors) Apr 7 12:11:06 atl-sipgateway1 kamailio: ERROR:core:set_mod_param_regex: parameter <fr_inv_timer_avp> not found in module <lcr> Apr 7 12:11:06 atl-sipgateway1 kamailio: CRITICAL:core:yyerror: parse error in config file, line 74, column 20-21: Can't set module parameter Apr 7 12:11:06 atl-sipgateway1 kamailio: ERROR:core:set_mod_param_regex: parameter <fr_inv_timer> not found in module <lcr> Apr 7 12:11:06 atl-sipgateway1 kamailio: CRITICAL:core:yyerror: parse error in config file, line 78, column 18-19: Can't set module parameter Here is my config for a working server. ## Enable LCR ## loadmodule "lcr.so" modparam("lcr","db_url","mysql://blah:blah@localhost/kamailio") modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr","gw_uri_avp", "$avp(i:709)") modparam("lcr","ruri_user_avp", "$avp(i:500)") modparam("lcr","fr_inv_timer",30) modparam("lcr","flags_avp", "$avp(i:712)") modparam("lcr","lcr_hash_size", 1024) modparam("lcr","fetch_rows", 5000) modparam("lcr","ping_interval", 180) modparam("lcr","ping_from","sip:something.com") The docs no longer make any reference to fr_inv_timer or fr_inv_timer_avp.. but I have a working 1.5 server running with this config... but my newly installed one won't start. thanks, Geoff

4 4

[Serusers] (no subject)
by hanane zahidi 07 Apr '09

07 Apr '09

hi; how can i make a test on destination uri in a sip message in ser.cfg . i need the syntax best regards

1 0

[Kamailio-Users] Limit the timing of session
by ALAEDDINE abbech 07 Apr '09

07 Apr '09

hi all, How can i limit the timing of session? thanks

4 4

[Serusers] load balacing
by hanane zahidi 06 Apr '09

06 Apr '09

hi all please, can you tell me is this scenario correct i want to use ser as a load balancer for two asterisk servers. i have two users which are registred in both astersik servers with the same extensions (sip.conf and extensions.conf are the same for the asterisk servers) i have a user registred in the ser server. i imagine load balancer like: when the ser user calls an asterisk extension .the ser redirect this call to one of the asterisk.when the ser user recall the extension the ser will redirect the call to the other asterisk please tell me if i am wrong and if possible give me the correct scenario. thank you

2 3

[Kamailio-Users] Nethelper Causes Kamailio to Crash
by Carlos A. Alvarez 06 Apr '09

06 Apr '09

I am running into a situation where the nathelper module is causing openser/kamailio to crash. The crash occurs when opensbc sends and invite to openser with an extra CR in the message body. I already contacted the opensbc folks about this. I just wanted you all to be aware that this could cause serious service interruptions. Error log: kernel: [599650.771067] kamailio[25461]: segfault at 18 ip b7a0a6c7 sp bfbaaa70 error 4 in nathelper.so[b79f7000+18000] Sample invite sent to kamailio.. Please look at the extra CR/LF after Supported. The nathelper module crashes after this. INVITE sip:phonenumber@chgw.commx.net SIP/2.0 From: "South Test" <sip:phonenumber@orig.domain.com;user=phone>;tag=392385664-1238777567965- To: <sip:phonenumber@chgw.commx.net> Via: SIP/2.0/UDP opensbc:5060;branch=z9hG4bKdec1a187dd1ede1193b6b6f3163a7dda;uas-addr=openser;rport CSeq: 913493359 INVITE Call-ID: BW1252479650304091860611707(a)orig.domain.com-0x000a Contact: <sip:+18135146513@opensbc> Max-Forwards: 9 Accept: multipart/mixed, application/dtmf-relay, application/media_control+xml, application/sdp Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE Proxy-Require: privacy Supported: 100rel Content-Type: application/sdp Content-Length: 214 v=0 o=BroadWorks 3675 1 IN IP4 gw1.domain.com s=- c=IN IP4 opensbc t=0 0 m=audio 10074 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000

3 9

[Kamailio-Users] LCR Configuration Confirmation
by geoffreymina＠gmail.com 06 Apr '09

06 Apr '09

Hello, I am modifying my LCR setup and just wanted to get some confirmation that I am following best practices. I am a I am loading up 18,500 routes into the table which all belong to a single group. There will be a single route in a second group which all non-matches will fail over to. That being said, this is what my configuration looks like: modparam("lcr","lcr_hash_size", 20000) modparam("lcr","fetch_rows", 5000) This server has 8G of RAM and I am currently using 4M of private memory (#define PKG_MEM_POOL_SIZE 4*1024*1024 ). Thanks, Geoff

4 6

Re: [Kamailio-Users] another instance of kamailio on the same machine
by Juan Perez 03 Apr '09

03 Apr '09

thank you ovidiu I do not have an init scrip, I start and stop using the "kamctl stop|start" which is a bash script.Is this the one that you are referring to? I intend to listen on the same port but in another IP, each instance will use a different ip, so I have to change the listening ip only. regards jp ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Juan Perez <jperezsip2008(a)yahoo.com> Cc: Iñaki Baz Castillo <ibc(a)aliax.net>; users(a)lists.kamailio.org Sent: Friday, April 3, 2009 3:53:25 PM Subject: Re: [Kamailio-Users] another instance of kamailio on the same machine All you need is to follow this simple steps: 1. make a clean install of kamailio 2. define your config files for each service: - kamailio1.cfg - kamailio2.cfg 3. create two seprate init scripts: kamailio1 and kamailio2 4. make sure that each init script is using the appropriate config file (use option -f <my_config file>) 5. make sure that you are not listening on the same sockets for both servers (i.e. set one server on port 506 and the second one on port 5070) 6. If you are creating pid files, make sure that the pid files are different for each service That's it. You don't need any symbolic links. Regards, Ovidiu Sas On Fri, Apr 3, 2009 at 4:46 PM, Juan Perez <jperezsip2008(a)yahoo.com> wrote: > > understood. > > so I just need 2 softlinks? > > > > > ----- Original Message ---- > From: Iñaki Baz Castillo <ibc(a)aliax.net> > To: users(a)lists.kamailio.org > Sent: Friday, April 3, 2009 3:43:24 PM > Subject: Re: [Kamailio-Users] another instance of kamailio on the same machine > > El Viernes 03 Abril 2009, Juan Perez escribió: >> hi guys , I tried to do it and it wont start >> >> my setup is: >> centos5 >> kamailio 1.4 >> >> I installed kamailio originally in /usr/local/install-dir/ >> Now I created a folder named "install-dir2" and created these 3 soft links: >> >> /usr/local/install-dir2/etc2/kamailio2/kamailio2.cfg --> /usr/local/install-dir/etc/kamailio/kamailio.cfg > > Why do you do it? ^^^^^^^^ > Doing that kamailio2.cfg is a link to kamailio.cfg, so when you run: > > kamailio2 -f /usr/local/install-dir2/etc2/kamailio2/kamailio2.cfg > > You are loading /usr/local/install-dir/etc/kamailio/kamailio.cfg !!! > > This is, kamailio2 is trying to bind the same sockets so it cannot start. > > > -- > Iñaki Baz Castillo <ibc(a)aliax.net> > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >

3 3

[Kamailio-Users] another instance of kamailio on the same machine
by Juan Perez 03 Apr '09

03 Apr '09

hi To install another instance of kamailio listening in another IP, does one need to do something in particular aside from install it in another directory? Also, when it is time to run the script to create the mysql tables, how can I tell the script to create a db named "openser2" to avoid deleting the existing one named by default "openser" ? Just by changing the line DBNAME=openser in kamctlrc to DBNAME=openser2? thank you jp

8 22

[Serusers] Fw: running rtpproxy on quad core processors
by Mundiline 03 Apr '09

03 Apr '09

I still have not been able to figure this one out. Any help will appreciated... Samuel a Seruser sent this link: http://www.b2bua.org/wiki/RTPproxy/FAQ Is this correct then: does it mean i need to replace the entry in the opensips.cfg script:modparam ("nathelper", "rtpproxy_sock", "udp:localhost:2222") For: modparam ("nathelper", "rtpproxy_sock", "unix:@l_prefix@/var/opensips/opensips_rtpproxy.sock1") modparam ("nathelper", "rtpproxy_sock", "unix:@l_prefix@/var/opensips/opensips_rtpproxy.sock2") modparam ("nathelper", "rtpproxy_sock", "unix:@l_prefix@/var/opensips/opensips_rtpproxy.sock"2) modparam ("nathelper", "rtpproxy_sock", "unix:@l_prefix@/var/opensips/opensips_rtpproxy.sock3") ? and then run this: rtpproxy -l X.Y.Z.W -L 8192 -p /var/run/rtpproxy.pid -s /var/run/rtpproxy.sock -m 25000 -M 34999 rtpproxy -l X.Y.Z.W -L 8192 -p /var/run/rtpproxy1.pid -s /var/run/rtpproxy1.sock -m 35000 -M 44999 rtpproxy -l X.Y.Z.W -L 8192 -p /var/run/rtpproxy2.pid -s /var/run/rtpproxy2.sock -m 45000 -M 54999 rtpproxy -l X.Y.Z.W -L 8192 -p /var/run/rtpproxy3.pid -s /var/run/rtpproxy3.sock -m 55000 -M 64999 ? Thanks. ----- Forwarded Message ---- From: Mundiline <mundiline(a)yahoo.com> To: serusers(a)iptel.org Sent: Tuesday, March 31, 2009 3:44:10 PM Subject: running rtpproxy on quad core processors Hi, I would like to run rtpproxy 1.2.0 on a server with Quad-core dual Xeon processor and load balance the load across the 4 processors. Do you know how do I enable this feature pls. ( I am not an expert in this field so keep it simple pls. thanks) Thanks in advance Regards, Tony

2 1

[Serusers] load balancing configuration file
by hanane zahidi 03 Apr '09

03 Apr '09

hi all;I am a new member in this mailing list and i wish you can help me. i have to do load balancing for two asterisk servers using the SER server.but i'm really lost. if you could help me by showing me an ser.cfg sample to do it best regards.

3 6

[Kamailio-Users] BIGINT MySQL problems in 1.5.0.
by Alex Balashov 03 Apr '09

03 Apr '09

Greetings, I just installed 1.5.0 and wired it up to a MySQL 5.0.x database, and it appears that I am getting empty/null values for BIGINT columns when accessing them from script. This is true when I do something like this with SQLOps, for example: sql_query("test", "SELECT someBigIntColumn FROM tbl", "ra"); $var(i) = $dbr(ra=>[0,0]); $var(i) will have an empty value. Same will be the case if the access subscript is not [0,0] but say, [0,1]. The problem does not seem to be specific to sqlops; the same occurs with AVPs: avp_db_query("SELECT someBigIntColumn FROM tbl", "$avp(S:bigInt)"); This is also empty. I did a debug and confirmed that the right value for someBigIntColumn is coming out of the database, and has the right data type. All that works fine. However, for some reason this value appears to be invisible within the evaluative scope of the route script. For now, I am working around it by doing CONVERT(someBigIntColumn, CHAR(20)) inside the query. However, in principle this means that BIGINT handling is broken. Has anyone else run into this? Any insights, or especially, solutions? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

3 5

[Serusers] 305 Use Proxy
by Rick van Rein 03 Apr '09

03 Apr '09

Hello, I am trying to setup a redirecting proxy server with Ser, but I find that my GXP-2020 phone gets into an endless loop. Is this common? The phone is configured to use my ser service which is set as a proxy. If it tries to dial things that this proxy shouldn't handle, it sends back a "305 Use Proxy" response. The phone seems to ignore the proxy and tries again. 1) Is it common for the proxy redirection not to work? I've seen it being ill-advised on iptel.org because the RFC isn't too clear how it works. 2) What would be the proper formula to get Ser to send back a proxy redirect? I may be approaching it wrongly, but I cannot find much documentation about it online. Thanks, Rick van Rein OpenFortress

