- sr-users - kamailio.org

Extracting P-Asserted-Identity value
by Grant Bagdasarian 11 Apr '16

11 Apr '16

Hello, How do I extract the value of "tel" from multiple P-Asserted-Identity headers? Example: P-Asserted-Identity: "Cullen Jennings" <sip:fluffy@cisco.com> P-Asserted-Identity: tel:+14085264000 What I got so far is: $var(p_asserted_identity) = $(hdr(P-Asserted-Identity)[*]); $var(caller_id) = $var(p_asserted_identity){param.value,'tel',':'}; But this does not work. I keep getting parser errors when doing the transformation (2nd line of code example): I'm not sure how to format the line of code. I tried multiple ways, but none work. Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: syntax error Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 53: '('')' expected (function call) Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 53: bad command: missing ';'? Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 59: '('')' expected (function call) Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 59: bad command: missing ';'? Apr 8 18:35:20 kamailio: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 60-64: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 65: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 66-68: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 408, column 6: syntax error Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 408, column 6: Regards, Grant

3 6

Help with Kamailio java module
by Bonjour Madame 11 Apr '16

11 Apr '16

Hi! I am a newbie and I need help understanding how to use the java module with Kamailio. I am trying to create a basic java application that will print the SIP message type being processed by Kamailio. I read the app_java module but can't still figure it out. Any help or reference will be greatly appreciated. Thanks Sent from Mail for Windows 10

2 3

Error in USERLOCDB
by Alberto Sagredo 11 Apr '16

11 Apr '16

Im getting this error using USERLOCDB Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Duplicate entry '0--1' for key 'connection_idx' (1062) Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: usrloc [ucontact.c:684]: db_insert_ucontact(): inserting contact in db failed xxxx (uloc-5706bebd-7415-3) Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: usrloc [urecord.c:591]: insert_ucontact(): failed to insert in database Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: registrar [save.c:522]: insert_contacts(): failed to insert contact My config # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "db_ops_ruid", 1) modparam("usrloc", "db_check_update", 1) #!endif I have been using it without any problems in other installations but in a new one with 4. KAMAILIO im getting this error. I can see registrations on kamctl ul show but seems that option 1 is not working . Option 2 does not update registrations from that device. Thanks

2 3

truncating sip message
by Yasin CANER 11 Apr '16

11 Apr '16

4 10

Multiple SIP-servers with SRV-records and authentication secrets
by Alfred E. Heggestad 09 Apr '16

09 Apr '16

Dear SIP-experts and DNS-SRV gurus; I have some questions to the deployers of SER/Kamailio and best current practice for multiple SIP-servers with SRV-records and authentication. This is not a question about Kamailio itself but rather experience with deployment of it in the field. The current usecase is: 1. Multiple SIP-servers are deployed for the same domain 2. The DNS is configured with SRV-records for load balancing, example: (lets call the domain "example.com") $ host -t SRV _sip._udp.example.com _sip._udp.example.com has SRV record 20 0 5080 alpha1.example.com. _sip._udp.example.com has SRV record 20 0 5080 alpha2.example.com. 3. when a SIP client registers, it resolves the domain using RFC3263 [1] and the first REGISTER request is sent to SIP-Server #1 4. SIP-server #1 replies with 401 containing the authentication challenge 5. The SIP Client adds the authentication header to the REGISTER request and re-sends it, but this time also using RFC 3263, and due to DNS rotation the request is sent to SIP-Server #2 6. Now, because the SIP-Servers are configured with _different_ secrets in the "auth" module [2], the REGISTER request fails with authentication error. Now, I know that it is common for SIP user-agents to send both requests to the same SIP-server instance. Baresip [3] is not doing that, it does a new RFC 3263 lookup for all requests (except e2e ACK/CANCEL). so here are my questions: - What is common practice in the field, to configure auth module with the same "secret" or different "secret" values? - Do you know if there is any reference to IETF documents about how this should be handled? RFC 3263 says that every request should be resolved, except: "The procedures here MUST be done exactly once per transaction, where transaction is as defined in [1]. That is, once a SIP server has successfully been contacted (success is defined below), all retransmissions of the SIP request and the ACK for non-2xx SIP responses to INVITE MUST be sent to the same host. Furthermore, a CANCEL for a particular SIP request MUST be sent to the same SIP server that the SIP request was delivered to." - What is common practice for SIP user-agents to do in this case? /alfred [1] https://tools.ietf.org/html/rfc3263#section-4.4 [2] http://www.kamailio.org/docs/modules/3.4.x/modules/auth.html#auth.secret [3] https://github.com/alfredh/baresip/issues/39

4 4

Kamailio Updates on VUC 588 - this week on Friday
by Daniel-Constantin Mierla 08 Apr '16

08 Apr '16

Hello, The weekly VoIP Users Conference (VUC) #588 moderated by Randy Resnick will host a session for Kamailio Updates. Everyone from community is invited to join and actively participate to discussions. There are many ways to connect, from SIP to classic PSTN dial in as well as listening or watching live streams on the web. The live session will start on Friday, April 8, at 18:00 Berlin timezone (17:00 London, 12:00EDT). More details about the session are available at: - https://www.kamailio.org/w/2016/04/vuc-588-kamailio-updates/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

