- sr-users - kamailio.org

Users are not registering on my Kamailio
by Toyima Dias 18 Aug '10

18 Aug '10

HEllo community, I'm having some problems with my kamailio configuration, following is the section of my authentication process on my /etc/kamailio/kamailio.cfg route [3] { if (is_method("REGISTER")) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403", "Forbidden auth ID"); exit; } } else { 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(); } } return; } I added a user with the following command: # kamctl add 1102 123456 And then i checked on mysql, table subscriber: mysql> select * from subscriber; +----+----------+---------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+---------------+----------+---------------+----------------------------------+----------------------------------+------+ | 3 | 1102 | dannydias.com | | | fe7a80817383cfd90950dea03d37eccf | dbb9b6ef93d4f3417038a6bf5cd176db | NULL | +----+----------+---------------+----------+---------------+----------------------------------+----------------------------------+------+ 1 row in set (0.00 sec) By the way, the domain field does not exist (dannydias.com) i'm not sure how this could affect the authentication process, this migh be the problem? And finally, ngrep-sip to check what is going on: # U 2010/08/17 22:28:38.548357 80.26.81.96:26932 -> 212.230.19.191:5060 *REGISTER sip:212.230.19.191 SIP/2.0'* Via: SIP/2.0/UDP 80.26.81.96:26932;branch=z9hG4bK-d2ee4e38' From: "1102" <sip:1102@212.230.19.191 <sip%3A1102(a)212.230.19.191> >;tag=36dcc698cfc16210o1' To: "1102" <sip:1102@212.230.19.191 <sip%3A1102(a)212.230.19.191>>' Call-ID: 4aab2f50-775b4d34(a)80.26.81.96' CSeq: 10409 REGISTER' Max-Forwards: 70' Authorization: Digest username="1102",realm="212.230.19.191",nonce="4c6b0d34000000c9c4e2068de2d41bec38c6d23029696501",uri="sip:212.230.19.191",algorithm=MD5,response="a09ea335a01051d714d2cbfe95e17820"' Contact: "1102" <sip:1102@80.26.81.96:26932>;expires=180' User-Agent: Linksys/SPA942-5.1.15(a)' Content-Length: 0' Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER' Supported: replaces' ' # U 2010/08/17 22:28:38.548357 212.230.19.191:5060 -> 80.26.81.96:26932 *SIP/2.0 401 Unauthorized'* Via: SIP/2.0/UDP 80.26.81.96:26932;branch=z9hG4bK-d2ee4e38' From: "1102" <sip:1102@212.230.19.191 <sip%3A1102(a)212.230.19.191> >;tag=36dcc698cfc16210o1' To: "1102" <sip:1102@212.230.19.191 <sip%3A1102(a)212.230.19.191> >;tag=00f209f5fddbfe1aa952cd699e41d531.97e5' Call-ID: 4aab2f50-775b4d34(a)80.26.81.96' CSeq: 10409 REGISTER' WWW-Authenticate: Digest realm="212.230.19.191", nonce="4c6b0d34000000caa164dcad25dbc5b4c6baefcef6dc84ae"' Server: Kamailio (1.5.4-notls (x86_64/linux))' Content-Length: 0' ' Could somebody help me? i can't find the real problem Thanks in advance DD

2 1

module load path and -L
by Daniel-Constantin Mierla 18 Aug '10

18 Aug '10

Hello, the path to modules can be specified via command line option -L or cfg parameter loadpath (aliased as mpath for K backward compat). loadpath overwrites the value given by -L. The question is whether the value given via -L shouldn't be effective all the time (i.e., loadpath cfg parameter ignored when -L is given) since it is via command line parameter -- my expectation is command line options should have higher priority. Changing it now will break backward compatibility (if anyone used it like now) while in testing, but if all think is a bug, then can be fixed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

2 2

set From and To attributes in assignments
by Daniel-Constantin Mierla 17 Aug '10

17 Aug '10

Hello, I committed in master branch (to be 3.1.0) code that allows to set uri, username, domain and display name for To and From headers using assignments to their respective PVs in configuration file. For example: $fu = "sip:anonymous@invalid"; $fn = '"Jon Doe"'; $tU = "+123455678"; Assignment of each such attribute should be done only once, otherwise you get concatenated values since it uses the internal lump system. Therefore, doing an update is not visible immediately in config unless you do use msg_apply_changes(). Also, use it carefully, in case you have sipv1 devices in your network then it can break dialog matching. Anyhow changing content of From and To headers was possible with remove_hf()/append_hf() or subst() functions, the new feature comes just to ease writing the config when one needs it. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

4 4

Freezing for 3.1 on Monday, Aug 16
by Daniel-Constantin Mierla 17 Aug '10

17 Aug '10

Hello, as I could spot, there was some intense activity from Andrei in the last days with a new internal statistics API. To allow some time for it as well as for others (like me) to get some work done that is good to be in 3.1, I propose next Monday (Aug 16) as timeline to freeze the repo for new features. Probably during the freeze we can still merge some modules (in the way of moving one version to modules and removing the other as long as remaining version includes the features of the other) -- thinking of pdt, ratelimit, sms, ... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

3 6

Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy
by JinKevin 17 Aug '10

17 Aug '10

Guys, Anyone has some feedback on this? Thanks, Kevin Hi All, I have the kamailio configed as below on Solaris 10 (Sparc): # ----- nathelper ----- #!ifdef WITH_NAT modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") 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", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif and rtpproxy is started as AppSer01:ipunity@/opt/ipunity$ ps -ef |grep rtp ipunity 4687 1 0 12:58:20 ? 0:00 /usr/local/bin/rtpproxy -l 210.xx.xx.xx -s udp:localhost:7722 AppSer01:ipunity@/opt/ipunity$ netstat -an |grep 7722 127.0.0.1.7722 Idle 127.0.0.1.41509 127.0.0.1.7722 Connected 127.0.0.1.41510 127.0.0.1.7722 Connected 127.0.0.1.41511 127.0.0.1.7722 Connected 127.0.0.1.41512 127.0.0.1.7722 Connected 127.0.0.1.41513 127.0.0.1.7722 Connected 127.0.0.1.41514 127.0.0.1.7722 Connected 127.0.0.1.41515 127.0.0.1.7722 Connected 127.0.0.1.41516 127.0.0.1.7722 Connected 127.0.0.1.41517 127.0.0.1.7722 Connected 127.0.0.1.41518 127.0.0.1.7722 Connected 127.0.0.1.41519 127.0.0.1.7722 Connected However, kamailio is not able to connect the rtpporxy, Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 788202 local0.error] ERROR: nathelper [nathelper.c:2457]: can't send command to a RTP proxy Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 844113 local0.error] ERROR: nathelper [nathelper.c:2492]: proxy <udp:127.0.0.1:7722> does not respond, disable it Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:3144]: no available proxies Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 788202 local0.error] ERROR: nathelper [nathelper.c:2457]: can't send command to a RTP proxy Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 844113 local0.error] ERROR: nathelper [nathelper.c:2492]: proxy <udp:127.0.0.1:7722> does not respond, disable it Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:3144]: no available proxies Jul 24 12:59:04 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22870]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:2627]: no available proxies Jul 24 12:59:04 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22870]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:2627]: no available proxies Any helps are appreciated. Thanks, Kevin 聊天+搜索+邮箱想要轻松出游,手机MSN帮你搞定! 立刻下载！ _________________________________________________________________ MSN十年回馈，每位用户可免费获得价值25元的卡巴斯基反病毒软件2010激活码，快来领取！ http://kaba.msn.com.cn/?k=1

3 16

Re: [SR-Users] Kamailio doesn't start
by Ernest Mavrel 17 Aug '10

17 Aug '10

2 4

please help me with domainpolicy module.
by truong ngoc THANH 15 Aug '10

15 Aug '10

hi all, I try to configure domainpolicy but it does not work. =========== my scenarios is : 1: i configure multi domain with kamailio, and i set two domain example.com and kamailio.com which have the same ip(192.168.1.40). 2:the sip client can register and make call in local domain . 3: i try to make policy in domain example.com to call to kamailio.com. my config as : ============ i add some line in kamailio.cfg loadmodule "domainpolicy.so" #---------------------domainpolicy---------------------------- modparam("domainpolicy", "db_url", "mysql://openser:openserrw@localhost/kamailio") modparam("domainpolicy", "dp_table", "supportedpolicies") modparam("domainpolicy", "dp_col_rule", "rules") modparam("domainpolicy", "dp_col_type", "type") modparam("domainpolicy", "dp_col_att", "att") modparam("domainpolicy", "dp_col_val", "val") modparam("domainpolicy", "port_override_avp", "portoverride") modparam("domainpolicy", "transport_override_avp", "transportoverride") modparam("domainpolicy", "domain_replacement_avp", "domainreplacement") modparam("domainpolicy", "domain_prefix_avp", "domainprefix") modparam("domainpolicy", "domain_suffix_avp", "domainsuffix") modparam("domainpolicy", "send_socket_avp", "sendsocket") if (is_method("INVITE")) { setflag(1); # do accounting dp_can_connect();#< i add dp_apply_policy();#< i add in mysql i create table : and add rules as : mysql> select * from supportedpolicies; +----+--------------+------+-------------------+--------------+--------------+-------------------+-------------------+--------------+--------------+-------------+ | id | rules | type | att | val | portoverride | transportoverride | domainreplacement | domainprefix | domainsuffix | description | +----+--------------+------+-------------------+--------------+--------------+-------------------+-------------------+--------------+--------------+-------------+ | 5 | kamailio.com | dom | domainreplacement | kamailio.com | 5060 | udp | kamailio.com | NULL | NULL | | | 6 | example.com | dom | domainreplacement | kamailio.com | NULL | NULL | kamailio | NULL | NULL | | +----+--------------+------+-------------------+--------------+--------------+-------------------+-------------------+--------------+--------------+-------------+ 2 rows in set (0.00 sec) the log when i make call from 1(a)example.com to 3(a)kamailio.com Aug 14 10:16:41 appliance /usr/local/sbin/kamailio[2140]: INFO: domainpolicy [domainpolicy.c:547]: looking up Domain itself: example.com Aug 14 10:16:41 appliance /usr/local/sbin/kamailio[2140]: INFO: domainpolicy [domainpolicy.c:429]: checking for 'example.com'. Aug 14 10:16:41 appliance /usr/local/sbin/kamailio[2140]: INFO: domainpolicy [domainpolicy.c:473]: querying database OK Aug 14 10:16:41 appliance /usr/local/sbin/kamailio[2140]: INFO: domainpolicy [domainpolicy.c:506]: DB returned domainreplacement/kamailio.com Aug 14 10:16:41 appliance /usr/local/sbin/kamailio[2140]: INFO: domainpolicy [domainpolicy.c:551]: found a match on domain itself Please help me to correct config to use domainpolicy thanks for help.peter_green

2 1

Issue using subst
by Uriel Rozenbaum 15 Aug '10

15 Aug '10

Hi guys, I'm having a weird situation when trying to modify some parts of the RPID on my config file. I'm using Kamailio 1.5.3-notls The code is the following: ... subst('/^Remote-Party-ID:(.*)sip:(.*)@(.*)$/Remote-Party-ID:\1sip:$avp(s:ANIegress)@\3/ig'); subst('/^Remote-Party-ID:(.*)"(.*)"(.*)sip:(.*)$/Remote-Party-ID:\1"$avp(s:ANIegress)"\3sip:\4/ig'); ... With the first line I'm trying, to change the user part of the SIP-URI to the value in $avp(s:ANIegress). Then i just replace the token between double quotes (if any) with the value in $avp(s:ANIegress). But when I check the captured (ngrep) results I have the following: Before: Remote-Party-ID: "60911000" <sip:541160911100@192.168.200.100>;privacy=off;screen=no. After: Remote-Party-ID: "60911000" <sip:1160911100@192.168.200.100>;privacy=off;screen=no.Remote-Party-ID: "1160911100" <sip:541160911100@192.168.200.100>;privacy=off;screen=no. Can anyone point me on the direction to solve the issue, or a better way to replace the values in RPID, PAI (and possibly From Header for some providers)? Thanks in advence! Uriel

3 6

Is this a BUG?
by dotnetdub 12 Aug '10

12 Aug '10

Hi All, I see this in every call setup.. CRITICAL: dialog [dlg_hash.c:591]: bogus event 6 in state 2 for dlg 0xaf933bd0 [3739:1530909163] with clid ' 5959362a340dfa7938cf52ae1ba4b9d7(a)my.proxy.com' and tags 'as5b636a8a' '' I am running SIP-Router 3.01 I note this thread: http://www.mail-archive.com/sr-dev@lists.sip-router.org/msg05668.html but it's inconclusive. It doesn't seem to cause any issue but I would like to get it sorted. Thanks Brian

5 9

[Kamailio-Users] kamailio and PRACK
by BERGANZ François 10 Aug '10

10 Aug '10

Hello, Does kamailio support PRACK method ? If yes, how? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

5 4

Billing
by Lens Oreste 10 Aug '10

10 Aug '10

Are you looking for an affordable billing solution for Kamailio? if so SGT got you cover, we are a commercial grade billing. We offer hosted billing solution for kamailio and opensips. Our billing is radius based and fully integrated with kamailio/opensips. please send us an email at info at sgthire.com or support at sgthire dot com

2 1

time formats (specifically in db_flatstore)
by SIP Projects 10 Aug '10

10 Aug '10

Using version 1.5.2. Trying to migrate acc from db_mysql to db_flatstore to reduce io wait during heavy dialing. However, the default time format changes from normal YYYY-MM-DD hh:mm:ss to unixtime (I think) in flatstore. I tried using $TF but its format is even more non-standard. Is there a way to change time format for either the default acc time field in the db_flatstore module or $TF so I can 'LOAD DATA' into mysql as datetime? Thanks, Matt