2 3

Re: [Kamailio-Users] Get transaction ID
by Daniel-Constantin Mierla 03 Apr '09

03 Apr '09

please cc to mailing lists ... private emails of such type get ignored. Thanks, Daniel On 04/02/2009 05:44 PM, kévin l'heveder wrote: > Thanks you to help me, > > I use the 'send' function to send the whole message from kamailio.cfg > to my management script. So i can parse the message, get the > informations i need. > > I've seen the function t_write and i understand how it works but in > the given example: > modparam("tm","tw_append","append1:Email=avp[i:12];UA=*hdr[User-Agent]*") > modparam("tm","tw_append","append2:body=*msg[body]*") > > > where could i find all the variable that i can use ?? because in core-cookbook doc (http://kamailio.org/dokuwiki/doku.php/core-cookbook:devel) > > i've never seen the variable msg[body] or hdr[User-Agent] or variable bind to transaction. > > Regards, > Koon > > > 2009/4/1 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 04/01/2009 05:08 PM, kévin l'heveder wrote: > > Hi all! > > I'd like to use the *t_reply* *MI* *function*, but one of the > parameter is *'trans_id'* and i don't know how to get it? > > from where do you get the details of the request to reply to? > > TM exports functions t_write_* that will give the id: > http://kamailio.org/docs/modules/1.5.x/tm.html#id2531309 > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > -- Daniel-Constantin Mierla http://www.asipto.com/

2 2

[Kamailio-Users] MI_XMLRPC
by Jason Penton 02 Apr '09

02 Apr '09

Hi *, Has anyone else noted poor stability with the mi_xmlrpc MI interface? We are noticing that after a while (very little load) it stop responding. Kind regards Jason

3 7

[Kamailio-Users] (no subject)
by kévin l'heveder 02 Apr '09

02 Apr '09

Thanks you to help me, I use the 'send' function to send the whole message from kamailio.cfg to my management script. So i can parse the message, get the informations i need. I've seen the function t_write and i understand how it works but in the given example: modparam("tm","tw_append","append1:Email=avp[i:12];UA=*hdr[User-Agent]*") modparam("tm","tw_append","append2:body=*msg[body]*") where could i find all the variable that i can use ?? because in core-cookbook doc (http://kamailio.org/dokuwiki/doku.php/core-cookbook:devel) i've never seen the variable msg[body] or hdr[User-Agent] or variable bind to transaction. Regards, Koon 2009/4/1 Daniel-Constantin Mierla <miconda(a)gmail.com> - Masquer le texte des messages précédents - Hello, On 04/01/2009 05:08 PM, kévin l'heveder wrote: > Hi all! > > I'd like to use the *t_reply* *MI* *function*, but one of the parameter is > *'trans_id'* and i don't know how to get it? > > from where do you get the details of the request to reply to? TM exports functions t_write_* that will give the id: http://kamailio.org/docs/modules/1.5.x/tm.html#id2531309 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

[Serusers] Change loglevel in runtime
by Jose Silva 02 Apr '09

02 Apr '09

Hi, Does anybody know if it is possible to change SERs log level in run-time? Thanks in advance, José silva

2 1

Re: [Kamailio-Users] TextOps - Removal of To
by jh＠tutpro.com 02 Apr '09

02 Apr '09

sasirekha jaganathan writes: > Even in the Wireshark the To Header goes with the tag-param. then you did something wrong. remove to header with remove_hf function and add a new one with insert_hf function and include the header body the way you want it to be. that must work. -- juha

1 0

[Kamailio-Users] TextOps - Removal of To
by sasirekha jaganathan 02 Apr '09

02 Apr '09

Hi, I need only the To value omitting the tag parameter in the To header. $avp(s:toadd)=$tu; if(remove_hf("To")){ xlog("Removed To tag \n"); } insert_hf("To: $avp(s:toadd)\r\n"); The log "Removed To tag" gets printed but after inserting if I try to print the header value with $hdr(To) it shows the value with tag-param too. Is there anything wrong? Thanks Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

3 2

[Kamailio-Users] Kamailio and Jabberd in same server
by Joao Gomes Pereira 01 Apr '09

01 Apr '09

Hello Is it possible to use JabberD and Kamailio (with XMPP Component Mode) in the same server? Thanks Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

2 3

[Kamailio-Users] Get transaction ID
by kévin l'heveder 01 Apr '09

01 Apr '09

Hi all! I'd like to use the *t_reply* *MI* *function*, but one of the parameter is * 'trans_id'* and i don't know how to get it? thanks. Koon

2 1

[Kamailio-Users] removal of tag-param in To Header
by sasirekha jaganathan 01 Apr '09

01 Apr '09

Hi , Can anyone point me to the function used to remove a tag in To Header field. For ex: To:sss@sss.com;tag=09888 In this case how to remove the tag param. Thanks Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

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Re: [Kamailio-Users] another instance of kamailio on the same machine >
by vallimamod abdullah 01 Apr '09

01 Apr '09

Hello Juan, On Wed, 1 Apr 2009 06:35:35 -0700 (PDT) Juan Perez <jperezsip2008(a)yahoo.com> wrote: > > we will use a different machine but we are limited with hardware now so in the mean time [and not for production traffic] we need to use the same hardware for > more than one instance to test some stuff > thank you guys If you look at a clean way to have multiple kamailio instances in one physical server, you should look at linux-vserver which work great under debian (http://wiki.debian.org/LinuxVserver) This will allow you to have separate environment for each instance with different IPs and no overhead. We use it for a long time now without problem. Don't hesitate to contact me if you need more informations. Regards, -vma .

1 0

[Kamailio-Users] remove_uri_param(param)?
by jh＠tutpro.com 01 Apr '09

01 Apr '09

siputils has function uri_param that can us used to check if r-uri has a given parameter. how about a function remove_uri_param(param) that would remove from r-uri the given parameter? does any efficient (non regex) means exist to achieve it? -- juha

2 1

[Kamailio-Users] Processing ACK.
by Carlos A. Alvarez 01 Apr '09

01 Apr '09

I am having an issue while setting up Kamailio. I using the carrierroute modules to process calls between carriers. The configuration works great. However I have and issue. On the SIP messages We are forcing all the media through an RTP proxy, however my issue has to do with the Contact field on the Message Header. While I am able to use force_rtp_proxy to replace the contact and connection fields on the SDP part of the message, I can seem to modify the Contact on the Header part. I tried using subst to change the Contact:, and while I was partly successful, it seems the ACKs going from one carrier to another are getting messed up and throwing Kamailio into a loop. We are trying to avoid providing Contact: information to our carriers. Basically keeping them from know about each other. Any ideas/help will be Greatly appreciated.