Question about debugging using `kamcmd pkg.stats`
by Lucian Balaceanu 08 Apr '16

08 Apr '16

Hi Daniel, While debugging a 'no private memory left' issue by using `kamcmd pkg.stats` we stumbled upon data we don't understand. Could you help with some information? `kamcmd pkg.stats`returns free values of about 8M per process with an exception: kamailio with no apparent problems: `timer` process: free value: 8M used value: 100M real_used: 308M. kamailio with no private memory left: `timer` process: free value: 8/9M used value: 98M real_used: 662M (both mentioned kamailios have roughly the same load) Are these timer values normal? What is this real_used value and how can it be bigger than the allocated pkg mem? Additionally, how come we don't see the fifo process pkg stats in the output of `kamcmd pkg.stats`? We get the error message when doing reloading backlists via kamctl fifo. Thank you, Lucian Balaceanu

2 6

Compatibility between Siremis 4.2 and Kamailio 4.4
by Jay Li 08 Apr '16

08 Apr '16

Dear all, I got into the Kamailio world recently, and just wanted to get some info here about the compatibility between Siremis and Kamailio. From what I read, Siremis is wonderful web management tool for Kamailio. However the latest Siremis to date is 4.2.0 which was released more than a year ago. From the release note, it was intended to work with Kamailio 4.2, but Kamailio 4.4 has already released. I'm wondering if there's known compatibility issues between Siremis 4.2 and later released of Kamailio, like 4.3 and 4.4. Any information is appreciated. Regards,Jay

2 1

Insert callerid in Kamailio
by Nelson Migliaro 07 Apr '16

07 Apr '16

Hello, I am planning to insert the caller id in my Kamailio before sending the call to sip vendor. I am planning to send a prefix before the number and use that prefix in the SQL query to select the CLID. I was thinking in the sqlops to select the CLID and then append the result in the Remote-Party-ID: modparam("sqlops","sqlcon","cb=>mysql://kamailio:abc@10.10.1.1/testdb") modparam("sqlops","sqlcon","ca=>dbdriver://username:password@dbhost/dbname") ... ... sql_query("ca", "select clid from mitable where prefix = $var(prefix)", "ra"); xlog("number of rows in table domain: $dbr(ra=>rows)\n"); sql_result_free("ra"); Question..... ¿Is there an specific module to do this or a there a better way? Thank you Nelson.-

2 1

SIP Message Body
by Vladimer Gabunia 07 Apr '16

07 Apr '16

How can I Catch Text Message in SIP Method Message ? ________________________________ [gh.ge] ვლადიმერ გაბუნია IT სამსახურის უფროსი ტელ: (+995) 32 2505222 +8183 მობ: (995) 577 095333 შპს "ჯეო ჰოსპიტალს" სათავო ოფისი თბილისი 0160, ვაჟა-ფშაველას გამზ. № 16; http://www.gh.ge <http://gh.ge>

5 4

rtp proxy truncates sdp body
by Yasin CANER 06 Apr '16

06 Apr '16

1 0

SQLOPS select database in realtime
by Jon Bonilla (Manwe) 06 Apr '16

06 Apr '16

Hi all I have multiple auth credentials split among multiple databases (same server). I want to extract the password from the correct database for each request and pass it as arameter to the auth functions. The problem is, how can I use sqlops and select the correct database in realtime? the db connection already contains the database to connect to. Would it if I use a default connection but I put the db I want i the query? Something like: select * from database.table ... select * from $avp(s:database).table where i=1... cheers, Jon

3 2

pseudo variables in strings
by Juha Heinanen 05 Apr '16

05 Apr '16

If I have a string that contains pseudo variables, e.g,, $var(uri_pattern) = "sip:$fU@tutpro.com"; is it somehow possible to replace them with their values, something like: $var(uri) = eval("$var(uri_pattern)"); -- Juha

3 8

about rtpengine versions
by Juha Heinanen 05 Apr '16

05 Apr '16

I noticed that 15 days ago there was rtpengine commit https://github.com/sipwise/rtpengine/commit/aac8899b612dc8103b89f3f9c921f88… titled "Release new version 4.4.0.0+0~mr4.4.0.0", but I could not find the corresponding branch or tag. If someone wants to use that version, is 'git archive aac8899b612dc8103b89f3f9c921f88af3501303' the only way to do so? -- Juha

2 2

Re: [SR-Users] Ka--כרmailio asאעא ער מאיכקטח- a presence server
by Shlomi Gutman 05 Apr '16