3 5

PDT ran out of pkg memory using pdt_list
by JR Richardson 10 Aug '10

10 Aug '10

Hi All, I loaded up the PDT database with about 35K records and when I issue the commad "kamctl fifo pdt_list" I get: 3(3018) ERROR: <core> [tree.c:139]: no more pkg mem 3(3018) ERROR: mi_fifo [fifo_fnc.c:509]: command (pdt_list) processing failed Searching around I found the http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory which suggest I adjust the pkg mem size in config.h. In config.h, I found: /*used only if PKG_MALLOC is defined*/ #define PKG_MEM_POOL_SIZE 4*1024*1024 So is this what I am supposed to adjust? Maybe try: #define PKG_MEM_POOL_SIZE 4*2048*2048 or #define PKG_MEM_POOL_SIZE 8*1024*1024 Which one would be a logical adjustment? Also, is there a correlation between pkg mem and database record size as related to pdt_list? The idea is to have a few 100K records loaded in kamailio and be able to perform "kamctl fifo pdt_list | grep 512345" to show this prefix route. But without enough memory, doesn't work. Thanks. JR -- JR Richardson Engineering for the Masses

8 29

[kamailio-users] floating value support
by Chandrakant Solanki 09 Aug '10

09 Aug '10

Hi Is kamailio support floating point operation like.. num1 = 1 num2 = 3 num3 = num1 / num2 i. e. 0.33 = 1 / 3 Is it possible...?? If yes, How ?? Any Idea?? -- Regards, Chandrakant Solanki

2 1

sip trace not logging the first invite
by alex pappas 09 Aug '10

09 Aug '10

Dear friends, I'm using the sip_trace module in order to log into a mysql DB all the SIP messages that are passing from my kamailio. I have the setflag(4) on the top of the route[0] and I have all the sip messages logged in the mysql Db. The problem is that I don't see the first INVITE of the call. Scenario: gateway1 --------------> kamailio ----------------> gateway2 The INVITE from gateway1 to kamailio is not logged. The trace is starting from the first 100 trying message. Is anyone aware on where the problem is ? P.S. By using the sip_trace() method I get the first INVITE but not the rest of the trace in the way that setflag(4) is logging. Thank you in advance Alex

2 1

Problem making Kamailio 3.0
by Uriel Rozenbaum 09 Aug '10

09 Aug '10

Hi guys, At the risk of showing unmatched stupidity I'm asking these few questions. I've downloaded the tarball sources for Kam3.0.2, uncompressed them and run make on a virtual machine that used to have Kam1.5.3. I edited the CFG file to make it work with the new version but still I cannot make it start. I get the following module errors: Aug 3 17:52:49 sbcdesa kamailio: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/db_mysql.so>: /usr/local/lib/kamailio/modules/db_mysql.so: undefined symbol: debug Aug 3 17:52:49 sbcdesa kamailio: : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 40, column 12-24: failed to load module Aug 3 17:52:49 sbcdesa kamailio: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/carrierroute.so>: /usr/local/lib/kamailio/modules/carrierroute.so: undefined symbol: debug Aug 3 17:52:49 sbcdesa kamailio: : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 67, column 12-28: failed to load module In my cfg file I've set mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" as stated on the documentation and the contents of thos folders are: [root@sbcdesa kamailio-3.0.2]# ls /usr/local/lib/kamailio/modules/ acc.so carrierroute.so db_text.so exec.so mi_fifo.so pdt.so rr.so speeddial.so uac_redirect.so alias_db.so cfg_db.so dialog.so group.so mi_rpc.so permissions.so rtimer.so sqlops.so uac.so auth_db.so cfg_rpc.so dispatcher.so htable.so msilo.so pike.so sanity.so sst.so uri_db.so auth_diameter.so cfgutils.so diversion.so imc.so nathelper.so pv.so siptrace.so statistics.so userblacklist.so auth.so ctl.so domainpolicy.so maxfwd.so nat_traversal.so qos.so siputils.so textops.so usrloc.so avpops.so db_flatstore.so domain.so mediaproxy.so path.so ratelimit.so sl.so tm.so xlog.so benchmark.so db_mysql.so enum.so mi_datagram.so pdb.so registrar.so sms.so topoh.so [root@sbcdesa kamailio-3.0.2]# ls /usr/local/lib/kamailio/modules_k/ acc.so call_control.so domainpolicy.so imc.so nathelper.so pua_mi.so rtimer.so sqlops.so uac.so alias_db.so cfgutils.so domain.so kex.so nat_traversal.so pv.so siptrace.so sst.so uri_db.so auth_db.so db_text.so drouting.so maxfwd.so path.so qos.so siputils.so statistics.so userblacklist.so auth_diameter.so dialog.so exec.so mi_datagram.so pdt.so ratelimit.so sl.so textops.so usrloc.so auth.so dispatcher.so group.so mi_fifo.so permissions.so registrar.so sms.so tmx.so xlog.so benchmark.so diversion.so htable.so msilo.so pike.so rr.so speeddial.so uac_redirect.so Is there anything I'm missing? Should I delete everything and start over? I reviewed the makefile to be sure I was not excluding modules, maybe they are excluded somewhere else. Thanks in advance, Uriel

2 1

Help needed for OpenSer with Radius
by Pratik Shrestha 06 Aug '10

06 Aug '10

Dear All, I am very new to OpenSer. I want to use latest version of OpenSer with Radius. I need the documentation/tutorial on how to do this. Googling, Ionly found for the old version. Please help me. Regards, Pratik

3 8

Kamailio doesn't start
by Ernest Mavrel 06 Aug '10

06 Aug '10

Hi I have problems with starting of kamailio. When is rule in table dialplan, kamailio doesn't start. Dialplan table: id dpid pr match_op match_exp match_len subst_exp repl_exp attrs 1 1 0 1 ^0[1-9] 0 ^(0)([1-9])(.+) 386\2 Cut_0_Add_386 Log: 0(11570) DEBUG: dialplan [dp_db.c:200]: init 0(11570) DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x95ff60 0(11570) DEBUG: db_mysql [km_res.c:68]: 8 columns returned from the query 0(11570) DEBUG: <core> [db_res.c:155]: allocate 64 bytes for result names at 0x960028 0(11570) DEBUG: <core> [db_res.c:165]: allocate 32 bytes for result types at 0x960118 0(11570) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x95fe18 0(11570) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x95fe18)[0]=[dpid] 0(11570) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type 0(11570) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x960310 0(11570) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x960310)[1]=[pr] 0(11570) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type 0(11570) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x960350 0(11570) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x960350)[2]=[match_op] 0(11570) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type 0(11570) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x960330 0(11570) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x960330)[3]=[match_exp] 0(11570) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type 0(11570) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[4] at 0x9602f0 0(11570) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9602f0)[4]=[match_len] 0(11570) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type 0(11570) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[5] at 0x9602d0 0(11570) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9602d0)[5]=[subst_exp] 0(11570) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type 0(11570) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[6] at 0x9602b0 0(11570) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9602b0)[6]=[repl_exp] 0(11570) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type 0(11570) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[7] at 0x960290 0(11570) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x960290)[7]=[attrs] 0(11570) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type 0(11570) DEBUG: db_mysql [km_dbase.c:342]: converting row 0 of 1 count 1 0(11570) DEBUG: <core> [db_row.c:119]: allocate 256 bytes for row values at 0x960370 0(11570) DEBUG: <core> [db_val.c:73]: converting INT [1] 0(11570) DEBUG: <core> [db_val.c:73]: converting INT [0] 0(11570) DEBUG: <core> [db_val.c:73]: converting INT [1] 0(11570) DEBUG: <core> [db_val.c:117]: converting STRING [^0[1-9]] 0(11570) DEBUG: <core> [db_val.c:73]: converting INT [0] 0(11570) DEBUG: <core> [db_val.c:117]: converting STRING [^(0)([1-9])(.+)] 0(11570) DEBUG: <core> [db_val.c:117]: converting STRING [386\2] 0(11570) DEBUG: <core> [db_val.c:117]: converting STRING [Cut_0_Add_386] 0(11570) DEBUG: dialplan [dp_db.c:374]: build_rule Any help? BR Ernest

3 20

Re: [SR-Users] Problem with t_relay() and VPN tunnel
by Alex Balashov 06 Aug '10

06 Aug '10

On 08/05/2010 07:58 PM, Carlos Ruiz Diaz wrote: > 2) I am sorry, I think I did not understand the question. Did you mean > if the messag exists when leaving tun0 in *B* going to tun0 in *A*? My concern is that the packet is going out on eth0 despite being intended to go out of tun0. Try to do a packet capture on eth0 and look for packets addressed to 10.8.0.0/24, which of course will not reach their destination--but that's not the point. Also, do you have the 'mhomed' option set to 1 in your Kamailio config? http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#mhomed It is a requirement in this case. > > /tcpdump udp /shows packets coming from *B* when perl is running, but no > packets are received using t_relay() and kamailio That is because the source socket on the Perl script side is probably interface-agnostic and just uses the IP routing table plainly. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 2

Recommendation
by Carsten Bock 05 Aug '10

05 Aug '10

Hi everybody, a former colleague and friend of mine is searching for a new job. He is currently working as NOC-Support-Engeneer and has good knowledge of the SIP-protocol and about running/supporting Kamailio (and supporting systems, e.g. databases). He would prefer to work in northern Germany; but that is not a must but more like a "nice-to-have". You can find his profile on Xing: https://www.xing.com/profile/Marcel_Weinberg3 I can provide further information/contact information if someone is interested. Kind regards, Carsten

1 0

help with configure kamailio authenticate with freeradius server.
by thanh.truong＠htklabs.com 05 Aug '10

05 Aug '10

Hi all I try to configure kamailio to register sip client with freeradius via kamailio. My configure as: ###################kamailio.cfg############# #!KAMAILIO # # $Id$ # # Kamailio (OpenSER) SIP Server v3.0 - basic configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: #!define WITH_DEBUG # # *** To enable mysql: #!define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql #!define WITH_AUTH # - add users using 'kamctl' # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no 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:10.0.0.10:5060 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.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 "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_radius.so" #!endif /* 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" #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!endif # ----------------- 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") /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") #!endif # ----- auth_db params ----- /* enable the DB based authentication */ #!ifdef WITH_AUTH #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") #modparam("auth_db", "db_url", # "mysql://openser:openserrw@localhost/kamailio") #modparam("auth_db", "load_credentials", "") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("auth_radius", "auth_extra", "Acct-Session-Id=$ci") modparam("auth_radius", "use_ruri_flag", 22) #!endif # ----- 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 ----- /* enable presence server support */ #!ifdef WITH_PRESENCE modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:10.0.0.10:5060") #!endif # ----- nathelper ----- #!ifdef WITH_NAT modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") 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) #!endif ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } # NAT detection route(NAT); 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(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); 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(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) 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"); route(RELAY); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(PRESENCE); if (is_method("REGISTER")) { if(isflagset(5)) { setbflag("6"); # uncomment next line to do SIP NAT pinging ## setbflag("7"); } if (!save("location")) sl_reply_error(); exit; } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(PSTN); # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($rc) { 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(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag("6"); } if (isflagset(5) || isbflagset("6")) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence server route route[PRESENCE] { #!ifdef WITH_PRESENCE 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; #!endif # 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 route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!radius_www_authorize("192.168.1.70")) { www_challenge("192.168.1.70", "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 (!radius_proxy_authorize("192.168.1.70")) { proxy_challenge("192.168.1.70", "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 } } #!endif return; } # Caller NAT detection route route[NAT]{ #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset("6") || isflagset(5))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { 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(); ##} } ############################################## I run command : kamctl start But : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed And I run : tail -f /var/log/message to show log: The log is: Aug 4 15:05:07 appliance kamailio: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/auth_radius.so>: libradiusclient-ng.so.2: cannot open shared object file: No such file or directory Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 145, column 12-27: failed to load module Aug 4 15:05:07 appliance kamailio: ERROR: <core> [modparam.c:154]: set_mod_param_regex: No module matching <auth_radius> found Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 234, column 92: Can't set module parameter Aug 4 15:05:07 appliance kamailio: ERROR: <core> [modparam.c:154]: set_mod_param_regex: No module matching <auth_radius> found Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 235, column 43: Can't set module parameter Aug 4 15:05:07 appliance kamailio: ERROR: <core> [modparam.c:154]: set_mod_param_regex: No module matching <auth_radius> found Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 236, column 60: Can't set module parameter Aug 4 15:05:07 appliance kamailio: ERROR: <core> [modparam.c:154]: set_mod_param_regex: No module matching <auth_radius> found Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 237, column 44: Can't set module parameter Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 479, column 50: unknown command, missing loadmodule? Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 494, column 53: unknown command, missing loadmodule? ############### I try to add : /usr/local/lib/* to /etc/ld.so.conf And run : ldconfig -v But the result is the same error . Please help me if someone know what is wrong. Any suggest is welcome. Thanks and Regards Thanh.Truong

2 1

help with sip client register with freeradius via kamailio.
by thanh.truong＠htklabs.com 05 Aug '10

05 Aug '10

Hi all. Please help me with my problem. I trying to set up kamailio and freeradius. I want sip client authenticate with freeradius server, but for a long time I can make it work. I have referenced many reference document but it cannot work. My configure file as my attachment . Please suggest if you have ideal. Kamailio -3.0.1 freeradius-1.1.3-1.5.el5_4.src.rpm radiusclient-ng-0.5.6 please help me as soon as possible. Thanks and regards.