2 1

[Serusers] running rtpproxy on quad core processors
by Mundiline 01 Apr '09

01 Apr '09

Hi, I would like to run rtpproxy 1.2.0 on a server with Quad-core dual Xeon processor and load balance the load across the 4 processors. Do you know how do I enable this feature pls. ( I am not an expert in this field so keep it simple pls. thanks) Thanks in advance Regards, Tony

2 1

[Serusers] Running multiple instances of rtpproxy? How to set up in SER?
by Frank Durda IV 01 Apr '09

01 Apr '09

I have reached the point where a single instance of rtpproxy is about maxed-out, showing that its single process is comsuming 85% or so and climbing of the CPU available. That works out to about 600 standing calls, the bulk of which are G.711. This is a multi-processor/core system and I have several other CPUs not really doing anything, so I would like to start up multiple instances of rtpproxy, which rtpproxy says it can do, controlling them via AF_INET sockets instead of the default AF_UNIX. How you set up rtpproxy for multiple instances seems reasonably straightforward, but what you do on the SER side to let it know that these additional copies of rtpproxy exist and how you or SER divides call load between them is less clear. Is there any documentation or examples showing this done, or better still, someone with a working ser.ctl file who would be willing to show at least that part of their configuration? Ideally, I'm looking for a case where calls are divided among rtpproxies regardless of where they came from or where they are going, as I have one particular call source/destination pair that is absurdly large compared to all others, and that so it in particular has to be distributed across across multiple rtpproxies. Thanks in advance!

3 6

[Kamailio-Users] Force core dump
by catalina oancea 01 Apr '09

01 Apr '09

Hello Can anybody please help me on how to force a core dump if a child crashes? I see this: /usr/sbin/kamailio[12328]: INFO:core:handle_sigs: child process 12379 exited by a signal 11 /usr/sbin/kamailio[12328]: INFO:core:handle_sigs: core was not generated /usr/sbin/kamailio[12328]: INFO:core:handle_sigs: terminating due to SIGCHLD /usr/sbin/kamailio[12393]: INFO:core:sig_usr: signal 15 received And if I run with gdb, it says 'Program exited normally' Has anybody managed to make a core dump in this case? Thanks Catalina

3 5

[Kamailio-Users] remove codec from SDP
by Klaus Darilion 31 Mar '09

31 Mar '09

Hi! Are there any features in K to remove a codec from the SDP? (execept manual removing with textops)? thanks klaus

3 4

[Kamailio-Users] sanity module trouble
by kokoska rokoska 31 Mar '09

31 Mar '09

Hi all, due to malformed INVITEs I'm receiving from clients I like to use sanity module. But after few test I'm affraid I don't understand correctly how it works :-) 1. I try to load the module: loadmodule "sanity.so" modparam("sanity", "default_checks", "999") and kamailio rejects config with the: ERROR:core:set_mod_param_regex: parameter <default_checks> not found in module <sanity> 2. When I comment out "modparam", kamailio starts successfuly, but: All INVITEs (from all clients) are reject through this snippet of code if (method=="INVITE") { if (sanity_check("128")) { xlog("L_ERR", "bad INVITE SDP body length from $si:$sp\n"); exit; } else if (sanity_check("1024", "7")) { xlog("L_ERR", "malformed INVITE from $si:$sp\n"); exit; } } I don't think that all of them could have bad length on SDP body (about 120 diffenet UAC types), so I assume I use "sanity" wrong way... Could you, please, shed some light to proper usage of sanity module? Thank you very much! Best regards, kokoska.rokoska

2 6

[Serusers] Network Monitoring
by Bill McNamara 31 Mar '09

31 Mar '09

We are running SER 2.0.0 under RedHat 4.5 (kernel 2.6.9). Anyone have any recommendations for network traffic analyzers, hardware or software, so we can monitor SIP traffic? We would like the solution to be able to detect dropped calls, generate statistics and possibly debug active connections. A Google search returns many hits, but I'd like to get some recommendations from folks actually running the monitors before I start looking at trial versions. Thanks, Bill

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Re: [Serusers] Network Monitoring
by Ilya Pichugin 31 Mar '09

31 Mar '09

Hi! Yes, CDRTool is what you need check this URL http://cdrtool.ag-projects.com/ you wrote 31 march 2009 г., 18:11:35: > >We are running SER 2.0.0 under RedHat 4.5 (kernel 2.6.9). >Anyone have any recommendations for network traffic analyzers, hardware or software, so we can monitor SIP traffic? >We would like the solution to be able to detect dropped calls, generate statistics and possibly debug active connections. >A Google search returns many hits, but I’d like to get some recommendations from folks actually running the monitors before I start looking at trial versions. >Thanks, >Bill -- Best Regards, i.pichugin(a)ptl.ru