05 Apr '16

אני אה כן בתאריך 4 באפר׳ 2016 8:45 לפנה״צ,‏ "Daniel-Constantin Mierla" < miconda(a)gmail.com> כתב: > Hello, > > can you run with debug=3 in config file and see if you get some hints from > syslog messages about what happens ? > > Cheers, > Daniel > > On 01/04/16 17:17, Igor Olhovskiy wrote: > > Hi! > I want to create custom Presence server. Main idea - phones subscribes to > it (for ex, extension 100@domain). This subscription actually lit BLF > button on a phone. > But main idea - change status of BLF can arrive from different locations > via PUBLISH, I assume. > Problem - phones has different type of SUBSCRIBE, like > dialog(application/dialog-info+xml) or presence(application/pidf+xml). Some > of phones are not fully follow RFC (Like using in SUBSCRIBE R-URI not > domain, but IP, but in From or To - domains are used) > Phones are registered elsewhere, not on Kamailio. After reading RFC > (especially event:dialog part) I’ve found about SIP-If-Match part, but > problem about sync this data across multiple PUBLISH locations. > Can be Kamailio be a presence server in this scenario? > > For ex, SUBSCRIBE > > SUBSCRIBE sip:212.232.26.232:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.88.60:5060;branch=z9hG4bK1656003104;rport > From: <sip:608@master.rufan.at>;tag=521811605 > To: <sip:*55@master.rufan.at>;tag=05ea6038656678bd0198b8977f3c0221.4e93 > Call-ID: 805970257-5060-2(a)BJC.BGI.II.GA > CSeq: 20202 SUBSCRIBE > Contact: <sip:608@192.168.88.60:5060> > X-Grandstream-PBX: true > Max-Forwards: 70 > User-Agent: Grandstream GXP2160 1.0.5.33 > Expires: 180 > Supported: replaces, path, timer, eventlist > Event: dialog > Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, > UPDATE, MESSAGE > Content-Length: 0 > > > I’ve tried send to Kamailio this PUBLISH > > PUBLISH sip:*55@master.rufan.at:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.0.20.71:5060;branch=z9hG4bK-6930-1-0 > Max-Forwards: 70 > To: <sip:*55@master.rufan.at> > From: <sip:*55@master.rufan.at:5060>;tag=1 > Call-ID: 1-6930(a)127.0.1.1 > CSeq: 1 PUBLISH > Contact: <sip:mod_sofia@127.0.1.1:5060> > Event: presence > Expires: 3600 > Content-Type: application/pidf+xml > Content-Length: 486 > > <?xml version="1.0" encoding="ISO-8859-1"?> > <presence xmlns='urn:ietf:params:xml:ns:pidf' > xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' > xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' > xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:* > 55(a)master.rufan.at'> > <tuple id='z9hG4bK-6930-1-0'> > <status> > <basic>open</basic> > </status> > </tuple> > <dm:person id='1'> > <rpid:activities> > <rpid:on-the-phone/> > </rpid:activities> > <dm:note>Night</dm:note> > </dm:person> > </presence> > > Kamailio answers with 200OK, but after this - no NOTIFY to subscribed > clients. > > -- > Best regards, > Igor > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlahttp://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > _______________________________________________ > 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 > >

1 0

Kamailio as a presence server
by Igor Olhovskiy 05 Apr '16

05 Apr '16

Hi! I want to create custom Presence server. Main idea - phones subscribes to it (for ex, extension 100@domain). This subscription actually lit BLF button on a phone. But main idea - change status of BLF can arrive from different locations via PUBLISH, I assume. Problem - phones has different type of SUBSCRIBE, like dialog(application/dialog-info+xml) or presence(application/pidf+xml). Some of phones are not fully follow RFC (Like using in SUBSCRIBE R-URI not domain, but IP, but in From or To - domains are used) Phones are registered elsewhere, not on Kamailio. After reading RFC (especially event:dialog part) I’ve found about SIP-If-Match part, but problem about sync this data across multiple PUBLISH locations. Can be Kamailio be a presence server in this scenario? For ex, SUBSCRIBE SUBSCRIBE sip:212.232.26.232:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.88.60:5060;branch=z9hG4bK1656003104;rport From: <sip:608@master.rufan.at>;tag=521811605 To: <sip:*55@master.rufan.at>;tag=05ea6038656678bd0198b8977f3c0221.4e93 Call-ID: 805970257-5060-2(a)BJC.BGI.II.GA CSeq: 20202 SUBSCRIBE Contact: <sip:608@192.168.88.60:5060> X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.5.33 Expires: 180 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 I’ve tried send to Kamailio this PUBLISH PUBLISH sip:*55@master.rufan.at:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.71:5060;branch=z9hG4bK-6930-1-0 Max-Forwards: 70 To: <sip:*55@master.rufan.at> From: <sip:*55@master.rufan.at:5060>;tag=1 Call-ID: 1-6930(a)127.0.1.1 CSeq: 1 PUBLISH Contact: <sip:mod_sofia@127.0.1.1:5060> Event: presence Expires: 3600 Content-Type: application/pidf+xml Content-Length: 486 <?xml version="1.0" encoding="ISO-8859-1"?> <presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:*55@master.rufan.at '> <tuple id='z9hG4bK-6930-1-0'> <status> <basic>open</basic> </status> </tuple> <dm:person id='1'> <rpid:activities> <rpid:on-the-phone/> </rpid:activities> <dm:note>Night</dm:note> </dm:person> </presence> Kamailio answers with 200OK, but after this - no NOTIFY to subscribed clients. -- Best regards, Igor