2 1

DBI Error on startup
by "Nicolas Rüger" 05 Aug '10

05 Aug '10

Hello, I'm using debian lenny, kamailio_3.0.0 and perl_5.10.0. I included the following line in a PERL-script (that I want to be started from the routing-logic in kamailio.cfg) for interaction of PERL with a MySQL Database: use DBI; The problem is that I get the follwing error at kamailio-startup: lenny:/opt/kamailio-3.0.0/sbin# 0(17385) ERROR: perl [openserxs.xs:1027]: perl error: Can't load '/usr/lib/perl5/auto/DBI/DBI.so' for module DBI: /usr/lib/perl5/auto/DBI/DBI.so: undefined symbol: PL_memory_wrap at /usr/lib/perl/5.10/DynaLoader.pm line 196. at /usr/lib/perl5/DBI.pm line 263 The PERL-script itself compiles without problems and other scripts work fine as well. What can I do??? Thanks for the help in advance. Regards, Nicolas -- GMX DSL: Internet-, Telefon- und Handy-Flat ab 19,99 EUR/mtl. Bis zu 150 EUR Startguthaben inklusive! http://portal.gmx.net/de/go/dsl

2 2

[OT] Problem with <sr-users@lists.sip-router.org> subscription
by Iñaki Baz Castillo 02 Aug '10

02 Aug '10

Hi, each N weeks I receive the following mail from sr-users-request(a)lists.sip-router.org: --------------------- Your membership in the mailing list sr-users has been disabled due to excessive bounces The last bounce received from you was dated 26-Jul-2010. You will not get any more messages from this list until you re-enable your membership. You will receive 4 more reminders like this before your membership in the list is deleted. -------------------- Not sure what could be the problem as I receive mail in ibc(a)aliax.net with no problem. Any tip please? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

INumWare: Inum Users Conference
by Meftah Tayeb 31 Jul '10

31 Jul '10

hello folk you're invited to join the inum ware conference this conference is for inum users that frequantly use inum (+883 country code) (http://www.inum.net) you can join it by dialing +883510001288888 from any provider that provide INum termination (see http://www.inum.net) thank you! -- Meftah Tayeb algérie télécom SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz

1 0

Loops using uac_replace_from
by Ján ONDREJ (SAL) 30 Jul '10

30 Jul '10

Hello, after some failures with uac_auth (uac_auth can't increase CSeq number, which is required by my provider) my provider maked me an testing access without autentification, but I still can't make calls with kamailio or openser. I can make calls with opensips, but I am still not chosen, which sip software to use in future (still using openser). This is a part of my configuration: route[6] { uac_replace_from("sip:556807505@as.xxxxxx.sk"); rewritehostport("122.33.44.200"); route(5); } When trying to sniff packets and diff them, I am still getting this from my kamailio server: U kamailio_ip:5060 -> client_ip:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL). From: "Ondrej Jan" <sip:xxxxxx@xxxx>;tag=59427D6F-81E63720. To: <sip:xxxxx@xxxxx;user=phone>;tag=cc70c0fe52d7af0accfb78a94d592620.f69c. CSeq: 1 INVITE. Call-ID: 37cadcfb-f93b8e1d-431cdf6@xxxxxxxxxxx. Server: kamailio (3.0.2 (i386/linux)). Content-Length: 0. Logs are below. Any idea, what happening and why a default opensips configuration works and kamailio/openser don't? SAL [root@kamailio ~]# kamailio -DddE loading modules under /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ Listening on udp: 123.123.242.20:5060 Aliases: udp: kamailio.xxxx.sk:5060 *: 123.123.242.20:* WARNING: no fork mode 0(2364) INFO: <core> [tcp_main.c:4150]: init_tcp: using epoll_lt as the io watch method (auto detected) 0(2364) INFO: usrloc [hslot.c:53]: locks array size 512 0(2364) INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2364) INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 4(2368) INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_lt io watch method (config) 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: <core> [action.c:1269]: WARNING: too many recursive routing table lookups (101) giving up! 0(2364) ERROR: <core> [action.c:1269]: WARNING: too many recursive routing table lookups (101) giving up! 0(2364) ERROR: <core> [action.c:1269]: WARNING: too many recursive routing table lookups (101) giving up! 0(2364) BUG: tm [t_lookup.c:1556]: tm: t_unref: REQ_ERR DELAYED should have been caught much earlier for 0xb5a15490: 17 (hex 11) ^C 4(2368) INFO: <core> [main.c:788]: INFO: signal 2 received 3(2367) INFO: <core> [main.c:788]: INFO: signal 2 received 1(2365) INFO: <core> [main.c:788]: INFO: signal 2 received 2(2366) INFO: <core> [main.c:788]: INFO: signal 2 received 0(2364) ALERT: <core> [main.c:719]: child process 2365 exited normally, status=0 0(2364) ALERT: <core> [main.c:719]: child process 2366 exited normally, status=0 0(2364) ALERT: <core> [main.c:719]: child process 2367 exited normally, status=0 0(2364) ALERT: <core> [main.c:719]: child process 2368 exited normally, status=0 0(2364) INFO: <core> [main.c:734]: INFO: dont_fork turned on, living on 0(2364) NOTICE: <core> [main.c:679]: Thank you for flying kamailio

3 2

[Kamailio-users] Weight based routing
by Chandrakant Solanki 30 Jul '10

30 Jul '10

Hello, I have setup 3 SIP server. It should be Kamailio, Asterisk or Any SIP Server. I have setup 3 server say Server A, Server B and Server C and each has weight like 50, 30, 20 in percentage. And I taken 10 calls and try to forward call based on weight. Is this possible in kamailio's module.. and if yes then which module will help me? Any Idea.. -- Regards, Chandrakant Solanki

4 6

Modifying replies in failure route
by Alex Balashov 29 Jul '10

29 Jul '10

I know failure routes operate relative to the request that failed, rather than the final reply received per se. Nevertheless, is there any way provided by Kamailio/SR at this time to modify a final reply (say, append a header, or something else theoretically permissible) prior to its being passed to the originating UAC? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 4

Re: [SR-Users] urgent :geographic number problem
by Alex Balashov 29 Jul '10

29 Jul '10

These (very poorly formatted!) requests and replies all relate to OPTIONS pings and have nothing to do with calls, failed or otherwise. On 07/29/2010 07:48 AM, hala alramli wrote: > some thing strang ,befor every thing work fine but now the most of the > geografic dont receive calls from outbound (landline) but recive calls > from our system it seem that it deal with it as local number just,when i > snif the backet by using ngrep it give me this: > 2010/07/29 12:58:28.111091 77.93.***.***:5060 -> 173.33.***.***:60005 > OPTIONS sip:173.33.***.***:60005 SIP/2.0..Via: SIP/2.0/UDP > 77.93.***.***:5060 ;branch=0..From: > sip:pinger@nogafone.com;tag=b4a8a831..To: sip:173.33.***.***: > :60005..Call-ID: f8d76aa2-5f768331-3b621@173.33.***.***:..CSeq > <mailto:f8d76aa2-5f768331-3b621@77.93.249.153..CSeq>: 1 > OPTIONS..Content-Length: 0.... > U 2010/07/29 12:58:28.111114 77.93.***.***:5060 -> 173.33.***.***::60004 > OPTIONS sip:173.33.***.***::60004 SIP/2.0..Via: SIP/2.0/UDP > 77.93.249.153:5060;branch=0..From: > sip:pinger@nogafone.com;tag=c4a8a831..To: sip:173.33.***.***: > :60004..Call-ID: f8d76aa2-6f768331-3b621@77.93.***.***:..CSeq > <mailto:f8d76aa2-6f768331-3b621@77.93.249.153..CSeq>: 1 > OPTIONS..Content-Length: 0.... > U 2010/07/29 12:58:28.247854 173.33.***.***:60004 -> 77.93.249.153:5060 > SIP/2.0 404 Not Found..To: > sip:173.33.***.***:60004;tag=c8b17aeb9f7688a1i0..From: > sip:pinger@nogafone.com;tag=c4a8a831..Call-ID: f8d76aa2-6f768331-3b621@ > 77.93.***.***:..CSeq: 1 OPTIONS..Via: SIP/2.0/UDP > 77.93.***.***::5060;branch=0..Server: > Linksys/PAP2T-3.1.15(LS)..Content-Length: 0.... > U 2010/07/29 12:58:28.255958 173.33.***.***:60005 -> 77.93.249.153:5060 > SIP/2.0 404 Not Found..To: > sip:173.33.***.***:60005;tag=c8b17aeb9f7688a1i0..From: > sip:pinger@nogafone.com;tag=b4a8a831..Call-ID: f8d76aa2-5f768331-3b621@ > 77.93.***.***:..CSeq: 1 OPTIONS..Via: SIP/2.0/UDP > 77.93.***.***:5060;branch=0..Server: > Linksys/PAP2T-3.1.15(LS)..Content-Length: 0.... > what is strang is that the replay is 404 not found while > 173.33.***.***:60005 is an IP address for an online adapter i will be > thankfull for any hint > regards > hala > > --- On *Wed, 7/28/10, Alex Balashov /<abalashov(a)evaristesys.com>/* wrote: > > > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] urgent :geographic number problem > To: "hala alramli" <doreme202002(a)yahoo.com> > Cc: "SR-Users" <sr-users(a)lists.sip-router.org> > Date: Wednesday, July 28, 2010, 5:52 PM > > On 07/28/2010 10:48 AM, hala alramli wrote: > > > our geographic number cant receive calls from outbound (from normal > > landlines number) but it receive from our system numbers just > ?where is > > the problem,is it from our server or from the company that provide us > > with this geographic number? > > It is impossible to know the answer to this question without digging > deeper into the underlying cause of the superficial effect you are > describing. The best way is to take a packet capture and see if the > requests from the provider are actually arriving; then you will know > at least whether they are reaching the server, and from there can > see why they may not be getting handled properly at the application > level. > > If the number worked before, and spontaneously stopped working, and > you did not change any configuration, it is most likely the provider. > > -- Alex Balashov - Principal > Evariste Systems LLC > 1170 Peachtree Street > 12th Floor, Suite 1200 > Atlanta, GA 30309 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

Fw: Re: urgent :geographic number problem
by hala alramli 29 Jul '10

29 Jul '10

--- On Thu, 7/29/10, hala alramli <doreme202002(a)yahoo.com> wrote: From: hala alramli <doreme202002(a)yahoo.com> Subject: Re: [SR-Users] urgent :geographic number problem To: "Alex Balashov" <abalashov(a)evaristesys.com> Date: Thursday, July 29, 2010, 2:48 PM some thing strang ,befor every thing work fine but now the most of the geografic dont receive calls from outbound (landline) but recive calls from our system it seem that it deal with it as local number just,when i snif the backet by using ngrep it give me this: 2010/07/29 12:58:28.111091 77.93.***.***:5060 -> 173.33.***.***:60005 OPTIONS sip:173.33.***.***:60005 SIP/2.0..Via: SIP/2.0/UDP 77.93.***.***:5060 ;branch=0..From: sip:pinger@nogafone.com;tag=b4a8a831..To: sip:173.33.***.***: :60005..Call-ID: f8d76aa2-5f768331-3b621@173.33.***.***:..CSeq: 1 OPTIONS..Content-Length: 0.... U 2010/07/29 12:58:28.111114 77.93.***.***:5060 -> 173.33.***.***::60004 OPTIONS sip:173.33.***.***::60004 SIP/2.0..Via: SIP/2.0/UDP 77.93.249.153:5060;branch=0..From: sip:pinger@nogafone.com;tag=c4a8a831..To: sip:173.33.***.***: :60004..Call-ID: f8d76aa2-6f768331-3b621@77.93.***.***:..CSeq: 1 OPTIONS..Content-Length: 0.... U 2010/07/29 12:58:28.247854 173.33.***.***:60004 -> 77.93.249.153:5060 SIP/2.0 404 Not Found..To: sip:173.33.***.***:60004;tag=c8b17aeb9f7688a1i0..From: sip:pinger@nogafone.com;tag=c4a8a831..Call-ID: f8d76aa2-6f768331-3b621@ 77.93.***.***:..CSeq: 1 OPTIONS..Via: SIP/2.0/UDP 77.93.***.***::5060;branch=0..Server: Linksys/PAP2T-3.1.15(LS)..Content-Length: 0.... U 2010/07/29 12:58:28.255958 173.33.***.***:60005 -> 77.93.249.153:5060 SIP/2.0 404 Not Found..To: sip:173.33.***.***:60005;tag=c8b17aeb9f7688a1i0..From: sip:pinger@nogafone.com;tag=b4a8a831..Call-ID: f8d76aa2-5f768331-3b621@ 77.93.***.***:..CSeq: 1 OPTIONS..Via: SIP/2.0/UDP 77.93.***.***:5060;branch=0..Server: Linksys/PAP2T-3.1.15(LS)..Content-Length: 0.... what is strang is that the replay is 404 not found while 173.33.***.***:60005 is an IP address for an online adapter i will be thankfull for any hint regards hala --- On Wed, 7/28/10, Alex Balashov <abalashov(a)evaristesys.com> wrote: From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] urgent :geographic number problem To: "hala alramli" <doreme202002(a)yahoo.com> Cc: "SR-Users" <sr-users(a)lists.sip-router.org> Date: Wednesday, July 28, 2010, 5:52 PM On 07/28/2010 10:48 AM, hala alramli wrote: > our geographic number cant receive calls from outbound (from normal > landlines number) but it receive from our system numbers just ?where is > the problem,is it from our server or from the company that provide us > with this geographic number? It is impossible to know the answer to this question without digging deeper into the underlying cause of the superficial effect you are describing. The best way is to take a packet capture and see if the requests from the provider are actually arriving; then you will know at least whether they are reaching the server, and from there can see why they may not be getting handled properly at the application level. If the number worked before, and spontaneously stopped working, and you did not change any configuration, it is most likely the provider. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

userblacklist Module - Documentation fault?
by "Nicolas Rüger" 29 Jul '10

29 Jul '10

Hello, in the documentation of the Module "Userblacklist" seems not really clear to me. Please clarify my doubts. Refering to: http://www.kamailio.org/docs/modules/3.0.x/modules_k/userblacklist.html in "4.1. check_user_blacklist" is written: "[...]If a match is found and it is not set to whitelist, false is returned. Otherwise, true is returned.[...]" Question: 1.) Is "not whitelisted" equal to "blacklisted"? 2.) Why is "false" returned if the blacklist_check is positive? 3.) Is "true" even returned if there's no match? I'd have expected the following: if (user_is_blacklisted) then return TRUE; else return FALSE; Can you please clarify the behavior? Thank you. Regards, Nicolas Example that seems clear to me: =============================== In "4.2. check_user_whitelist" is written: "[...]If a match is found and it is set to whitelist, true is returned. Otherwise, false is returned.[...]" cause it means to me: if (user_is_whitelisted) then return TRUE; else return FALSE; -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