1 0

Re: [Kamailio-Users] Force core dump
by Daniel-Constantin Mierla 31 Mar '09

31 Mar '09

do not forget to cc the mailing list ... On 03/31/2009 02:44 PM, catalina oancea wrote: > Hello > I did that. I have: > > ulimit -a > core file size (blocks, -c) unlimited > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) 38400 > max locked memory (kbytes, -l) 32 > max memory size (kbytes, -m) unlimited > open files (-n) 1024 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) 819200 > real-time priority (-r) 0 > stack size (kbytes, -s) 10240 > cpu time (seconds, -t) unlimited > max user processes (-u) 38400 > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > [root@voipnow2 ~]# ulimit -a > core file size (blocks, -c) unlimited > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) 38400 > max locked memory (kbytes, -l) 32 > max memory size (kbytes, -m) unlimited > open files (-n) 1024 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) 819200 > real-time priority (-r) 0 > stack size (kbytes, -s) 10240 > cpu time (seconds, -t) unlimited > max user processes (-u) 38400 > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > > I still see 'core was not generated' in kamailio log. Should I look > for a core dump anyway? Where do I find it? > if the message is 'core was not generated' then no core file around. Do you get any other error message before signal 11? What is the kamailio version on operating system? Cheers, Daniel > > > > > > > 2009/3/31 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Hello, >> >> set: 'ulimit -c unlimited' >> >> Cheers, >> Daniel >> >> >> On 03/31/2009 02:08 PM, catalina oancea wrote: >> >>> Hello >>> >>> Can anybody please help me on how to force a core dump if a child >>> crashes? I see this: >>> >>> /usr/sbin/kamailio[12328]: INFO:core:handle_sigs: child process 12379 >>> exited by a signal 11 >>> /usr/sbin/kamailio[12328]: INFO:core:handle_sigs: core was not generated >>> /usr/sbin/kamailio[12328]: INFO:core:handle_sigs: terminating due to >>> SIGCHLD >>> /usr/sbin/kamailio[12393]: INFO:core:sig_usr: signal 15 received >>> >>> And if I run with gdb, it says 'Program exited normally' >>> >>> Has anybody managed to make a core dump in this case? >>> >>> Thanks >>> Catalina >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> >> -- >> Daniel-Constantin Mierla >> SIP Router Masterclass - Kamailio (OpenSER) Training >> http://www.asipto.com/index.php/sip-router-masterclass/ >> >> >> -- Daniel-Constantin Mierla SIP Router Masterclass - Kamailio (OpenSER) Training http://www.asipto.com/index.php/sip-router-masterclass/

2 1

[Kamailio-Users] Issue with Carrierroute in 1.4
by Uriel Rozenbaum 30 Mar '09

30 Mar '09

Guys, I'm getting this error on the logs for failure route but just for some of the calls I pass. If I call afterwards to that number it rings OK, so it is a random event: ERROR:carrierroute:cr_load_next_domain: during set_next_domain_recursor, prefix '1054111547533610', carrier 2, domain 1 I'm filling carrierfailureroute with an empty string when I want taht route to be the final one; is it OK to do that? Thanks in advance, Uriel

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Re: [Kamailio-Users] Call not Accounted
by Daniel-Constantin Mierla 30 Mar '09

30 Mar '09

Hello, please cc all the time to the list so people learn what was the solution or if was a mistakenly reported issue. Thanks, Daniel On 03/30/2009 02:36 PM, Alberto Listas wrote: > Hi, > > I found the problem the default kamailio.cfg in my version has a > "domain" instead of a "acc" in the middle of the acc section. > > Strange that this not generate any type of error that I could see. > > Cheers, > > Alberto > > ----- Original Message ----- From: "Daniel-Constantin Mierla" > <miconda(a)gmail.com> > To: "Alberto Listas" <listas(a)b2br.net> > Cc: <users(a)lists.kamailio.org> > Sent: Monday, March 30, 2009 6:15 AM > Subject: Re: [Kamailio-Users] Call not Accounted > > >> Hello, >> >> On 03/28/2009 02:48 PM, Alberto Listas wrote: >>> Hi, >>> >>> I am a new user and recently installed my first Kamailio. I was able >>> to turn MySQL on, create users and do my first calls (extension to >>> extension). >>> But I can't get Kamailio to write the call info to MySQL. >>> I attached Kamailio.cfg. >> have you created the extra columns in acc and missed_calls table? >> >> Any error in syslog? >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> SIP Router Masterclass - Kamailio (OpenSER) Training >> http://www.asipto.com/index.php/sip-router-masterclass/ >> > > > -------------------------------------------------------------------------------- > > > > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.11.31/2028 - Release Date: > 03/28/09 07:16:00 > ------------------------------------------------------------------------ > > > No virus found in this outgoing message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.11.31/2029 - Release Date: 03/29/09 16:56:00 > > -- Daniel-Constantin Mierla SIP Router Masterclass - Kamailio (OpenSER) Training http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] Problem login Siremis
by Alberto Furtado 30 Mar '09

30 Mar '09

Hi, I am a new user, just installed Kamailio 1.4.3. The Siremis interface will be of great help but I am getting this error when I try to login. [2009-02-13 18:22:53 (GMT)] An exception occurred while executing this script: Error message: #256, No Database information found in the config file. Script name and line number of error: /var/www/siremis-0.9.1/openbiz/bin/Configuration.php:42 I created the ob_users Table, inserted a user (admin,admin), but I can't seem to figure out what I did wrong. Maybe its not finding the Config.xml ??? Thanks in advance for any help, Alberto Furtado No virus found in this outgoing message. Checked by AVG - http://www.avg.com Version: 8.0.176 / Virus Database: 270.10.23/1951 - Release Date: 13/2/2009 06:51

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[Kamailio-Users] Call not Accounted
by Alberto Listas 30 Mar '09

30 Mar '09

Hi, I am a new user and recently installed my first Kamailio. I was able to turn MySQL on, create users and do my first calls (extension to extension). But I can't get Kamailio to write the call info to MySQL. I attached Kamailio.cfg. Thanks, Alberto Furtado No virus found in this outgoing message. Checked by AVG - www.avg.com Version: 8.0.238 / Virus Database: 270.11.31/2028 - Release Date: 03/28/09 07:16:00

2 1

[Kamailio-Users] presence in kamailio 1.5
by rachid rachid 30 Mar '09

30 Mar '09

hello, i save a sip address (sip:alice@10.193.5.112 <sip%3Aalice(a)10.193.5.112>) in data base Mysql with an equivalent MSN address (alice(a)live.fr) then when i start kamailio1.5 and i save sip adress in X-lite ,the kamailio post a error message: account alice(a)live.fr not found thank you for your help

2 1

[Kamailio-Users] CARRIERROUTE MODULE scan_prefix starting with *
by Asim Riaz 26 Mar '09

26 Mar '09

Hi List, is it possible to add a scan_prefix like *222* in carrierroute? if i add a prefix like this in DB(carrierroute), I dont get any error but Kamailio does not run. any idea why? Thanks, Asim

2 5

[Serusers] Common Log File
by Jiri Kuthan 26 Mar '09

26 Mar '09

I'm wondering what folks think about this the Common Log File (CLF) activity in the IETF? http://www.ietf.org/internet-drafts/draft-gurbani-sipping-clf-01.txt http://tools.ietf.org/html/draft-roach-sipping-clf-syntax-00 Comments to me or directly to the authors welcome. -jiri Abstract Well-known web servers such as Apache and web proxies like Squid support event logging using a common log format. The logs produced using these de-facto standard formats are invaluable to system administrators for trouble-shooting a server and tool writers to craft tools that mine the log files and produce reports and trends. Furthermore, these log files can also be used to train anomaly detection systems and feed events into a security event management system. The Session Initiation Protocol does not have a common log format, and as a result, each server supports a distinct log format that makes it unnecessarily complex to produce tools to do trend analysis and security detection. We propose a common log file format for SIP servers that can be used uniformly for proxies, registrars, redirect servers as well as back-to-back user agents.