3 4

dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)
by TEG AMJG 04 Apr '16

04 Apr '16

Hi I am new to Kamailio and i am just trying to setup a kamailio (v4.2.3) dispatcher to distribute calls to 2 asterisk using the call load distribution algorithm (alg 10). I started to test and see how all it is working and in the logs appears that the call load record it cannot be found. - ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (call id generated by asterisk) I am getting this error trying a 2 call leg conversation when asterisk is creating the BYE message destinated to the callee after receiving a BYE message from the caller, and the BYE message by Asterisk reach Kamailio to then apply the ds_load_update function >From what i understand about ds_load_update it is that it sets the load to an specific destination if Kamailio receives a 2xx reply of an INVITE message, and removes it if Kamailio receives a BYE message based from the Call-id received in that message, where it is actually failing in my implementation at the moment Here are the blocks of code of the dispatcher parameters and where the function ds_load_update() is being used: #!ifdef WITH_LOADBALANCE #Parameters using Dispatcher from DB # ---------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") # ---------- #Enabling failover mechanism # ----------- modparam("dispatcher", "flags", 2) # ----------- #Setting dispatcher parameters in AVP # ----------- modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # ----------- #Setting dispatcher result in PV variables # ----------- modparam("dispatcher", "attrs_pvname", "$var(attrs)") # ----------- #Enabling dispatcher algorithm according to call loads # ----------- modparam("dispatcher", "ds_hash_size",8) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) # ----------- # Setting parameters for probing dispatcher destinations # ----------- modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_from", "sip:kamailio1@<Kamailio IP>") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold", 2) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404") # ----------- ... route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #!ifdef WITH_LOADBALANCE if(is_method("BYE|CANCEL")){ xlog("L_ERROR","UBICAR BYE ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs\n"); ds_load_update(); } #!endif ... } ... onreply_route[MANAGE_REPLY] { #!ifdef WITH_LOADBALANCE if(is_method("INVITE")) { if(status=~"2[0-9][0-9]") { xlog("L_ERROR","UBICAR setting ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs"); ds_load_update(); }else if(status=~"[3-7][0-9][0-9]") { ds_load_unset(); } } #!endif if(status=~"[12][0-9][0-9]"){ route(NATMANAGE); } } This is what i am getting from logs after the 2 call legs conversation completes: kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR: FROMASTERISK 70175143146f92975a5bff8876435911(a)192.168.10.194 kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194 method=INVITE callid=28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to= sip:104@192.168.10.194 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4083]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194:5060 method=INVITE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.10.194:5060 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4084]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.10.190:5060 from=sip:103@192.168.30.1:5060 method=BYE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.30.1:5060 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.30.2:22523;transport=UDP from= sip:103@192.168.10.194 method=BYE callid= 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to=sip:104@192.168.10.194 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for ( 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194) Kamailio IP: 192.168.10.194 Asterisk IP: +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | 1 | 0 | sip:192.168.10.190:5060 | 2 | 0 | duid=PBX1;my=pbx1;maxload=200 | pbx1 | | 2 | 0 | sip:192.168.10.191:5060 | 2 | 0 | duid=PBX2;my=pbx2;maxload=200 | pbx2 | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+

3 4

get_redirects with t_next_contacts
by Marrold 04 Apr '16

04 Apr '16

Hi all, I am having some issues using get_redirects() from the UAC_Redirect module in conjunction with t_next_contacts from the TM Module. I've searched Google and the list but the only references I can find are several years old. I am attempting to configure Kamailio so it if receives a 3XX redirect message, it will store the Contacts it contains, and attempt them in 'Q' parameter order. So for the Contact below- Contact: <sip:10.0.97.21:5060>;q=0.2,<sip:5.29.13.139:5060>;q=0.5. I want Kamailio to try 5.29.13.139, and if that times out, or is rejected in someway, try 10.0.97.21. If I use t_relay rather than t_next_contacts, kamailio branches the calls and relays them in parallel regardless of Q value - this is not what I want. If I use t_next_contacts, it attempts 5.29.13.139 which is rejected with a 401, but it doesn't attempt 5.29.13.139 and responds "408", "Request Timeout MH 2" Relevant script section is below, has anyone been able to get this working? failure_route[302] { if (t_check_status("302")){ xlog("L_ALERT","Failure Route: 302\n$mb"); get_redirects("*"); if (!t_load_contacts()) { send_reply("500", "Server Internal Error - Cannot load contacts MH"); exit; }; if (!t_next_contacts()) { t_reply("408", "Request Timeout MH"); } else { t_on_failure("302"); t_relay(); exit; }; } else if (t_check_status("[456][0-9][0-9]")){ if (!t_next_contacts()) { t_reply("408", "Request Timeout MH 2"); } else { t_on_failure("302"); t_relay(); exit; } } } Thanks, Matthew

1 2

Using 302 redirects with Dispatcher Module
by Marrold 04 Apr '16

04 Apr '16

Hi All, I'd like to use the Dispatcher Module along with 30X redirects, to monitor gateway availability with SIP options and to provide redundancy / fail over. Some sources online suggest it's as easy as - if ( method=="INVITE" ) { dst_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } However when the 30X reply is sent, it contains no contact header. Please could someone advise if it's possible to use the Dispatcher module with 302 redirects, and what they syntax is? Thanks, Matthew