2 2

Meeting in Chicago, Cluecon 2010, next week
by Daniel-Constantin Mierla 29 Jul '10

29 Jul '10

Hello, I will give a presentation about latest Kamailio - SIP Router news and usage together with FreeSwitch Media Server at Cluecon 2010 (http://www.cluecon.com), next week in Chicago. If you are around and want to meet during the event or out of it, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

Kamailio Transformation Assignment to an AVP
by Brandon Armstead 29 Jul '10

29 Jul '10

Hello, I have the following transformation on the SDP Body: $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s}) However when I assign this to an AVP, i.e. $avp(s:sdp-payloads) = $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s}); I receive back a NULL result / transformation "regex does not match" is what I receive with high verbose syslog on kamailio. HOWEVER. When I do something like this: xlog("L_INFO", "Payloads Available: $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s})"); It executes properly -- transformation regex matches and the payloads are displayed as expected. If anyone can provide any insight as to what I may be doing wrong it would be greatly appreciated. P.S. Kamailio SVN Revision # 2:5906M (1.5.2-notls). Also I CC'ed devel list (as I do believe this may be a bug). Thanks! Sincerely, Brandon Armstead

3 5

Re: [SR-Users] how to set up iptel.org sip outbound
by Daniel-Constantin Mierla 29 Jul '10

29 Jul '10

Hello, please keep the mailing list cc-ed, private emails related to mailing lists discussions will not be answered. On 7/29/10 1:46 AM, j w wrote: > Hi Daniel, > > Thanks for your reply. Regarding set up outbound from ATA. I have > check with the Linksys tech support. They checked and said the > setting that they can do is all done, I have check with the outbound > provider. > > Can your outbound service allow me to dial out with a landline phone > via ATA. Would you please give me some setting information for your > outbound. Then it is not my service, I pointed you to a mailing list that is more appropriate for this subject. What I can give you as hints, try to put the ata in a network hub and sniff the sip traffic (with ngrep, wireshark, tcdump, etc...) to see if there is some attempt to send registrations/calls to sip server. Also, check maybe the device has some troubleshooting tool embedded (i.e., dedicated web page for logs). Cheers, Daniel > > Much obliged. > > Joseph > > ------------------------------------------------------------------------ > Date: Wed, 28 Jul 2010 10:54:24 +0200 > From: miconda(a)gmail.com > To: w7269826(a)hotmail.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] how to set up iptel.org sip outbound > > Hello, > > On 7/19/10 8:35 PM, j w wrote: > > Hi, > > I registered iptel.org for an account and set up my Linksys > SPA3102 ATA gateway. The inbound is working but outbound cannot > dial out. > > First, the mailing list for iptel.org service is: > http://lists.iptel.org/mailman/listinfo/services > > You should get better answers there. > > Can any one help me: > 1. how to set up the sip outbound in order for me to dial out via ATA. > > > This has to do with ATA configuration, not much related to the server > side. You should have a user manual for that device. > > 2. how to get Canada phone number. > > > I don't think iptel.org is offering this, if you need for incoming, > you probably can get it from somewhere else and forward it to you > iptel.org accoung. > > Thanks a lot. > > I am not sure how to retrieve the post/mailing list, would you > please forward me your help reply to my > e-mail:w7269826@hotmail.com <mailto:e-mail:w7269826@hotmail.com>. > > > This is a public mailing list based on subscriptions. You have to > subscribe via web at: > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > Enter your email address, you will get a confirmation email with > further instructions. Once you follow them, you will receive all > emails sent to mailing list. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

urgent :geographic number problem
by hala alramli 28 Jul '10

28 Jul '10

our geographic number cant receive calls from outbound (from normal landlines number) but it receive from our system numbers just ?where is the problem,is it from our server or from the company that provide us with this geographic number? please help me regards

2 1

urgent :geographic number problem
by hala alramli 28 Jul '10

28 Jul '10

our geographic number cant receive calls from outbound (from normal landlines number) but it receive from our system numbers just ?where is the problem,is it from our server or from the company that provide us with this geographic number? please help me regards

1 0

multiple instances on different ports
by Robert R 28 Jul '10

28 Jul '10

Hi, I am trying to run multiple different instances of openser on the same server using different ports. Regardless of receiving INVITE on any of the ports (5061, 5062, ...) I want openser always to proxy the message to port 5060. invite:5061 ---> openser --> invite:5060 invite:5062 ---> openser --> invite:5060 By adding two listen ports as follow, openser sends the traffic to the same outgoing port as the incoming port: listen=udp:209.16.123.160:5061 listen=udp:209.16.123.160:5062 Is there a way to hard code the outgoing port? or I should overwrite the port? Thanks, R

4 5

Missing fields in ACC
by David 28 Jul '10

28 Jul '10

Hello, I am doing db logging, and my ACCs are being added to the table, but it is missing the leg_src and leg_dst columns. I set the leg_src and leg_dst columns, I do this : modparam("acc", "multi_leg_info", "leg_src=$avp(s:caller);leg_dst=$avp(s:dialed)") In my routing script on the initial INVITE I do this : $avp(s:dialed) = $tu; $avp(s:caller) = $fu; xlog("L_WARNING", "Call Tracked - leg_dst=$avp(s:dialed) leg_src=$avp(s:caller)\n"); The xlog shows the leg_src and leg_dst, but they do not appear in the leg_dst and leg_src columns of my acc table. I have no warnings or errors ( other than "ERROR: ctl: could not delete unix socket /tmp/ser_ctl: Operation not permitted" ), but that's not worrysome. I do not use t_newtran(), t_check_trans() is called before I set the values, but this does not create a transaction. After the above code is called, I call t_relay(); Any idea what might cause the ACC record to be added without the multi_leg_info ? BTW, the avpops table in the database is a ENGINE=BLACKHOLE. As far as I can tell, the ACC is the same as my Kamailio 1.5.3 setup, but it would seem in migrating to 3.0, something broke, any ideas ? Thanks, David

2 1

no signal in phone line ,server slow
by hala alramly 28 Jul '10

28 Jul '10

hello, we have a problem with our kamailio server sometimes it being very slow ,and i cant hear the signal in the phone line for a while then it fix him self ,when i return to syslog every thing normal, what is the reasons ? regards, hala _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969

2 1

how to set up iptel.org sip outbound
by j w 28 Jul '10

28 Jul '10

Hi, I registered iptel.org for an account and set up my Linksys SPA3102 ATA gateway. The inbound is working but outbound cannot dial out. Can any one help me: 1. how to set up the sip outbound in order for me to dial out via ATA. 2. how to get Canada phone number. Thanks a lot. I am not sure how to retrieve the post/mailing list, would you please forward me your help reply to my e-mail:w7269826@hotmail.com. Much obliged. Joseph _________________________________________________________________ MSN Dating: Find someone special. Start now. http://go.microsoft.com/?linkid=9734384

2 1

[Serusers] Error while running icscf.sh
by Abdul Khadar 28 Jul '10

28 Jul '10

Hi, Can you please help me in resolving this issue while setting OpenIMSCore setup. Error : [root@hss OpenIMSCore]# sh icscf.sh * 0(18935) ERROR: load_module: could not open module </opt/OpenIMSCore/ser_ims/modules/mysql/mysql.so>: /opt/OpenIMSCore/ser_ims/modules/mysql/mysql.so: undefined symbol: log * 0(18935) parse error (30,13-14): failed to load module ERROR: bad config file (1 errors) 0(18935) INFO:cdp:cdp_exit(): CDiameterPeer stoping ... icscf.sh: line 5: 18935 Segmentation fault /opt/OpenIMSCore/ser_ims/ser -f /opt/OpenIMSCore/icscf.cfg -D -D please help me to resolve this problem. -- thanks & regards, Abdul Khadar

4 4

Kamailio 3.0.2 mysql create table domain definition
by Stagg Shelton 28 Jul '10

28 Jul '10

Hello, The script in the source scripts/mysql/my_create.sql uses the the following definition CREATE TABLE domain ( did VARCHAR(64) NOT NULL, domain VARCHAR(128) NOT NULL, flags INT UNSIGNED NOT NULL DEFAULT '0', UNIQUE KEY domain_idx (domain), KEY did_idx (did) ); The script in the source utils/kamctl/mysql/domain-create.sql uses the following definition CREATE TABLE domain ( id INT(10) UNSIGNED AUTO_INCREMENT PRIMARY KEY NOT NULL, domain VARCHAR(64) DEFAULT '' NOT NULL, last_modified DATETIME DEFAULT '1900-01-01 00:00:01' NOT NULL, CONSTRAINT domain_idx UNIQUE (domain) ) ENGINE=MyISAM; Which table definition is the correct definition to be used. Thanks Stagg Shelton

3 2

Doubs about using append_branch ...
by Raúl Alexis Betancor Santana 26 Jul '10

26 Jul '10

Hi all I'm trying to do a "followme" from the proxy side, so I receive a request to some SIP URI and that URI have configured it's followme function that in this case means ... progress a call to the original URI + a call to a new URI, just a parallel branching, but I'm no luck ... :-( if(is_avp_set($avp(s:followme)) { append_branch($avp(s:followme)); } That doesn't run, because append_branch does not accept a PV as param, so ... ... $var(s:tmp_ru) = $ru; append_branch(); $ru = $avp(s:followme); ... But now ... only one of the branches progress ..., the new created branch (after the append_branch call) ... must go to a PSTN number ... but no way, because I must "mark" the branch in some way that I could send it directly to the go_to_pstn route block ... any clue or example about how to work with the new branch ONLY ? ... because branch 1 works ok, I need to do changes only on branch 2 Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual

3 16

Re: [SR-Users] Doubs about using append_branch ...
by Alex Balashov 25 Jul '10

25 Jul '10

It sounds like what you need is to do branch-specific logic in a branch route: http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#branch_route Not all functions can be used in a branch route, but those that can be operate only on that branch. On 07/25/2010 07:07 PM, Raúl Alexis Betancor Santana wrote: > > Hi all > > I'm trying to do a "followme" from the proxy side, so I receive a request to > some SIP URI and that URI have configured it's followme function that in this > case means ... progress a call to the original URI + a call to a new URI, > just a parallel branching, but I'm no luck ... :-( > > if(is_avp_set($avp(s:followme)) > { > append_branch($avp(s:followme)); > } > > That doesn't run, because append_branch does not accept a PV as param, so ... > > ... > $var(s:tmp_ru) = $ru; > append_branch(); > $ru = $avp(s:followme); > ... > > But now ... only one of the branches progress ..., the new created branch > (after the append_branch call) ... must go to a PSTN number ... but no way, > because I must "mark" the branch in some way that I could send it directly to > the go_to_pstn route block ... > > any clue or example about how to work with the new branch ONLY ? ... because > branch 1 works ok, I need to do changes only on branch 2 > > Best regards -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

Kamailio 3.0 cann't access the RTPProxy
by JinKevin 24 Jul '10

24 Jul '10

Hi All, I have the kamailio configed as below on Solaris 10 (Sparc): # ----- nathelper ----- #!ifdef WITH_NAT modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") 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", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif and rtpproxy is started as AppSer01:ipunity@/opt/ipunity$ ps -ef |grep rtp ipunity 4687 1 0 12:58:20 ? 0:00 /usr/local/bin/rtpproxy -l 210.xx.xx.xx -s udp:localhost:7722 AppSer01:ipunity@/opt/ipunity$ netstat -an |grep 7722 127.0.0.1.7722 Idle 127.0.0.1.41509 127.0.0.1.7722 Connected 127.0.0.1.41510 127.0.0.1.7722 Connected 127.0.0.1.41511 127.0.0.1.7722 Connected 127.0.0.1.41512 127.0.0.1.7722 Connected 127.0.0.1.41513 127.0.0.1.7722 Connected 127.0.0.1.41514 127.0.0.1.7722 Connected 127.0.0.1.41515 127.0.0.1.7722 Connected 127.0.0.1.41516 127.0.0.1.7722 Connected 127.0.0.1.41517 127.0.0.1.7722 Connected 127.0.0.1.41518 127.0.0.1.7722 Connected 127.0.0.1.41519 127.0.0.1.7722 Connected However, kamailio is not able to connect the rtpporxy, Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 788202 local0.error] ERROR: nathelper [nathelper.c:2457]: can't send command to a RTP proxy Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 844113 local0.error] ERROR: nathelper [nathelper.c:2492]: proxy <udp:127.0.0.1:7722> does not respond, disable it Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:3144]: no available proxies Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 788202 local0.error] ERROR: nathelper [nathelper.c:2457]: can't send command to a RTP proxy Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 844113 local0.error] ERROR: nathelper [nathelper.c:2492]: proxy <udp:127.0.0.1:7722> does not respond, disable it Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:3144]: no available proxies Jul 24 12:59:04 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22870]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:2627]: no available proxies Jul 24 12:59:04 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22870]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:2627]: no available proxies Any helps are appreciated. Thanks, Kevin _________________________________________________________________ SkyDrive电子画册，带你领略精彩照片，分享“美”时“美”刻! http://www.windowslive.cn/campaigns/e-magazine/ngmchina/?a=c

1 0

Regex Transformations
by Brandon Armstead 22 Jul '10

22 Jul '10

Hello, I am trying to match a multi-line psuedo variable, i.e. $rb However I am wishing to pull out the payload values i.e. "0 18 101" "18 101" etc.. etc... I am having trouble matching this. Any help would be appreciated. One part that is giving me trouble is that it seems xlog prints out \r\n as \015\012 I am not able to successfully just pull out the "payloads" I have tried many different variations, however here is one of my latest: xlog("L_INFO", "[$ci] $(rb{re.subst,/^(.*)m=audio ([0-9]+) ([0-9 ]+)\015\012(.*)$/\3/s})"); Any help / input is greatly appreciated, thank you ahead of time! Happy 4th of July (for those who celebrate) Sincerely, Brandon Armstead

2 9

Sniffer SIP
by Fernando Pessoa 22 Jul '10

22 Jul '10

Friend you know tell me if it is possible to make the packets sniffer sip with SER or SEMS? thanks, -- Atenciosamente, Fernando Pessoa ITIntegration Diretor 85-34865449 85-99222598