1 0

[Kamailio-Users] Problems compiling with Solaris 5.10
by Victor Diaz Diaz 26 Mar '09

26 Mar '09

Hi all, I am experimenting some problems trying to compile "kamailio-1.5.0-tls" with Solaris 5.10. I have installed the next packages: Development/Languages - gcc-3.3.2 - GNU bison v1.35 - GNU flex v2.54 Development/Tools - GNU make utility v3.80 Finally I just be able to compile in "debug" mode, but in whatever other mode I obtain the next result: Makefile:491: warning: overriding commands for target `/' Makefile:482: warning: ignoring old commands for target `/' Compiling action.c gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -mtune=nocona -DNAME='"kamailio"' -DVERSION='"1.5.0-notls"' -DARCH='"x86_64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.6"' -D__CPU_x86_64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"/etc/kamailio/"' -DPKG_MALLOC -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DUSE_LOCAL_ROUTE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -D_POSIX_PTHREAD_SEMANTICS -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o Assembler: action.c "/var/tmp//ccSQUbQn.s", line 294 : Syntax error Near line: " cs ; jl .L47" "/var/tmp//ccSQUbQn.s", line 395 : Syntax error Near line: " cs ; jl .L47" "/var/tmp//ccSQUbQn.s", line 417 : Syntax error Near line: " cs ; jl .L8" "/var/tmp//ccSQUbQn.s", line 1050 : Syntax error Near line: " cs ; jl .L605" "/var/tmp//ccSQUbQn.s", line 1308 : Syntax error Near line: " cs ; jle .L463" "/var/tmp//ccSQUbQn.s", line 2413 : Syntax error Near line: " cs ; jl .L605" "/var/tmp//ccSQUbQn.s", line 2506 : Syntax error Near line: " cs ; jl .L646" "/var/tmp//ccSQUbQn.s", line 2626 : Syntax error Near line: " cs ; js .L688" "/var/tmp//ccSQUbQn.s", line 3430 : Syntax error Near line: " cs ; jl .L618" "/var/tmp//ccSQUbQn.s", line 4514 : Syntax error Near line: " cs ; ja .L324" "/var/tmp//ccSQUbQn.s", line 4622 : Syntax error Near line: " cs ; jl .L318" "/var/tmp//ccSQUbQn.s", line 4743 : Syntax error Near line: " ds ; je .L510" "/var/tmp//ccSQUbQn.s", line 4761 : Syntax error Near line: " cs ; je .L624" "/var/tmp//ccSQUbQn.s", line 4813 : Syntax error Near line: " cs ; jl .L605" "/var/tmp//ccSQUbQn.s", line 5025 : Syntax error Near line: " cs ; jle .L361" "/var/tmp//ccSQUbQn.s", line 5082 : Syntax error Near line: " cs ; jle .L361" "/var/tmp//ccSQUbQn.s", line 5151 : Syntax error Near line: " cs ; jne .L497" "/var/tmp//ccSQUbQn.s", line 5229 : Syntax error Near line: " cs ; je .L361" "/var/tmp//ccSQUbQn.s", line 5282 : Syntax error Near line: " cs ; jl .L146" "/var/tmp//ccSQUbQn.s", line 5318 : Syntax error Near line: " cs ; jl .L146" "/var/tmp//ccSQUbQn.s", line 5371 : Syntax error Near line: " cs ; je .L393" "/var/tmp//ccSQUbQn.s", line 5383 : Syntax error Near line: " cs ; je .L361" "/var/tmp//ccSQUbQn.s", line 5473 : Syntax error Near line: " cs ; jl .L146" "/var/tmp//ccSQUbQn.s", line 5653 : Syntax error Near line: " ds ; jne .L718" "/var/tmp//ccSQUbQn.s", line 5766 : Syntax error Near line: " cs ; jle .L731" "/var/tmp//ccSQUbQn.s", line 6024 : Syntax error Near line: " cs ; jle .L768" make: *** [action.o] Error 1 Do you have some ideas and/or suggestions about what could happen? Thanks in advance. Victor. ------------------------------------------------------------------ This e-mail and the documents attached are confidential and intended solely for the addressee; it may also be privileged. If you receive this e-mail in error, please notify the sender immediately and destroy it. As its integrity cannot be secured on the Internet, the Atos Origin group liability cannot be triggered for the message content. Although the sender endeavours to maintain a computer virus-free network, the sender does not warrant that this transmission is virus-free and will not be liable for any damages resulting from any virus transmitted. Este mensaje y los ficheros adjuntos pueden contener informacion confidencial destinada solamente a la(s) persona(s) mencionadas anteriormente pueden estar protegidos por secreto profesional. Si usted recibe este correo electronico por error, gracias por informar inmediatamente al remitente y destruir el mensaje. Al no estar asegurada la integridad de este mensaje sobre la red, Atos Origin no se hace responsable por su contenido. Su contenido no constituye ningun compromiso para el grupo Atos Origin, salvo ratificacion escrita por ambas partes. Aunque se esfuerza al maximo por mantener su red libre de virus, el emisor no puede garantizar nada al respecto y no sera responsable de cualesquiera danos que puedan resultar de una transmision de virus. ------------------------------------------------------------------