3 6

kamailio number translation and route
by liwei＠neulinx.com 04 Apr '16

04 Apr '16

hi all; I want to use kamailio achieve: First phase: 1) inbound caller number translation; 2) outbound caller number translation; 3) inbound callee number translation; 4) outbound callee number tanslation; Example : Call Flow backend server ------------------> kamailio --------------------------> GateWay ------------------->PSTN caller -->callee caller ---> callee 1000 --> 2000 111000--->222000 inbound outbound caller --->callee caller ---> callee 1000 ---> 2000 11000 ---> 22000 | translation | | translation | 11000 ---> 22000 111000---> 222000 Second phase: / ---->gateway1 ----->pstn / backend server ---> kamailio ----->gateway2 -----> pstn \ \ ----> gateway3 ------->pstn Could you give me some suggest or example configuration？ liwei(a)neulinx.com

2 1

Load balancing traffic based on SIP URI
by NITESH BANSAL 04 Apr '16

04 Apr '16

Hello, I want to use Kamailio to load balance traffic across multiple asterisks.The problem is that my SIP traffic is related to conference, so if a SIP callfor a particular URI is sent to an Asterisk instance, any subsequent calls for thatSIP URI should be sent to the same Asterisk instance.Is there anyway to achieve this in Kamailio? Thanks,Nitesh

3 5

SIP Authentication
by Girish 04 Apr '16

04 Apr '16

Hello Sir, I am a newbie to Kamailio and I am trying my first setup. I would like to use Kamailio to take in traffic using a SIP Trunk and Route the calls to another SIP trunk based on prefix. I have added the incoming SIP trunk IP address to the address table on the MYSQL but I am unable to get a call routed out. Is there any sample configuration for Authentication via IP address and route the incoming SIP INVITE. Regards, Phoenix

2 1

RPID and Auth DB
by Alberto Sagredo 04 Apr '16

04 Apr '16

Hi! I plan to use rpid field but when trying to get from database, subscriber table has rpid field filled, i get always NULL. Is there any more needed than: modparam("auth_db", "load_credentials", "rpid") ? Just a check of variable xlog(" Valor de RPID es : $avp(s:rpid)"); Shows me NULL all times Thanks

2 1

KAMAILIO tcp_send query, connection problem
by Ed Todd 04 Apr '16

04 Apr '16

KAMAILIO tcp_send query, connection problem I have a KAMAILIO SIP server on a static IP from Comcast, using a modem/router Cisco DPC39412B. I have a PC application server which is successfully signing on to the SIP server, via a LAN cable on the same network as the SIP server (Cisco). In my second test, the PC application server is connected to the internet via a ZTE Falcon Z-917. In a third test, the PC application server is connected to the internet via a Talk-Talk fiber broadband. The connection does not occur. >From the log files, the successful connection says: Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (7817 (14)), reusing fd Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f6270b8a4a0 n=565 fd=1 On the failed connection: Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1818]: tcp_send(): tcp_send: no open tcp connection found, opening new one Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 192.168.0.146 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 55133, type 3 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 1776:802:0, 803 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSc<default A while later, I get: Apr 1 09:41:37 SIP-Server kamailio[7804]: DEBUG: usrloc [urecord.c:357]: wb_timer(): Binding 'hvs_us_iphone','sip:hvs_us_iphone@172.56.8.182:44341;transport=TLS;line=75dc89bd64369c7' has expired Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7f6270b8a4a0 (ticks=862851529, timeout=862909176 (3602 s), wr_timeout=862851529 (0 s)), write queue: 197 bytes Apr 1 09:41:38 SIP-Server kamailio[7820]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.0.146:55133 failed (timeout) Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7f6270b8a4a Any clues or tips would be greatly appreciated! Ed