3 2

Outbound calls
by Alan 22 Jul '10

22 Jul '10

I'm trying to set up my account so outbound calls will go through a proxy: free.sipout.com:161 I've set it up under OTHER in my iptel.org account so I access pbxes.com and put free.sipout.com:161 into a trunk and set it as an outbound route. I then dial +001 area code and number but it doesn't work. What am I doing wrong? Thanks, Alan

2 1

Perl Code Execution via kamailio.cfg fails
by "Nicolas Rüger" 22 Jul '10

22 Jul '10

Hello, I try to execute some PERL-code within kamailio v3 for some research due to my thesis. I'm using debian lenny, kamailio_3.0.0 and perl_5.10.0. I added the follwoing lines to "kamailio.cfg": loadmodule "perl.so" modparam("perl", "filename", "/opt/kamailio-3.0.0/PERL/myperl.pl") modparam("perl", "modpath", "/opt/kamailio-3.0.0/lib/kamailio/perl") I added a subroutine called "myprint" to the file "myperl.pl" for testing. I tried to exec the subroutine by adding the following line to "kamailio.cfg" just before "Routing Logic" (...even tried within "route{..}") as described in the perl_module description : perl_exec_simple("myperl"); Unfortunately it seems the code isn't executed, as it should write some lines in a log-file. What can I do? Thank you!! Cheers Nicolas additional information: ======================= Kamailio is starting without problems it just shows the follwing warning: "0(28679) WARNING: <core> [sr_module.c:445]: /opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will notbe supported in newer versions; consider using mod_register() instead 0(28679) DEBUG: <core> [mem/f_malloc.c:265]: fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048" -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

3 13

Re: [SR-Users] PDT ran out of pkg memory using pdt_list [SOLVED}
by JR Richardson 21 Jul '10

21 Jul '10

>> try to play with mysql module timeout parameter in case the records are not >> loaded fast enough: >> >> http://kamailio.org/docs/modules/stable/modules/db_mysql.html#id2928400 >> >> Regards, >> Ramona > > Ok, I figured out that sr was taking about 9-10 seconds to load 300K > records so I increased: > # ----- db_mysql params ----- > modparam("db_mysql", "timeout_interval", 20) > > And found that the table would load, so I started increasing the > record count by 60K, then the fault condition occurred again at around > 400K records, then went away by itself. I got all the way to 760K > records without a problem then hit another wall at 800K records. > > I think there is something flaky going on, I can not reproduce it > reliable, sometimes the table loads just fine and other times it does > not load at all. > > I started monitoring mysqld. When I restart sr or issue pdt_reload > and msqld has a load, the table seems to load fine, regardless of the > table size or "timeout_interval" setting. When the problem is > occurring, myslqd has zero usage. When the issue was occurring, I > restarted the mysql process and the table loaded into sr. I'm now > suspecting I have a buggy mysqld so I'm going to try to update that > and see if I can reproduce the problem. So for now I can't isolate > the issue, but I'm still working on it. > > I do see a difference with the "timeout_interval" during sr restarts, > the longer the time the better it does. Also there is a time > difference when loading the table, sometimes it takes 30 seconds, > sometimes 90 seconds. I think maybe there is a combination of issues > going on here. I need to rebuild a fresh install and test a bit more > methodically to find a root cause. Also I want to grab a newer > version of mysql-server. > > I'll update with my findings. # ----- db_mysql params ----- modparam("db_mysql", "timeout_interval", 20) Time interval (in seconds) after that an connection attempt, read or write request is aborted. The value counts three times, as several retries are done from the driver before it gives up. "The values counts 3 times" this was my issue, as I was adjusting the value between 10, 20, 30, 60, 120, and having inconsistent table load times from mysqld. After a reboot, mysql started performing better, more consistent with load times. My table with 800K records was taking between 30-38 seconds to load. With timeout_interval set to 10 (30 second timeout), I was right on the edge of the table loading or not loading, sometimes it would and other times it would not. So giving a bit of cushion on the timeout_interval seems to resolve the problem. Adjusting this value I could consistently reproduce the problem. So for large database tables, this timeout value should be set to about 1 second per 20K records. As far as the other issue goes with using pdt_list and pkg memory size, compileing mi using system mem solved this issue as well. Also pdt_list has the option to search for prefixes which works in my case without the need to increase pkg memory or system memory: sip-router2:~# kamctl fifo pdt_list . 99999 PDT:: SDOMAIN=* PREFIX=999990 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999991 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999992 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999993 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999994 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999995 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999996 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999997 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999998 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999999 DOMAIN=2 Using the list prefix search option will pipe out about 10,000 records, but errors out with "no more pkg mem" past that. Thanks for all your input on this thread to help me isolate the issue. JR -- JR Richardson Engineering for the Masses

1 0

freezing for major release 3.1
by Daniel-Constantin Mierla 21 Jul '10

21 Jul '10

Hello, as discussed during our devel meeting, next major release 3.1 is scheduled for this autumn. Therefore the freezing should happen soon. Would be 2 weeks from now enough for everyone to push planned new features to repo? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

4 9

Dialplan module examples
by Raúl Alexis Betancor Santana 20 Jul '10

20 Jul '10

Anyone known about examples for the dialplan table ? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual

2 2

kamailio server load balancing
by RAJNIKANT VANZA 20 Jul '10

20 Jul '10

Hi Guys, I have 2 kamailio servers. I need to load balancing for this 2 kamailio servers. Scenario: ------------ Each kamailio server call limit is 5000 approx. kamailio server 1 kamailio server 2 If kamailio server 1 reach call limit (5000) then call go on kamailio server 2. What is the best way for achieve this? Thanks in advance. -- Best Regards, Rajnikant Vanza

2 1

our kamailio server very slow , and there is no signal in the line
by hala alramli 17 Jul '10

17 Jul '10

Hello, some time ,there is no signal in phone line for a while ,also the server become very slow then it fix him self ,i don't know how . ,could any body give me possible reasons ,and how i can fix that, i checked syslog file every thing normal as i can see ,please help me. regards, hala

2 1

Simple Least Cost Routing NPA-NXX?
by JR Richardson 16 Jul '10

16 Jul '10

Hi All, I'm in the lab with sip router 3.0 mocking up some trunk-group prefix routing with PDT and load balancing with Dispatcher. This is working great and as expected. I would like to incorporate some sort of LCR matching on NPA-NXX but I'm having some difficulty. The PDT module automatically strips the prefix so for trunk group routing it works great but really can't be used for matching NPA-NXX. The Carrierroute module is pretty nice and works as expected but the issue I'm having, it appears that I have to have a NPA-NXX entry for each domain gateway I want to send calls to. For instance if I have 3 gateways in domain 1, then I would need the same NPA-NXX record entry in the database for each gateway. The database entries would be [NPA-NXX] x [Number of Gateways] per domain. So having 100K routes would mean having 300K records. Am I looking at this correctly? This is really what it seems like to me but logically I would think this should not be the case? I really like the Dispatcher polling functions to actively know what gateways are on-line and off-line plus the different distribution algorithms to choose from. So it makes sense to me to implement a pseudo PDT/Distpacher scenario as an LCR but with PDT stripping the prefix, that can not be done. Is there another mechanism whereby I can dip the database looking for the NPA-NXX, extract the domain group and send that to the Dispatcher function, without stripping the prefix? Or can a PDT module parameter be added to not strip the prefix. But I'm not sure this will work, can the PDT module hold in memory a few hundred thousand records? Thanks. JR -- JR Richardson Engineering for the Masses

2 3

Load Balancing REDIRECT
by Maw Zin Tun - Real Telekom 16 Jul '10

16 Jul '10

Hi, I started using Kamailio 3.0 as redirect server about 2 months ago, basically I'm quite new to it. Now I need to change current setup to redundant active-active system but I'm struggling where to find documentation that can lead me to achiving it. Can anyone pls advise me where to start looking ? Many thanks, Maw Tun

4 5

No error line printed ?
by Raúl Alexis Betancor Santana 15 Jul '10

15 Jul '10

Hi, I'm trying to test the new 3.0.2 release ... but with no luck ... sbc02:/etc/kamailio# kamailio -ddddddddd -u kamailio -g kamailio -c -E 0(1036) DEBUG: <core> [main.c:1800]: read 343643572 from /dev/urandom 0(1036) DEBUG: <core> [main.c:1806]: seeding PRNG with 1622837431 0(1036) DEBUG: <core> [main.c:1810]: test random numbers 396346077 2140170049 1800197589 0(1036) : <core> [cfg.lex:1390]: included file name must be between quotes If i'm not wrong ... I should get the line/caracter at witch the parser get the error ... but that's not true ... I'm missing something really stupid ? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual

2 12

Dispatcher: ds_select_domain(set, alg) with a pseudo variable.
by Geoffrey Mina 15 Jul '10

15 Jul '10

How difficult would it be to make ds_select_domain(set, alg) accept a pseudo variable for the "set" argument? I would like to map the $rU to a specific dispatcher set via a database table. I suppose i could just hard code three or four sets with a conditional if() statement, but I would prefer to make it a little more dyanimc. Correct me if I am wrong in my assumption that ds_select_domain() doesn't accept pseudo variables. Thanks.

3 4

CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7f1106077a60
by Iñaki Baz Castillo 15 Jul '10

15 Jul '10

Hi, what does it mean this error message?: CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7f1106077a60 Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

Fwd: Re: Fwd: Re: Kamailio and NAPTR lookup with TLS
by Matteo Campana 14 Jul '10

14 Jul '10

Hi Daniel, I have followed your instruction (I'm using kamailio 1.5) and before the t_relay("0x05") statement I've setted: $du="sip:" + $rd + ";transport=tls"; but I get the following mesasge: Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790] Request leaving server, *D_-URI='sip:garr.it;transport=tls'_* - M=INVITE RURI=sip:xxx@garr.it F=sip:yyyyy@sip.mydomqin.it:5066 T=sip:123456789@sip.mydomain.it:5066 Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790]: CRITICAL:core:mk_proxy: could not resolve hostname: "garr.it" Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790]: ERROR:tm:uri2proxy: bad host name in URI <sip:garr.it;transport=tls> Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790]: ERROR:tm:t_forward_nonack: failure to add branches Therefore I would like to use information about TLS connection (tls port?) got from NAPTR record. Doesn't it possible with kamailio 1.5 ? Many thanks, Daniel Il 08/07/2010 18.12, Matteo Campana ha scritto: > > > -------- Messaggio originale -------- > Oggetto: Re: [SR-Users] Kamailio and NAPTR lookup with TLS > Data: Thu, 08 Jul 2010 18:10:35 +0200 > Mittente: Daniel-Constantin Mierla <miconda(a)gmail.com> > A: matteo.campana(a)klarya.it > CC: sr-users(a)lists.sip-router.org > > > > Hello, > > On 7/8/10 5:59 PM, Matteo Campana wrote: > > > > Hi all, > > I'm using kamailio 1.5 with TLS module. > > I need to make ENUM query and get NAPTR record. > > > From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol. > > > > How can I tell Kamailio to use TLS protocol ( instead of udp) after NAPTR lookup ? > > > > I've try to set : > > > > dns_tls_pref=1 > > dns_udp_pref=2 > > dns_tcp_pref=3 > > > > in the general section of kamailio.cfg, but I get a parse error. > > > > > these parameters were introduced in kamailio with version 3.0. > > If you need TLS then it is recommended to use 3.0 anyhow, it is a far > better implementation. That will make the life easier to migrate to > upcoming 3.1 that will bring asynchronous TLS. > > No matter what you have in R-URI, you can force TLS via setting outbound > proxy address to be a TLS uri: > > $du = "sip:__ip_or_host__;transport=tls"; > t_relay(); > > The IP or host you can take from R-URI without any problem via PV $rd. > Other option is to use function from tm - t_relay_to_tls(): > > http://kamailio.org/docs/modules/stable/modules/tm.html#t_relay_to_udp > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > >

2 1

dialog module: in-early-dialog requests / race conditions
by Timo Reimann 13 Jul '10

13 Jul '10

Hello, I have two issues related to dialog module state keeping/logging that caught my attention today: 1.) The module considers requests routed in early (but not yet confirmed) dialogs to be bogus, as can be seen from a few lines of state machine code in dlg_hash.c: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=modules_… In the logs, it shows up something like this: CRITICAL:dialog:log_next_state_dlg: bogus event 8 in state 2 for dlg 0x7f9f237ffff8 [2901:1286709577] with clid '<Call-ID>' and tags '<caller tag>' '<callee tag, always empty>' I believe that such in-early-dialog requests should not trigger bogus-event log messages as they look fine from a standard's point of view. 2.) Various race conditions may generate log messages such as following when DID mode is enabled: WARNING:dialog:dlg_onroute: unable to find dialog for <REQUEST TYPE> with route param 'faf.f69e7b4' [4015:79161711] Observed REQUEST TYPEs at our site were "BYE" when BYE requests were transmitted by both caller and callee at roughly the same time due to user behavior; and "PRACK" when a PRACK was received by the UAS after a 486 (Busy Here)/ACK pair of messages was already exchanged between the UAS and the proxy. In both cases, the associated dialog had already been destroyed when another request was received. My suggestion is to lower the log level for this kind of message because it can simply happen too often but doesn't affect proper dialog handling. Feedback? Cheers, --Timo

3 8

Several BYEs conveying Kamailio
by Maciej Bylica 11 Jul '10

11 Jul '10

Dear ALL, I have configured Kamailio 1.5 from svn with radius (radiusclient-ng) support for acc. All seem to work okay, except one thing. >From time to time I am receiving following radius error (debug from radius server on different ip) Fri Jul 9 12:22:58 2010 : Error: rlm_sql (sql): Couldn't update SQL accounting STOP record - Column 'AcctStartTime' cannot be null Fri Jul 9 12:23:08 2010 : Error: rlm_sql (sql): Couldn't update SQL accounting STOP record - Column 'AcctStartTime' cannot be null Fri Jul 9 12:23:18 2010 : Error: rlm_sql (sql): Couldn't update SQL accounting STOP record - Column 'AcctStartTime' cannot be null A few lines from radiusclient.conf # time to wait for a reply from the RADIUS server radius_timeout 10 # resend request this many times before trying the next server radius_retries 3 After closer look at this, it turned out that there is one START request and four STOPs with the same CallID (Acct-Session-Id). START and first STOP are treated as a pair, the next ones are odd. Taking about sip i do have following scenario: MGW(1)---Kamailio(2)---SIP_Provider(SER based)(3) The call flow that generate aforemtioned errors looks like below (debug on kamailio) 1 - (2) is receiving INVITE from (3) 2 - (2) is sending Giving a Try to (3) 3 - (2) is sending INVITE to (1) 4 - (2) is receiving Trying from (1) 5 - (2) is receiving Ringing from (1) 6 - (2) is receiving OK from (1) 7 - (2) is sending OK to (3) 8 - (2) is receiving ACK from (3) 9 - (2) is sending ACK to (1) ---- call---- 10 - (2) is receiving BYE from (1) (tearing down from (1) side, STOP radius request is generated) 11 - (2) is sending BYE to (3) 12 - (2) is receiving BYE from (3) (second STOP is generated, radius_retries 3, radius_timeout 10) 13 - (2) is sending BYE to (1) 14 - (2) is receiving OK from (1) 15 - (2) is sending OK to (3) 16 - (2) is receiving OK to (3) 17 - (2) is sending OK to (1) For sure there is sth wrong with SIP_Provider but how Kamilio should behave is such situation. Should second BYE (p.12) with the same callid as in p.10 be continued with p.13? Thanks in advance, Maciej.