2 3

Re: [Kamailio-Users] t_forward_nonack
by Joao Gomes Pereira 26 Mar '09

26 Mar '09

Hello X-Lite says: "500 Server error occurred". Here is the debug: [19632]: LOG: route 1 [19632]: LOG ---> T_ON_REPLY [19632]: ERROR:tm:t_forward_nonack: no branch for forwarding [19632]: ERROR:tm:w_t_relay: t_forward_nonack failed [19632]: LOG ---> T_RELAY and here is the code: # ROUTE 1 route[1] { log(3, "LOG: route 1 \n"); if (check_route_param("nat=yes")) { setbflag(6); } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); log(3, "LOG ---> T_ON_REPLY \n"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { log(3, "LOG ---> T_RELAY \n"); sl_reply_error(); } exit; } # BRACH ROUTE 1 branch_route[1] { log(3, "LOG: branch_route 1\n"); xdbg("new branch at $ru\n"); } # ON REPLY ROUTE 1 onreply_route[1] { log(3, "LOG: ON REPLY ROUTE 1\n"); xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } # FAILURE ROUTE 1 failure_route[1] { log(3, "LOG: FAILURE ROUTE 1\n"); if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } } What could be missing? Thanks Joao Pereira Daniel-Constantin Mierla wrote: > Hello, > > do you call t_relay() or route(1) in your failure route? > > Post your failure route here. Seems you try to forward a failed > transaction without adding a new branch. > > Cheers, > Daniel > > On 03/24/2009 09:31 PM, Joao Gomes Pereira wrote: >> Hello >> I'm getting this error when I try to do URI calls: >> >> LOG ---> T_ON_REPLY ERROR:tm:t_forward_nonack: no branch for forwarding >> ERROR:tm:w_t_relay: t_forward_nonack failed >> CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) >> for code 408 >> >> And X-Lite says: "Server error occurred" >> >> My config is based in the sample configuration file that comes with >> Kamailio. >> Here is my route[1], where the call should be routed: >> >> >> route[1] { >> >> if (check_route_param("nat=yes")) { >> setbflag(6); >> } >> if (isflagset(5) || isbflagset(6)) { >> route(5); >> } >> >> >> /* example how to enable some additional event routes */ >> if (is_method("INVITE")) { >> #t_on_branch("1"); >> log(3, "LOG ---> T_ON_REPLY \n"); >> t_on_reply("1"); >> t_on_failure("1"); >> } >> >> if (!t_relay()) { >> sl_reply_error(); >> } >> exit; >> } >> >> >> My "branch_route", "on_reply_route" and "failure_route" are just like >> in kamailio.cfg.sample. >> >> What could be causing this errors? >> Thanks >> Regards >> Joao Pereira >> >> >> >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >

2 7

[Kamailio-Users] Kamailio (OpenSER) v1.4.4 Released
by Daniel-Constantin Mierla 25 Mar '09

25 Mar '09

A new release in 1.4 series is out. Kamailio 1.4.4 is based on the latest version of branch 1.4, including many fixes in code and documentation, therefore those running 1.4.0, 1.4.1, 1.4.2 or 1.4.3 are advised to upgrade. /Note that latest stable version of the project is Kamailio 1.5.0, released on March 02, 2009, release that has added many new features since branch 1.4. More details about 1.5.0 at:/ http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes/ <http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes> Kamailio 1.4.4 is provided as a patch fix release for those still running 1.4.x version. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.4.4/src/ <http://www.kamailio.org/pub/kamailio/1.4.4/src> Detailed changelog: http://www.kamailio.org/pub/kamailio/1.4.4/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.4.4/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Modules' documentation: http://www.kamailio.org/docs/modules/1.4.x/ <http://www.kamailio.org/docs/modules/1.4.x> What is new in 1.4.x release series is summarized in the announcement of v1.4.0: http://www.kamailio.org/mos/view/KAMAILIO-v1.4.0-Released -- Daniel-Constantin Mierla

1 0

[Kamailio-Users] Dealing with REGISTER "retransmissions"
by mayamatakeshi 25 Mar '09

25 Mar '09

Hello, we are running kamailio 1.4. Some days ago one of our subscribers SIP phone went crazy and bombarded our proxy with thousands of REGISTER requests. Actually, it didn't wait for any response of the proxy, it just kept sending REGISTER messages (same packet always). Since we are running with nonce_reuse disabled, it stalled the server due to "auth:challenge: failed to generate nonce". I don't have the whole figure but the packet capture I inspected showed that the terminal sent more than 200 thousand REGISTER messages in 15 seconds. We may not be able to replace those terminals at this moment. So to avoid this problem to happen again, we enabled nonce_reuse. However this only will solve the nonce problem but still we will have a lot of hits to database. So I was thinking, maybe I should create a transaction before sending the challenge so that REGISTER retransmissions would be absorbed. I was thinking in doing something like this: if (!proxy_authorize("", "subscriber")) { t_newtran(); proxy_challenge("", "0"); exit; } Does anyone see any problem/shortcoming with this? regards, takeshi

6 11

[Kamailio-Users] portal for OpenSER
by Metshet Seleshi 25 Mar '09

25 Mar '09

Hi all, I am looking to build a protal for openser, I am not having much luck with serMyAdmin, anybody have used this before or any other similar portal recommendation , any sugession will be appriciated, regards

2 1

[Kamailio-Users] xcap and pua modules
by Luntras Loredana 25 Mar '09

25 Mar '09

Hello, I am still trying to install a presence server and a RLS. What I understood is that I have to install the xcap module on Kamailio and a Xcap server. I began with the pua and xcap_client module. When installing xcap_client I have the following errors: xcap_client.c:162: attention : implicit declaration of function «curl_global_cleanup" make[1]: *** [xcap_client.o] Erreur 1 make[1]: quittant le répertoire « /opt/kamailio-1.5.0-tls/modules/xcap_client » make: *** [modules] Erreur 2 And when I am restarting kamailio server (I've added "loadmodule pua.so) I also have an error (I am in the debug mode) Mar 25 10:12:48 [11907] ERROR:core:db_check_table_version: invalid version 5 for table pua found, expected 6 (check table structure and table "version") Mar 25 10:12:48 [11907] ERROR:pua:mod_init: error during table version check. Mar 25 10:12:48 [11907] ERROR:core:init_mod: failed to initialize module pua Mar 25 10:12:48 [11907] ERROR:core:main: error while initializing modules I have kailio 1.5. Thanks for your help, Loredana