2 1

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 03 Apr '16

03 Apr '16

Sure it is! This is exactly what I was looking for! Actually I wasn't able to see that there was a way to perform this action! Thanks for your help, Olle! I will try to create this, and when I have it done, I will post my script here, so if someone else need the same problem I can help. Now, I have just one more question about my scenario: when I register a user, is there a way to create something like a "custom field" on the location? To explain better: If possible, I can add something like a custom header that says if the user is registered from a phone device or the web interface, then I can perform the loop and replace only the oldest contact using that device... *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > This is my problem... I want to have 2 contacts for AOR, and not only one. > > When the 3rd arrives, it must remove the oldest and continue with 2... > > *If I am using the 0x04 parameter, it will save just one contact, and the > user will not be able to get the Invite both, in the mobile device and web > interface.* > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Hello, *Olle!* >> >> *Thanks for your help in this case, and don't worry about the time >> (actually I wasn't even expecting to get an answer today).* >> >> >> *I know the max_contacts parameter. But the problem is that if I set the >> max_contacts to 2, when the user tries to register again, he gets an error >> message 503. The point is that I want to get the user registered, and >> replace the oldest location entry for the newest one.* >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> Hello all. >>> >>> I have problem here and I tried to find a solution and search over >>> internet, but without success. >>> >>> My scenario is the following: I have an application that must be >>> registered in Kamailio when a user logs in the web interface, so he can get >>> calls (something like a web softphone using webrtc). At this point, we are >>> OK, and everything is working fine. >>> >>> To get these register functions I am using the parameter >>> "save("location", "0x04")" as described in the REGISTRAR module >>> documentation, so if a user logs in another web browser or computer, only >>> the last one will continue registered and all calls will be forwarded to >>> him. >>> >>> The point is that now we are creating a phone app that will do almost >>> the same thing as the web interface, allowing users to receive calls using >>> the mobile device, and here is my problem: I want to allow my users to be >>> registered on two devices at the same time, but if a user logs into another >>> device, I don't want to reply with a 503. I want to allow the user to >>> register again, deregistering the oldest contact. >>> >>> I saw that there is a module named "ims_usrloc_scscf" and on its >>> description it says "implemented overwrite oldest contact behaviour", >>> but I couldn't find any documentation about it. >>> >>> So, is there a way to get this working today? >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> >> >> >

3 2

kamailio proxy
by liwei＠neulinx.com 03 Apr '16

03 Apr '16

hi ， I want to use kamailio as a proxy ; / gateway1 ----->pstn / 1、 call from backend server ---> kamailio ----->gateway2 -----> pstn \ \ gateway3 ------->pstn 2、caller and callee number translation ; Could you please give me some suggest; Wade From China liwei(a)neulinx.com

2 1

SQlite DB Lock with Presence
by Kristian Kielhofner 02 Apr '16

02 Apr '16

Hello everyone, I'm trying to use BLF with the config here (except with SQlite): http://kb.asipto.com/kamailio:presence:k43-blf BLF seems to work perfectly except after the call is ended I get "sqlite commit failed: database is locked" error messages as seen here: http://pastebin.com/xVijj98H BLF is then broken at this point. kamailio -v version: kamailio 4.3.5 (x86_64/linux) 950657 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 950657 compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 Has anyone ever seen this before? Thanks! -- Kristian Kielhofner

3 9

MSRP question
by Hussein Said 02 Apr '16

02 Apr '16

2 4

kamailio proxy
by liwei＠neulinx.com 02 Apr '16

02 Apr '16

hi ， I want to use kamailio as a proxy ; / gateway1 ----->pstn / 1、 call from backend server ---> kamailio ----->gateway2 -----> pstn \ \ gateway3 ------->pstn 2、caller and callee number translation ; Could you please give me some suggest; Wade From China liwei(a)neulinx.com

1 0

Trouble initializing TLS on Fedora 23
by Anthony Messina 02 Apr '16

02 Apr '16

Happy New Year! I've just upgraded my Kamailio (build from master@c7e411e) instance to Fedora 23 from Fedora 22. I've built the packages specifically for Fedora 23 with the following current sources: openssl-1.0.2e-3.fc23.x86_64 openssl-libs-1.0.2e-3.fc23.x86_64 openssl version reports... OpenSSL 1.0.2e-fips 3 Dec 2015 Even so, the following error occurs. It seems like Kamailio having trouble detecting that I'm using running with the same version that I have installed, and the same version that I have compiled against. tls [tls_init.c:557]: init_tls_h(): ERROR: tls: init_tls_h: installed openssl library version is too different from the library the Kamailio tls module was compiled with: installed "OpenSSL 1.0.0-fips 29 Mar 2010" (0x10000003), compiled "OpenSSL 1.0.2d-fips 9 Jul 2015" (0x1000204f). Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) CRITICAL: <core> [main.c:2558]: main(): could not initialize tls, exiting... Any pointers? -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

3 7

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 01 Apr '16

01 Apr '16

Hello, *Olle!* *Thanks for your help in this case, and don't worry about the time (actually I wasn't even expecting to get an answer today).* *I know the max_contacts parameter. But the problem is that if I set the max_contacts to 2, when the user tries to register again, he gets an error message 503. The point is that I want to get the user registered, and replace the oldest location entry for the newest one.* *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all. > > I have problem here and I tried to find a solution and search over > internet, but without success. > > My scenario is the following: I have an application that must be > registered in Kamailio when a user logs in the web interface, so he can get > calls (something like a web softphone using webrtc). At this point, we are > OK, and everything is working fine. > > To get these register functions I am using the parameter "save("location", > "0x04")" as described in the REGISTRAR module documentation, so if a user > logs in another web browser or computer, only the last one will continue > registered and all calls will be forwarded to him. > > The point is that now we are creating a phone app that will do almost the > same thing as the web interface, allowing users to receive calls using the > mobile device, and here is my problem: I want to allow my users to be > registered on two devices at the same time, but if a user logs into another > device, I don't want to reply with a 503. I want to allow the user to > register again, deregistering the oldest contact. > > I saw that there is a module named "ims_usrloc_scscf" and on its > description it says "implemented overwrite oldest contact behaviour", but > I couldn't find any documentation about it. > > So, is there a way to get this working today? > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com >