2 3

Use AVP as regex in avp_subst
by Uriel Rozenbaum 10 Jul '10

10 Jul '10

Hey guys, I'm using Kamailio 1.5.3-notls and need to apply some re operation to an AVP. The problem I have is that my re expression is held in an AVP, so the parser is not recognizing it :( Would it be very hard to change that behavior? Any ideas/tips to implement? This code route[16] { ... avp_subst("$avp(s:ANI)/$avp(s:ANIegress)", "$avp(s:carrierAniRegex)"); ... } returns this error ERROR:avpops:fixup_subst: avpops: bad subst re $avp(s:carrierAniRegex) This code route[16] { ... avp_subst("$avp(s:ANI)/$avp(s:ANIegress)", $avp(s:carrierAniRegex)); ... } returns this error CRITICAL:core:yyerror: parse error in config file, line 751, column 51-74: syntax error CRITICAL:core:yyerror: parse error in config file, line 751, column 74-75: bad arguments Do you think some other way I can achieve this? Be aware I'm using SQLops to get realtime the value of the expression... Thanks in advance! Uriel

3 6

Kamailio, spiral with TM
by David 09 Jul '10

09 Jul '10

Hey, I have setup two Kamailio servers, my INVITEs will go through server A, which forwards to server B which forwards it back to server A. This is intential, the trouble is when the packet comes back through server B, nothing has changed so TM sees it as a retransmission. The difference between the first time through and the second time is the port 5061 instead of 5060. How can I tell tm that these are not the same transactions ? Would it be easier to have a seperate instance of Kamailio for 5061 ? Thanks, David

4 7

How to force snd_port?
by Victor Pascual Avila 09 Jul '10

09 Jul '10

Hi Folks, How could I force my sip-router to send from a given port number? Many thanks in advance, -- Victor Pascual Ávila

4 8

Kamailio and NAPTR lookup with TLS
by Matteo Campana 09 Jul '10

09 Jul '10

Hi all, I'm using kamailio 1.5 with TLS module. I need to make ENUM query and get NAPTR record. >From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol. How can I tell Kamailio to use TLS protocol ( instead of udp) after NAPTR lookup ? I've try to set : dns_tls_pref=1 dns_udp_pref=2 dns_tcp_pref=3 in the general section of kamailio.cfg, but I get a parse error. Regards, Daniel

4 4

Fwd: Re: Fwd: Re: Kamailio and NAPTR lookup with TLS
by Matteo Campana 09 Jul '10

09 Jul '10

Hi klaus, Suppose I can't access to NAPTR settings. I need to manage SIP URI, so , If I right understand, the only way to use TLS protocol in kamailio 1.5 is to append ";transport=tls" in R-URI before relay. In other words I need to rewrite R-URI: $ru = $ru + ";transport=tls" ; # and the t_relay t_relay() ; something like that? Regards, Daniel Il 08/07/2010 18.45, Matteo Campana ha scritto: > > > -------- Messaggio originale -------- > Oggetto: Re: [SR-Users] Kamailio and NAPTR lookup with TLS > Data: Thu, 08 Jul 2010 18:44:27 +0200 > Mittente: Klaus Darilion <klaus.mailinglists(a)pernau.at> > A: Daniel-Constantin Mierla <miconda(a)gmail.com> > CC: matteo.campana(a)klarya.it, sr-users(a)lists.sip-router.org > > > > Am 08.07.2010 18:10, schrieb Daniel-Constantin Mierla: > > Hello, > > > > On 7/8/10 5:59 PM, Matteo Campana wrote: > >> > >> Hi all, > >> I'm using kamailio 1.5 with TLS module. > >> I need to make ENUM query and get NAPTR record. > >> > From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol. > >> > >> How can I tell Kamailio to use TLS protocol ( instead of udp) after > >> NAPTR lookup ? > >> > >> I've try to set : > >> > >> dns_tls_pref=1 > >> dns_udp_pref=2 > >> dns_tcp_pref=3 > >> > >> in the general section of kamailio.cfg, but I get a parse error. > >> > > these parameters were introduced in kamailio with version 3.0. > > > > If you need TLS then it is recommended to use 3.0 anyhow, it is a far > > better implementation. That will make the life easier to migrate to > > upcoming 3.1 that will bring asynchronous TLS. > > > > No matter what you have in R-URI, you can force TLS via setting outbound > > proxy address to be a TLS uri: > > > > $du = "sip:__ip_or_host__;transport=tls"; > > t_relay(); > > IIRC we do have NAPTR support in Kamailio 1.5 - don't we? > > Then I think it should work when putting a domain into $du and makeing > sure that there is no transport parameter, no port, and NAPTR TLS record > has highest priority. > > regards > klaus > > > > > The IP or host you can take from R-URI without any problem via PV $rd. > > Other option is to use function from tm - t_relay_to_tls(): > > > > http://kamailio.org/docs/modules/stable/modules/tm.html#t_relay_to_udp > > > > Cheers, > > Daniel > > > >

2 1

Differentiation between ACK INVITE and ACK CANCEL
by inge 08 Jul '10

08 Jul '10

Hello all, I need to work on particular ACK within the configuration of SER 0.9.9. Firstly, I was thinking to make a difference using the CSeq field, but it seems that the CSeq on ACK contains the ACK himself. Probably because ACK is a request and not a response. Would it be possible to proceed in another way? Regards, Adrien .L

4 9

Kamailio integration - Registration issue
by Lucas Alvarez 08 Jul '10

08 Jul '10

Hi, I new with kamailio, I've been able to integrate kamailio 3.02 with asterisk 1.6. The only issue I'm having is if I have to restart asterisk( for some config update) I loose all the sip registration in asterisk, is there any way of fixing this?

9 13

Issue building sip-router (kamailio flavour)
by Santiago Gimeno 08 Jul '10

08 Jul '10

Hi, When I build sip-router (from master) with kamailio flavour using: make config mode=debug FLAVOUR=kamailio; make all; make install I'm getting this error when trying to start kamailio: ERROR: <core> [sr_module.c:396]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/tmx.so>: /usr/local/lib/kamailio/modules_k/tmx.so: undefined symbol: _tm_table On the other hand, if I use: make FLAVOUR=kamailio cfg; make all; make install kamailio starts correctly. I've tried also: make mode=debug FLAVOUR=kamailio cfg with no luck either. What's the problem? How should I compile kamailio with debugging so it starts correctly? Thank you in advance. Regards, Santi

3 5

Re: [SR-Users] no Contact in REFER generated with dlg_bridge
by Anton Roman 08 Jul '10

08 Jul '10

Hi, regarding the missing Contact header in the REFER message, it can be solved by including some lines in /modules_k/dialog/dlg_ transfer.c. In this file, the Contact of the initial INVITE generated with dlg_bridge command can be easily changed as well. The problem now is only in the processing of the ACK generated by the TM module. When this ACK leaves the Kamailio server the branch value of the topmost Via header is '0'. Below you can find the debug output where the ACK is generated and routed by Kamailio. Any idea about where the error can be? thanks in advance, regards Anton 11(5056) DEBUG: tm [t_msgbuilder.c:791]: building ACK for out-of-dialog INVITE (using RS in RR set). 11(5056) DEBUG: tm [t_msgbuilder.c:967]: ACK RURI: `sip:235@10.1.3.15:5061', NH: `sip:10.1.2.122;lr=on'. 11(5056) DEBUG: tm [t_reply.c:1059]: ->>>>>>>>> T_code=180, new_code=200 11(5056) DEBUG: tm [t_reply.c:1794]: DEBUG: local_reply: branch=0, save=0, winner=0 11(5056) DEBUG: tm [t_reply.c:356]: DEBUG: update_totag_set: new totag 11(5056) DEBUG: tm [t_reply.c:1831]: DEBUG: local transaction completed 11(5056) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x7f9767eb1268, callback type 256, id 0 entered 11(5056) DEBUG: dialog [dlg_transfer.c:207]: completed with status 200 11(5056) DEBUG: <core> [parser/parse_to.c:179]: DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-3bcd 9(5052) DEBUG: <core> [parser/msg_parser.c:612]: SIP Request: 9(5052) DEBUG: <core> [parser/msg_parser.c:614]: method: <ACK> 9(5052) DEBUG: <core> [parser/msg_parser.c:616]: uri: < sip:235@10.1.3.15:5061> 9(5052) DEBUG: <core> [parser/msg_parser.c:618]: version: <SIP/2.0> 9(5052) DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bKf7bc.00856364.0>; state=16 9(5052) DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 9(5052) DEBUG: <core> [parser/msg_parser.c:500]: parse_headers: Via found, flags=2 9(5052) DEBUG: <core> [parser/msg_parser.c:502]: parse_headers: this is the first via 9(5052) DEBUG: <core> [receive.c:137]: After parse_msg... 9(5052) DEBUG: <core> [receive.c:177]: preparing to run routing scripts... 9(5052) DEBUG: sl [sl_funcs.c:335]: to late to be a local ACK! 9(5052) DEBUG: <core> [parser/parse_to.c:179]: DEBUG: add_param: tag=94cf927f9c6eed24i1 9(5052) DEBUG: <core> [parser/parse_to.c:808]: end of header reached, state=29 9(5052) DEBUG: <core> [parser/msg_parser.c:174]: DEBUG: get_hdr_field: <To> [43]; uri=[sip:235@10.1.2.122 <sip%3A235(a)10.1.2.122>] 9(5052) DEBUG: <core> [parser/msg_parser.c:176]: DEBUG: to body [ sip:235@10.1.2.122 <sip%3A235(a)10.1.2.122>] 9(5052) DEBUG: <core> [parser/msg_parser.c:154]: get_hdr_field: cseq <CSeq>: <10> <ACK> 9(5052) DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: get_hdr_body : content_length=0 9(5052) DEBUG: <core> [parser/msg_parser.c:90]: found end of header 9(5052) DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! 9(5052) DEBUG: <core> [parser/parse_to.c:179]: DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-3bcd 9(5052) DEBUG: <core> [parser/parse_to.c:808]: end of header reached, state=29 9(5052) DEBUG: sanity [mod_sanity.c:220]: all sanity checks passed 9(5052) ERROR: <script>: kamailio!!!: ACK 9(5052) DEBUG: siputils [checks.c:68]: totag found 9(5052) DEBUG: rr [loose.c:85]: is_preloaded: No 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.1] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.2] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==10 && [10.1.3.15] == [10.1.2.122] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.1] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.2] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==10 && [10.1.3.15] == [10.1.2.122] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [forward.c:356]: check_self: host != me 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 10==9 && [10.1.2.122] == [127.0.0.1] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5060 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 10==9 && [10.1.2.122] == [127.0.0.2] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5060 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 10==10 && [10.1.2.122] == [10.1.2.122] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5060 9(5052) DEBUG: rr [loose.c:781]: Topmost route URI: 'sip:10.1.2.122;lr=on' is me 9(5052) DEBUG: <core> [parser/msg_parser.c:90]: found end of header 9(5052) DEBUG: rr [loose.c:257]: No next Route HF found 9(5052) DEBUG: rr [loose.c:800]: No next URI found 9(5052) DEBUG: rr [rr_cb.c:97]: callback id 0 entered with <lr=on> 9(5052) DEBUG: dialog [dlg_handlers.c:797]: Route param 'did' not found 9(5052) DEBUG: tm [t_lookup.c:1387]: DEBUG: t_newtran: msg id=10 , global msg id=9 , T on entrance=0xffffffffffffffff 9(5052) DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=52095, isACK=1 9(5052) DEBUG: tm [t_lookup.c:720]: DEBUG: t_lookup_request: e2e proxy ACK found 9(5052) DEBUG: tm [t_reply.c:276]: DEBUG: totag for e2e ACK found: 0 9(5052) DEBUG: tm [t_funcs.c:315]: SER: forwarding ACK statelessly 9(5052) DEBUG: <core> [msg_translator.c:200]: check_via_address(10.1.2.122, 10.1.2.122, 0) 9(5052) DEBUG: <core> [forward.c:509]: Sending: ACK sip:235@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.2.122;branch=0 Via: SIP/2.0/UDP 10.1.2.122;branch=z9hG4bKf7bc.00856364.0 From: sip:click2dial@172.22.5.17 <sip%3Aclick2dial(a)172.22.5.17> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-3bcd Call-ID: 2bef8d5474ff8cd4 To: sip:235@10.1.2.122 <sip%3A235(a)10.1.2.122>;tag=94cf927f9c6eed24i1 CSeq: 10 ACK User-Agent: kamailio (3.0.1 (x86_64/linux)) Content-Length: 0