2 1

[Kamailio-Users] New Error - Anyone Point me in the direction of what might cause this?
by Geoffrey Mina 25 Mar '09

25 Mar '09

Hello, I have a couple instances of the following error in my logs. I am not sure where to start here. ERROR:core:udp_send: sendto(sock,0x2ac741ab2828,999,0,0x2ac741ab0e50,16): Invalid argument(22) CRITICAL:core:udp_send: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net ERROR:tm:msg_send: udp_send failed ERROR:tm:t_forward_nonack: sending request failed The system is being used for two things. 1 - a PSTN routing engine using the LCR module. I have two gateways in the same group with the same weight, so I suppose I am just doing round-robin load balancing on the two. 2 - a SIP proxy where I send INVITE to the Kamailio server with a Route header set with a fully qualified SIP URI. I am not sure which of these two scenarios could cause this issue... mostly because I don't really know what I am looking at. Please note, this system processed about 25,000+ successful outbound calls today, and this error only cropped up 3 times in the logs. Thanks! Geoff

3 3

[Serusers] about SER load balance
by 静于 25 Mar '09

25 Mar '09

hi~ I wonder is there any load balance sheme used by SER? If there are multiple servers responsible for a domain. One is used as main server. Suppose main server is full weight load. To which server the main server decides to relay? ___________________________________________________________ 好玩贺卡等你发，邮箱贺卡全新上线！ http://card.mail.cn.yahoo.com/

1 0

[Kamailio-Users] Using OpenSer as presence server, with OpenIMS and UCT client
by Luntras Loredana 24 Mar '09

24 Mar '09

Hello all, I have a new problem. I am using Open SER (1.3.3 tls) as a presence server with an Open IMS Core. On the event presence, the OpenIMS server redirects to OpenSER. I am also using UCTImsclient for testing. The registration is ok, the messages also. The problem is the presence: it works once and after that it doesn't work all the day. Have anyone tested all theses? Thanks for your help, Loredana

5 10

[Serusers] lookup_contacts with IP's and URL's
by Bill McNamara 24 Mar '09

24 Mar '09

> We are having something of an issue with lookup_user() and > lookup_contacts(). > > > > Our UA's are added to the proxy using "ser_ctl user add > 1234567890(a)ourdomain.com -p password" which works great > > And the UA's register just fine. > > > > When a call comes in from a carrier, the invite is for > 1234567890(a)12.34.56.78 where 12.34.56.78 is the IP address of our proxy. > > > > When we do a lookup to see if the call is destined for a local user via > lookup_user("$tu.uid","@ruri") it fails because the registration is done > using the URL and not the IP address. >>If you are not yet using the domain module, you should start doing so. >>You can then add both ourdomain.com and 12.34.56.78 to the same domain >>id, which you in turn use in the uri table for your users. Thanks Martin. I saved the current UA's from ser.credentials, then using ser_ctl removed them and the domain. I added the domain plus the URL as an alias and recreated the UA's using ser_ctl. Everything now works great. Thanks again! Bill

1 0

[Kamailio-Users] [dialog module] set_dlg_profile() not working?
by Iñaki Baz Castillo 24 Mar '09

24 Mar '09

Hi, perhaps I miss something but I don't get set_dlg_profile() working: ---------------- modparam("dialog", "dlg_flag", FLAG_DIALOG) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "profiles_no_value", "inbound ; outbound") if is_method("INVITE") { setflag(FLAG_DIALOG); set_dlg_profile("outbound"); ... ---------------- But when I run "kamctl fifo profile_list_dlgs outbound" I see nothing (the dialog does appear listed in "kamctl fifo dlg_list"). I've also added into the loose_route section: if (is_in_profile("outbound")) { xlog("L_CRIT", "*** this request belongs to an outbound call\n"); } This should appear when the ACK-200 arrives, also in any re-INVITE or BYE, but it doesn't ¿? Do I miss something? Thanks. -- Iñaki Baz Castillo

2 10

[Kamailio-Users] [Re: Re: BIGINT in avp]
by kokoska rokoska 24 Mar '09

24 Mar '09

Daniel-Constantin Mierla napsal(a): > Hello, > > On 03/21/2009 11:30 AM, kokoska.rokoska wrote: >> Hi all, >> >> I have small (and may be stupid) question: >> >> could I store BIGNT from MySQL DB into avp using avp_db_query? And if >> yes, how? >> > no, avps can hold only string or integer values. Thank you very much, Daniel, for your answer! Maybe you can get what > you want via string - keep the bigint as string. Probably in the future > will take this in consideration. > I'm affraid I can't do that, 'cause it is primamy key and there are a lot of foreign keys linked to it. And due to performance loss I worry about to change all of them to strings... May be I should "cast" them (by SELECT) to strings. > Now the focus is to integrate http://sip-router.org -- several modules > left there and then we can take new features request on board: > http://sip-router.org/wiki/devel/kamailio-integration#module_integration_st… > :-) Thanks once more, Daniel! Best regards, kokoska.rokoska

2 2

[Kamailio-Users] FAX T38 failed...Help
by michel freiha 24 Mar '09

24 Mar '09

Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW-->Asterisk-->kamailio-->Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '76c64f8373def68d5088690816c67493@Asterisk_IP'. Giving up On kamilio everything looks great except the received value in the INVITE message...I guess it should be the Public IP of the endpoint and not asterisk IP address... Can you please help me in order to find the real issue? the log is under http://pastebin.com/d1e7ba841 Regards

2 1

[Kamailio-Users] BIGINT in avp
by kokoska.rokoska 24 Mar '09

24 Mar '09

Hi all, I have small (and may be stupid) question: could I store BIGNT from MySQL DB into avp using avp_db_query? And if yes, how? BTW: I tried to do it, but avp was always null. When I tried sql_query, I saw one row was red (with one column) but $dbr(ra=>[0,0]) was empty. If I changed colmun (well, better to say, I changed returning type from my stored function) to INT, everything worked well... Any hint is welcome :-) Best regards, kokoska.rokoska

2 1

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