2 3

Multiple Registrations - Overwrite oldest contact
by Bruno Emer 01 Apr '16

01 Apr '16

Hello all. I have problem here and I tried to find a solution and search over internet, but without success. My scenario is the following: I have an application that must be registered in Kamailio when a user logs in the web interface, so he can get calls (something like a web softphone using webrtc). At this point, we are OK, and everything is working fine. To get these register functions I am using the parameter "save("location", "0x04")" as described in the REGISTRAR module documentation, so if a user logs in another web browser or computer, only the last one will continue registered and all calls will be forwarded to him. The point is that now we are creating a phone app that will do almost the same thing as the web interface, allowing users to receive calls using the mobile device, and here is my problem: I want to allow my users to be registered on two devices at the same time, but if a user logs into another device, I don't want to reply with a 503. I want to allow the user to register again, deregistering the oldest contact. I saw that there is a module named "ims_usrloc_scscf" and on its description it says "implemented overwrite oldest contact behaviour", but I couldn't find any documentation about it. So, is there a way to get this working today? *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com

2 2

Make Kamailio Great Again!
by Alex Balashov 01 Apr '16

01 Apr '16

For immediate release: ATLANTA, GA (1 April 2016)--Alex J. Balashov, a self-styled businessman based in Atlanta, Georgia, USA, has a plan to "Make Kamailio Great Again". "Evariste Systems is huge. My name is on the building," said Balashov of his iconic VoIP consulting brand. "And you know what, I have been very successful. Everybody loves me." Balashov has capitalised on a contentious election cycle marked by deep political polarisation, growing income inequality and geopolitical challenges such as global terrorism. And his sharp message of alarm about the declining influence of the Kamailio SIP server project has resonated with increasing numbers in the CxO suite, vaulting him to the lead in the race for the IETF SIP Working Group nomination, according to recent polls of primary voters. He has been quick to tout his competitive credentials in a tough global open-source ecosystem. At a recent colloqium on unified communications, he asked: "When was the last time anybody saw us beating, let's say, OpenSIPS in Git commits? They kill us. I beat OpenSIPS all the time. All the time." As Balashov sees it, a major cause of the beleaguered Kamailio project's woes lies in its liberal patch acceptance policy and lax scrutiny of third-party contributions: "When GitHub sends its people, they're not sending their best. They're not sending you. They're not sending you. They're sending people that have lots of problems, and they're bringing those problems. They're bringing drugs. They're bringing crime. They're rapists. And some, I assume, are good people." He has proposed a controversial solution that has drawn ire from liberal ranks in the open-source community, but has also attracted applause and standing ovations at his speaking engagements: "We have to have a firewall around the Kamailio source code. We have to have an access control list. And in that firewall, we're going to have a big fat door where commits and pull requests can come into the master branch, but they have to come in legally. The firewall will go up, and GitHub will start behaving." Balashov's firewall proposal has been met with scorn from critics who deride it as impractical and quixotic. In particular, commentators have raised questions about funding and resources as well as GitHub's willingness to entertain a boundary around a project in its vicinity. Balashov isn't concerned, however: "I will build a great firewall--and nobody builds firewalls better than me, believe me--and I'll build them very inexpensively. I will build a great, great stateful packet inspection wall on our border with GitHub, and I will make GitHub pay for that wall. Mark my words." He has also been rebuked by rival IETF leadership candidates for his often acerbic Twitter remarks directed at Lennart Poettering and the developers of "firewalld". As he sees it, however, the network effects of social media are a strength: "My Twitter has become so powerful that I can actually make my enemies tell the truth." He scoffed at the suggestion that his characterisations of industry actors behind the RedHat-led "systemd" movement are misleading: "RedHat was the worst Steward of Linux in the history of the kernel. There has never been a Steward so bad as RedHat. The source code blew up around us. We lost everything, including all synergies. There wasn't one good thing that came out of that administration or them being Stewards of Linux." Balashov's idiosyncratic campaign is not standing still. He has proven to be a capable populist, adapting rapidly to an evolving sense of the kinds of pronouncements that activate his swelling crowds of devotees. Along the way, he has deftly deflected calls to subject his policy proposals to expert review. "I know what I'm doing, and I listen to a lot of people, I talk to a lot of people, and at the appropriate time I'll tell you who the people are. But I speak to a lot of people, but my primary consultant is myself, and I have a good instinct for this stuff." At a recent gathering of SIP stack interoperability specialists, Balashov the latest pillar of his platform to "Make Kamailio Great Again", in view of growing security vulnerabilities in the latest Kamailio modules: "Alex J. Balashov is calling for a total and complete shutdown of commits entering the master branch from the territory of the European Union until our project's representatives can figure out what's going on. According to Netcraft, among others, there are a lot of buffer overflows in Kamailio by large segments of the EU population."