2 2

LCR, problem with routing (prefix, from_uri, grp_id)
by Maciej Bylica 07 Jul '10

07 Jul '10

Hi ALL, I have a problem to configure LCR module to work properly (kamailio rel. 1.5) The task in my case is to handle: 1) All calls coming with CLD number prefix 482223344 from my MGW (77.77.77.77) towards kamilio and direct them to sip world (88.88.88.88) 2) Calls with CLD starting with 48 coming from SIP world towards kamailio and direct them to my MGWs. Additionaly i need to route it on the basis of CLI number. So MGWs <---> Kamailio <---> SIPProxies (88.88.88.88) In db: +----+-----------+----------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+-----------+----------------+--------+----------+ | 1 | 48 | 48 | 1 | 1 | | 2 | 48 | 49 | 2 | 1 | | 3 | 482223344 | 77.77.77.77 | 3 | 1 | +----+-----------+----------------+--------+----------+ gw, +----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+ | id | gw_name | grp_id | ip_addr | hostname | port | uri_scheme | transport | strip | tag | weight | ping | flags | +----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+ | 1 | CLI48 | 1 | 77.77.77.70 | NULL | 5060 | NULL | NULL | NULL | NULL | 150 | 0 | 0 | | 2 | CLI48backup | 1 | 77.77.77.71 | NULL | 5060 | NULL | NULL | NULL | NULL | 150 | 0 | 0 | | 3 | CLI49 | 2 | 77.77.77.75 | NULL | 5060 | NULL | NULL | NULL | NULL | 15 | 0 | 0 | | 4 | CLI49backup | 2 | 77.77.77.76 | NULL | 5060 | NULL | NULL | NULL | NULL | 15 | 0 | 0 | | 5 | SIPWORLD | 3 | 88.88.88.88 | NULL | 5060 | NULL | NULL | NULL | NULL | 250 | 0 | 0 | +----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+ Unfortunately: - call is originated on 77.77.77.77 with CLD 48222334455 number and kamailio forward this call to grp_id=3 SIPWORLD which is okay. The problem is that if the call fail, kamailio will try to use 77.77.77.70 and 77.77.77.71 from grp_id=1 which is wrong. I have no idea how to provide a kind of huntstop in grp_id=3. I've been trying with +----+-----------+----------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+-----------+----------------+--------+----------+ | 1 | 48 | 88.88.88.88 | 1 | 1 | | 2 | 48 | 88.88.88.88 | 2 | 1 | | 3 | 482223344 | 77.77.77.77 | 3 | 1 | +----+-----------+----------------+--------+----------+ but in this case i am unable to provide kamailio with CLI number routing. Here is a part of my config file... if (!load_gws()) { sl_send_reply("503", "Unable to load gateways"); exit; } if (!next_gw()) { sl_send_reply("503", "Unable to find a gateway"); exit; } route(1); exit; and failover for route(1)... failure_route[11] { # In case of failure, send it to an alternative route: if (t_check_status("408|404|5[0-9][0-9]")) { if (!next_gw()) { t_reply("503", "Service not available, no more gateways"); exit; } else { t_on_failure("11"); t_relay(); } exit; } } Anybody could help my to get out of that? Thx, Maciej.

3 19

CANCEL before INVITE
by David 07 Jul '10

07 Jul '10

Hey, I am using a Cisco WIP310 wifi phone. Seeing as wifi is very battery demanding, the phone goes into a standby mode. When it's in the standby mode, it takes a few seconds to come back on. So I send an INVITE to the phone, statefully using TM, I send out a CANCEL before the phone returns the "180 ringing" message. Somehow the device is answering the CANCEL before the INVITE, so the result is that it responds to the CANCEL with "481 Call Leg/Transaction Does Not Exist.", after that it responds to the INVITE with a "180 Ringing", the phone than rings indefinitely because the CANCEL is not sent out again as the transaction is completed from the 481 SIP message. I had a look at the CANCEL, there is no Route: header or tag on the To, so it looks like it is part of a new dialog ( I believe that's what rfc3261 says. As far as I can tell, my Kamailio is working properly. It is retransmitting the messages at the proper intervals, and it is passing along the messages as it receives them. The trouble is that the device answers the initial CANCEL before it answers any of the retransmitted INVITEs. Is there something that I can do in Kamailio to resolve this issue ? Is there an option that I can set that will cause Kamailio to relay the CANCEL only to devices that have already returned a 100 Trying or 180 Trying ? What information do you need to know about my config? What parts of the SIP trace do you need ? Thanks, David

6 19

Summary of Devel Meeting, Berlin 2010
by Daniel-Constantin Mierla 06 Jul '10

06 Jul '10

Hello, fyi: summary of the devel meeting in Berlin, June 8, 2010, was uploaded to: http://sip-router.org/wiki/meetings/berlin-2010-minutes Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

2 1

TimeStamp header
by mustafa rifaee 06 Jul '10

06 Jul '10

Hello all; I need some Clarification about timestamp header in RFC 3261; i need to use it to estimate the RTT value; how it is work in this scenario?: Invite UA --------------------> Proxy1 | | Invite | | ------------------------------ --------->Proxy2 | 100 trying | | | <---------------------------| | invite | | | | | | --------------------------------------->UA2 | | <-------100 trying------------------- | | | | | <--------180 Ringing-------------- | | | <---------------180 Ringing------- | | | | | | |<---180 Ringing----------| | | UA send invite to proxy1 with timestamp=10 ; then the Proxy1 send the invite message to proxy2 and reply with provisional response (100 trying) and with timestamp=10 0.5; When UA receive 100trying; UA will adapt its retransmition timer T1 to the calculated value from timestamp; then UA adapt T2 Timer value; is This true? Here Can Proxy1 send invite message with timestamp header?; for example timestamp=8 and when proxy2 recive invite request, it send it to UA and he send 100 trying to proxy1 with timestamp=8 0.4; Please Help me; Thanks; Mustafa AL Rifaee

1 0

About setflag()'s parameter
by zhou tianjun 06 Jul '10

06 Jul '10

Hi all, I'm in doubt of the means of setflag()'s parameter. I have looked for it everywhere including internet and the ser source code, but no any docs explained it in detail. can anybody tell me about it ? -- With best regards by sunone

5 16

$rU rewrite problem
by Dmitri Korotkov 05 Jul '10

05 Jul '10

Hello, I have problems with rewriting $rU. Kamailio 3.0.1 on debian squeeze x86_64 In config file I have: $avp(s:dnumber) = $(ru{re.subst,/^sip:\*99\*(.*)@.*/\1/}); $rU=$avp(s:dnumber); And log file shows following warnings: Jul 5 17:22:30 voice /usr/sbin/kamailio[23641]: WARNING: <core> [rvalue.c:987]: automatic string to int conversion for "sip:1234567890@voice.serv.net" failed Jul 5 17:22:30 voice /usr/sbin/kamailio[23641]: WARNING: <core> [rvalue.c:1839]: rval expression conversion to int failed (630,22-630,23) Could you please show any samples how to change $rU with regexp matched part of $rU ? BR, Dmitri

2 2

Cancel rtp proxy changes to SDP
by David 05 Jul '10

05 Jul '10

3 2

How SER verdict a session is over when missed BYEs
by zhou tianjun 05 Jul '10

05 Jul '10

Hi all, I'm using SER, SEMS and RTPPROXY, they all work fine. When caller or callee want to stop the dialog, one of them may send BYE to proxy, but the sip proxy doesn't receive it. so dows the other's BYE. In this case, how the proxy verdict the dialog is over and log it to mysql ? -- With best regards by zhoutj

2 1

About building ACD
by Trung Anh 04 Jul '10

04 Jul '10

Hi everyone, I have a problem about Agents status. Now, i want to develop ACD system but i don't know how to collect agent status. Please help me. Thanks in advance! -- Best Regards, Mr House Cricket

3 4

voicemail
by mattias jonsson 03 Jul '10

03 Jul '10

Have the server voicemail kameleo

2 1

Missing ACK
by Ole Kaas 03 Jul '10

03 Jul '10

Hello, See transmission below. For some reason the ACK package is lost/dropped and the remote server retransmits the 200 OK status. But Kamailio does not retransmit the ACK on receiving the retransmitted OK's. Eventually the remote server gives up and pulls the plug after ~20 seconds. Is this a bug or should I look closer at my routing script? The script is for the ACK handling pretty much the same at the vanilla script delivered with the source. I'm still on version 3.0.0. /Ole -> Request: INVITE sip:phonenum@destination, with session description <- Status: 100 Trying <- Status: 180 Ringing <- Status: 183 Session Progress, with session description <- Status: 200 OK, with session description -> Request: ACK sip:phonenum@destination <- Status: 200 OK, with session description <- Status: 200 OK, with session description <- Status: 200 OK, with session description <- Status: 200 OK, with session description <- Status: 200 OK, with session description <- Status: 200 OK, with session description -> Request: BYE sip:phonenum@destination -> Request: BYE sip:phonenum@destination <- Status: 481 Call leg/transaction does not exist

6 17

Re: [SR-Users] change ptime / RTP packet size
by Carsten Bock 02 Jul '10

02 Jul '10

In order to change the attribute in the SDP, you need manually to alter the SDP (at least for 1.5.x). Tha NAT-Helper module will not do this for you. Please see the Textops-Module on how to do this. Carsten 2010/7/2 Laurent Schweizer <laurent.schweizer(a)peoplefone.com> > hello, > > > > I use kamailio 1.5.4 with rtpproxy 1.2.1 > > > > I try to change the ptime (RTP packet size) with the Nathelper module . > > > > to do it I call the function rtpproxy_offer with the option "z20" but the > "a=ptime:30" in SDP is not changed > > > > any idea ? > > > > Regards > > > > Laurent > > > > Log of RTPproxy: > > > > > > Jul 2 09:40:49 dns2 rtpproxy[19050]: DBUG:handle_command: received command > "19072_4 UZ20c0,8,101 2ab4d3ad6ff9676a 217.162.26.26 3000 112a4c5d45;1" > > Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: new session > 2ab4d3ad6ff9676a, tag 112a4c5d45;1 requested, type strong > > Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: new session on a > port 54858 created, tag 112a4c5d45;1 > > Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: pre-filling > caller's address with 217.162.26.26:3000 > > Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: RTP packets from > caller will be resized to 20 milliseconds > > Jul 2 09:40:49 dns2 rtpproxy[19050]: DBUG:doreply: sending reply "19072_4 > 54858 15.43.10.128 " > > Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:process_rtp: session timeout > > Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: RTP stats: 0 in > from callee, 0 in from caller, 0 relayed, 0 dropped > > Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: RTCP stats: 0 in > from callee, 0 in from caller, 0 relayed, 0 dropped > > Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: session on ports > 54858/0 is cleaned up > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Carsten Bock Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Home +49 40 34927217 Fax +49 40 34927218 mailto:carsten@bock.info

4 5

Re: [SR-Users] sr-users Digest, Vol 62, Issue 10
by marco＠bungalski.de 02 Jul '10

02 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

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Re: [SR-Users] XCAP SERVER MODULE
by Daniel-Constantin Mierla 02 Jul '10

02 Jul '10

Hello, please address the questions to public mailing lists (e.g., sr-users(a)lists.sip-router.org), emails sent to private address only may not be answered at all or replied after long time. There are many people on ML that can answer faster and better. On 6/17/10 5:11 PM, michael ayoola wrote: > Hello Daniel, > > Thanks for your work on XCAP Server. I am quite new to kamailio. I > have succeeded in installing kamailio server and I am trying to follow > your documentation on XCAP server. However I do not know how to load > the db module. Terminology of 'db module' refers to a database connection module, like db_mysql, db_postgres, etc. Depending where you store the data, in a mysql, postgres or other type of sql server, you have to load the appropriate module. > I noticed that the sl module is already loaded as indicated in the > kamailio.cfg file (i.e loadmodule "sl.so"). There seem to many db > related module. Which do I use? Please, it will be nice if you can help. > Note that xcap_server module is not completed yet, should work with very basic xcap operations over http/1.0 or http/2.0 non-chunked body for PUT operations. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

permissions module address_reload not working
by JR Richardson 02 Jul '10

02 Jul '10

Hi All, I'm running kamailio 3.0.2. # ----- permissions params ----- modparam("permissions", "db_url", "mysql://openserro:openserro@localhost/openser") #modparam("permissions", "db_mode", 1) snip in route block: if (!allow_source_address_group()) { sl_send_reply("403", "Forbidden"); exit; } >From command line: sip-router2:/etc/kamailio# kamctl fifo address_reload 400 Address table reload failed debug output: during initial kamailio startup: 0(10655) DEBUG: permissions [address.c:122]: Number of rows in address table: 4 0(10655) DEBUG: permissions [address.c:164]: Tuple <1, 10.10.12.0, 24, 0> inserted into subnet table 0(10655) DEBUG: permissions [address.c:164]: Tuple <1, 10.10.14.0, 24, 0> inserted into subnet table 0(10655) DEBUG: permissions [address.c:148]: Tuple <1, 10.10.15.97, 32> inserted into address hash table 0(10655) DEBUG: permissions [address.c:148]: Tuple <1, 10.10.15.98, 32> inserted into address hash table 0(10655) DEBUG: <core> [db_res.c:81]: freeing 4 columns 0(10655) DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x827c158 0(10655) DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x827c220 0(10655) DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x827c168 0(10655) DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x827c230 0(10655) DEBUG: <core> [db_res.c:94]: freeing result names at 0x827c250 0(10655) DEBUG: <core> [db_res.c:99]: freeing result types at 0x827c1b8 0(10655) DEBUG: <core> [db_res.c:54]: freeing 4 rows 0(10655) DEBUG: <core> [db_row.c:97]: freeing row values at 0x827c710 0(10655) DEBUG: <core> [db_row.c:97]: freeing row values at 0x827c6b8 0(10655) DEBUG: <core> [db_row.c:97]: freeing row values at 0x827c660 0(10655) DEBUG: <core> [db_row.c:97]: freeing row values at 0x827c598 0(10655) DEBUG: <core> [db_res.c:62]: freeing rows at 0x827ca10 0(10655) DEBUG: <core> [db_res.c:136]: freeing result set at 0x827c1f8 0(10655) DEBUG: <core> [db_pool.c:102]: removing connection from the pool 0(10655) DEBUG: permissions [address.c:181]: address table reloaded successfully. result when issuing "kamctl fifo address_reload" 3(10389) DEBUG: mi_fifo [fifo_fnc.c:489]: entered consume 3(10389) DEBUG: mi_fifo [fifo_fnc.c:489]: **** done consume 3(10389) DEBUG: mi_fifo [fifo_fnc.c:503]: done parsing the mi tree 3(10389) ERROR: permissions [address.c:82]: db_handle already exists The address table loads fine during startup and if I reload kamailio it will load any new addresses or subnets so I know there is no syntax error in the database. I can reload trusted table and pdt table without an error but address table is not working. Any ideas on how I can trouble shoot this further? Thanks. JR -- JR Richardson Engineering for the Masses

4 16

Re: [SR-Users] change ptime / RTP packet size
by Alex Balashov 02 Jul '10

02 Jul '10

On 07/02/2010 04:15 AM, Laurent Schweizer wrote: > hello, > > I use kamailio 1.5.4 with rtpproxy 1.2.1 > > I try to change the ptime (RTP packet size) with the Nathelper module . > > to do it I call the function rtpproxy_offer with the option "z20" but > the "a=ptime:30" in SDP is not changed Try use force_rtp_proxy() instead. Last time I tried to use it, rtpproxy_offer()/answer() were still broken, although admittedly that was a while ago. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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Re: [SR-Users] sr-users Digest, Vol 62, Issue 9
by marco＠bungalski.de 02 Jul '10

02 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

change ptime / RTP packet size
by Laurent Schweizer 02 Jul '10

02 Jul '10

hello, I use kamailio 1.5.4 with rtpproxy 1.2.1 I try to change the ptime (RTP packet size) with the Nathelper module . to do it I call the function rtpproxy_offer with the option "z20" but the "a=ptime:30" in SDP is not changed any idea ? Regards Laurent Log of RTPproxy: Jul 2 09:40:49 dns2 rtpproxy[19050]: DBUG:handle_command: received command "19072_4 UZ20c0,8,101 2ab4d3ad6ff9676a 217.162.26.26 3000 112a4c5d45;1" Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: new session 2ab4d3ad6ff9676a, tag 112a4c5d45;1 requested, type strong Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: new session on a port 54858 created, tag 112a4c5d45;1 Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: pre-filling caller's address with 217.162.26.26:3000 Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: RTP packets from caller will be resized to 20 milliseconds Jul 2 09:40:49 dns2 rtpproxy[19050]: DBUG:doreply: sending reply "19072_4 54858 15.43.10.128 " Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:process_rtp: session timeout Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: session on ports 54858/0 is cleaned up

1 0

UAC, parsing auth header failed
by Ján ONDREJ (SAL) 02 Jul '10

02 Jul '10

Hello, our voip provider requires autentication to voip services. Trying to use UAC module, I get "401 Unauthorized" packet, but looks like all ser based programs have problems to parse this packet. Can somebody make a patch to fix this? Here is message from logs: Jun 17 15:01:42 kamailio /usr/sbin/kamailio[1291]: ERROR: uac [auth_hdr.c:267]: parse error in <DIGEST realm="BroadWorks",qop="auth",algorithm=MD5,nonce="BroadWorksXgajlo4shTqsrkgdBW"> around 0 And ngrep sniffed packet: SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 100.100.242.20;branch=z9hG4bK119f.8a41a626.0. Via: SIP/2.0/UDP 100.100.16.77;branch=z9hG4bK119f.7cc235e.0. Via: SIP/2.0/UDP 100.100.10.10;branch=z9hG4bK8ec5c892A88F71C9. From: "Ondrej Jan" <sip:556807505@as.vvn.t-com.sk>;tag=89D71D23-32F3D9BC. To: <sip:0248213335@as.vvn.t-com.sk;user=phone>;tag=1621075049-1276779704129. CSeq: 2 INVITE. Call-ID: 41c7437-58259a9d-d157be06(a)100.100.10.10. WWW-Authenticate: DIGEST realm="BroadWorks",qop="auth",algorithm=MD5,nonce="BroadWorksXgajlo4shTqsrkgdBW". Content-Length: 0. I tryed to look at sources, but this "C" code is too complicated for me. :-( SAL

6 11

One way audio on calls from pstn
by Dmitri Korotkov 02 Jul '10

02 Jul '10

Hello, I have kamailio installation WITH_PSTN, WITH_NAT and rtpproxy. Using following scenario: [kamailio]<-sip trunk ->[asterisk gw] <->sip trunk <-> [PSTN provider] All kamailio sip subscribers are behind nat in different networks. 1. OK. Local kamailio users can call one to other even they are on different networks behind nat. 2. OK. Outgoing calls from kamailio users to PSTN work also very well. 3. Not OK. Incoming from PSTN side calls have only one way audio. I tcpdump'ed kamailio box and found, that pstn provider sends RTP packets to kamailio IP in case of answered call. I guess that rtpproxy is not active in case of pstn call. Is it true ? I am using more less default kamailio config Could you please suggest solution ? BR, Dmitri

2 6

Re: [SR-Users] sr-users Digest, Vol 62, Issue 8
by marco＠bungalski.de 01 Jul '10

01 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

Re: [SR-Users] sr-users Digest, Vol 62, Issue 7
by marco＠bungalski.de 01 Jul '10

01 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

Re: [SR-Users] sr-users Digest, Vol 62, Issue 6
by marco＠bungalski.de 01 Jul '10

01 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

Re: [SR-Users] sr-users Digest, Vol 62, Issue 5
by marco＠bungalski.de 01 Jul '10

01 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 12.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

Re: [SR-Users] error loading perl module
by Daniel-Constantin Mierla 01 Jul '10

01 Jul '10

Hello, keep cc-ing the mailing list for any message, so next people that come and scan mailing list archive for the same issue discover the solution and whether was good or not. Cheers, Daniel On 6/30/10 11:40 PM, "Nicolas Rüger" wrote: > Hello Daniel, > > thanks for the fast answer. I followed your suggestion and now (after fixing a few minor bugs) it works! :) > > Many many thanks... > > Nicolas > > > > > > -------- Original-Nachricht -------- > >> Datum: Wed, 30 Jun 2010 21:44:36 +0200 >> Von: Daniel-Constantin Mierla<miconda(a)gmail.com> >> An: "Nicolas Rüger"<NicolasRueger(a)gmx.de> >> CC: sr-users(a)lists.sip-router.org >> Betreff: Re: [SR-Users] error loading perl module >> > >> Hello, >> >> On 6/30/10 9:24 PM, "Nicolas Rüger" wrote: >> >>> Hello, >>> >>> I try to use the PERL module in kamailio v3 for some research due to my >>> >> thesis. >> >>> I'm using debian lenny, kamailio_3.0.0 and perl_5.10.0. >>> >>> >>> 1.) The Problem >>> >>> Starting Kamailio via "./kamctl start" drops the general error JUST when >>> >> I try to load the PERL module: >> >>> ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start >>> >> failed >> >>> >>> >>> >>> 2.) The search for errors >>> >>> I got the following errors (and warnings) in DEBUG mode by starting >>> >> "kamailio -d" for debugging: >> >>> [...] >>> >>> 0(10776) DEBUG:<core> [sr_module.c:382]: load_module: trying to >>> >> load</opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so> >> >>> 0(10776) WARNING:<core> [sr_module.c:445]: >>> >> /opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will >> not be supported in newer versions; consider using mod_register() instead >> 0(10776) >> >>> [...] >>> >>> 0(10778) ERROR: perl [perl.c:330]: insufficient module parameters. >>> >> Module not loaded. >> >>> 0(10778) ERROR:<core> [sr_module.c:874]: init_mod(): Error while >>> >> initializing module perl >> >>> >>> >> seems you don't have the parameter 'filename' set: >> >> http://kamailio.org/docs/modules/stable/modules_k/perl.html#id2928387 >> >> Cheers, >> Daniel >> >> >> >>> >>> >>> 3.) What I did before... >>> >>> 3.1) Compiled Kamailio >>> >>> I compiled Kamailio 3.0.0 using the following commands, where $1 is the >>> >> target: >> >>> make prefix=$1/ include_modules="db_mysql tls perl" cfg >>> >>> make Q=1 all >>> >>> make install >>> >>> #make prefix=$1/ TLS=1 >>> #make prefix=$1/ TLS=1 modules >>> #make prefix=$1/ TLS=1 install >>> >>> #make prefix=$1/ include_modules="db_mysql" modules >>> #make prefix=$1/ include_modules="db_mysql" install >>> >>> >>> 3.2) Set the environment variables as described in >>> >>> http://kamailio.org/docs/modules/stable/modules_k/perl.html#id2920768 >>> >>> You need to set the environment variables PERLLDOPTS, PERLCCOPTS and >>> >> TYPEMAP to values similar to the output of >> >>> PERLLDOPTS: perl -MExtUtils::Embed -e ldopts >>> PERLCCOPTS: perl -MExtUtils::Embed -e ccopts >>> TYPEMAP: echo "`perl -MConfig -e 'print >>> >> $Config{installprivlib}'`/ExtUtils/typemap" >> >>> 3.3) Trying to load the perl-module via kamailio.cfg >>> >>> loadmodule "perl.so" >>> >>> >>> >>> >>> Can you PLEASE help me to solve the problem! >>> >>> Thank you very much... >>> >>> Cheers Nicolas >>> >>> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com/ >> > -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

using avpops for querying an external db on kamailio 3.0.2
by yaron nahcum 01 Jul '10

01 Jul '10

Hi, I have recently installed a kamailio server with release 3.0.2. I would like to use the kamilio as a redirect server that would recieve REGISTER messages from users and query a proprietary database that holds users information - if the users are registered to the service it would redirect the user to the registrar. I need it for security reasons which I can't fully explain. Now, I am using the avpops module. I have been able to make it work using the following configuration: modparam("avpops","db_url","mysql://user:pass@localhost/mysql") modparam("avpops","avp_table","proptable") modparam("avpops","db_scheme","scheme0:username_col=sipUsername;value_col=sipDomainName;table=proptable") ---------- if (is_method("REGISTER")) { if (avp_db_load("$fu/username","$avp(s:CONSTANT_DOMAIN)/$scheme0")) { append_branch("sip:registrarIP"); sl_send_reply("302","Test Redirect"); } else { sl_send_reply("404","Test Not found"); } -------------- The thing is that I don't need to match the string CENTREX against the domain- it is something I added because I couldn't find another way to do it. I would like to match only the username. I tried wildcards or leaving it empty but it didn't work. I tried not to use the scheme and instead use the table it self but again it didn't work. Is it possible? Do you know a better to do it? Thanks, Yaron.

5 5

error loading perl module
by "Nicolas Rüger" 30 Jun '10

30 Jun '10

Hello, I try to use the PERL module in kamailio v3 for some research due to my thesis. I'm using debian lenny, kamailio_3.0.0 and perl_5.10.0. 1.) The Problem Starting Kamailio via "./kamctl start" drops the general error JUST when I try to load the PERL module: ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed 2.) The search for errors I got the following errors (and warnings) in DEBUG mode by starting "kamailio -d" for debugging: [...] 0(10776) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so> 0(10776) WARNING: <core> [sr_module.c:445]: /opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead 0(10776) [...] 0(10778) ERROR: perl [perl.c:330]: insufficient module parameters. Module not loaded. 0(10778) ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module perl 3.) What I did before... 3.1) Compiled Kamailio I compiled Kamailio 3.0.0 using the following commands, where $1 is the target: make prefix=$1/ include_modules="db_mysql tls perl" cfg make Q=1 all make install #make prefix=$1/ TLS=1 #make prefix=$1/ TLS=1 modules #make prefix=$1/ TLS=1 install #make prefix=$1/ include_modules="db_mysql" modules #make prefix=$1/ include_modules="db_mysql" install 3.2) Set the environment variables as described in http://kamailio.org/docs/modules/stable/modules_k/perl.html#id2920768 You need to set the environment variables PERLLDOPTS, PERLCCOPTS and TYPEMAP to values similar to the output of PERLLDOPTS: perl -MExtUtils::Embed -e ldopts PERLCCOPTS: perl -MExtUtils::Embed -e ccopts TYPEMAP: echo "`perl -MConfig -e 'print $Config{installprivlib}'`/ExtUtils/typemap" 3.3) Trying to load the perl-module via kamailio.cfg loadmodule "perl.so" Can you PLEASE help me to solve the problem! Thank you very much... Cheers Nicolas -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

2 1

force_send_socket - multiple times per dialog, subsequent times fail
by Geoffrey Mina 30 Jun '10

30 Jun '10

I am having an issue with 1.5.4 where force_send_socket() isn't behaving as I would expect. I have an LCR scenario where depending on which gateway I am sending to, I send from a different socket. It appears that calling force_send_socket from failure_route has no affect on the routing. In my initial route block I call force_send_socket(X.X.X.180:5060). In my failure_route I am calling force_send_socket(X.X.X.179:5060), yet after calling the subsequent time, my sip trace and tcpdump are clearly showing kamailio is still sending from X.X.X.180:5060. I have added a lot of debug lines, so I am 100% sure the second call to force_send_socket is happening. Am I misusing this module in some way?

5 8

[nathelper] clarification about rtpproxy_retr parameter
by Iñaki Baz Castillo 30 Jun '10

30 Jun '10

Hi, according to the doc: 1.5.9. rtpproxy_retr (integer) How many times nathelper should retry to send and receive after timeout was generated. Default value is “5”. So imagine this configuration: modparam("nathelper", "rtpproxy_tout", 2) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_retr", 10) Let's imagine that the RtpProxy servers crashes and is down for 1 day. Does this configuration mean that the rtpproxy server will NEVER be contacted again by nathelper module since 60x10 seconds < 1 day? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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