2 1

Acc INVITE dialog start timing difference
by Dirk Teurlings - Signet B.V. 01 Apr '16

01 Apr '16

Hi, I'm seeing a difference in timestamp for acc modules recorderd INVITES and the dialog start event. This is the log entry: Apr 1 12:55:59 host /usr/sbin/kamailio[15642]: INFO: <script>: 5d3ff3407b9761b24c4f75195c6ccb1d@192.168.10.245:5060 - dialog:start And this has been added by ACC in the database: INVITE as2880319e 56FE53B9-49E3774-3E86ED1B 5d3ff3407b9761b24c4f75195c6ccb1d@192.168.10.245:50... 200 OK 2016-04-01 12:56:01 Why are these two seconds apart? Shouldn't the acc module record the same dialog start event time as the start of a dialog? When I look at the value of DLG_lifetime this confirms my findings, as it results in: dialog_time = TS - dlg_start When looking at the database BYE-time - INVITE time, the call is two seconds shorter. Any insight would be appreciated! Cheers, Dirk

1 0

Kamailio start up issue
by NITESH BANSAL 01 Apr '16

01 Apr '16

Hello, I installed Kamailio 4.3 on debian wheezy machine using debian package from https://www.kamailio.org/wiki/packages/debsThe installation went fine, but I'm unable to start kamailio. It seems that Kamailio is having issue locating the issues.Kamailio installed all the modules in location '/usr/lib/x86_64-linux-gnu/kamailio/modules', when building from source, it usedto copy modules in '/usr/lib/kamailio/modules'.Any suggestions on how can I fix it? Thanks,Nitesh

1 1

Framework/Lib to create SIP packets
by Igor Olhovskiy 01 Apr '16

01 Apr '16

Hi, all Not a Kamailio-related question, but. I want to form SIP-packets with info I need and receive answers. At first iteration in would be PUBLISH (with XML info) messages (PUBLISH - 200 OK) sent to Kamailio. I want to use Python (more preferable) or Lua (less preferable). Is there any lib I can create messages with, or forget and just use plain text + UDP socket? -- Best regards, Igor

5 13

HOW send 4XX to the caller when the callee response to bye is 481 Call leg/transaction does not exist
by Humberto H 01 Apr '16

01 Apr '16

Hello; [image: Inline image 1] HOW can I send 4XX to the caller when the callee response to a BYE message is: *481 Call leg/transaction does not exist. BUT the caller is waiting.* Before this message. I drop the bye sent by the callee, my purpose here is "*Do not send the BYE TO THE CALLER"* JUST send a message in order to the caller do failover, If I send *BYE* to the caller it can't do failover Thanks in advance. 10.929596| | BYE | | |SIP Request BYE CSeq:104 | | |(5060) ------------------> (5060) | | |10.929981| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist | | |(5060) <------------------ (5060) | | |10.929983| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist Call leg/transaction does not exist | | |(5060) ------------------> (5060) | | |10.930445| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist | | |(5060) ------------------> (5060) | | |10.930451| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist

2 1

How can I send 404 to the caller instead of BYE?
by Humberto H 01 Apr '16

01 Apr '16

Hello I am trying to send 404 to the caller instead of BYE. Using the configuration showned bellow. when the callee send BYE Kamailio sends -- *404* to the callee (to the source of the BYE signaling) --- and *BYE* to the caller But I need to send 404 response to the caller only Thanks in advance event_route[dialog:end] { if(is_method("BYE") && ((src_ip==XXXXXXX) )){ send_reply("404", "Not found"); xlog("L_INFO","ENVIO 404 ON BYE \n"); } exit; }

2 1

stateful messages in a dialog
by Al S 01 Apr '16

01 Apr '16

Hi, I would like to know how I can make subsequent messages in a dialog (ACK and BYE) to be stateful.Your help will be greatly appreciated. // reroute INVITE after timeout.failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} Thanks,AS

2 1

Record-Route header
by jignesh gandhi 01 Apr '16

01 Apr '16

Hello All, I have a need to save the Record-Route header from the INVITE and use it as a route header in requests like BYE. PROXY --> Kamailio(as PROXY) --> Asterisk I am using rr module to add a record_route when passing the INVITE to Asterisk. Thanks, Jignesh Gandhi

2 4

Re: [SR-Users] Proxy configuration
by Jignesh Gandhi 31 Mar '16

31 Mar '16

Thanks Alex for you reply. I understand the dispatcher module can be used to load balance inbound requests. But my questions is more related with outbound requests thru Kamailio. Are you saying that in-dialog request can be routed automatically by Kamailio to the correct Asterisk instance on the back end ? And this is included in the stock config ? I was just looking for an example. Thanks, --Jignesh On 3/30/16, 6:00 AM, "sr-users on behalf of sr-users-request(a)lists.sip-router.org" <sr-users-bounces(a)lists.sip-router.org on behalf of sr-users-request(a)lists.sip-router.org> wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Destination Timeout (ACK message) (Al S) > 2. Proxy configuration (Jignesh Gandhi) > 3. Re: Proxy configuration (Alex Balashov) > 4. Local Kamailio share a hosted Kamailio/Asterisk sip server > (S?bastien Brice) > 5. SQlite DB Lock with Presence (Kristian Kielhofner) > 6. Kamailio local and hosted (S?bastien Brice) > 7. Re: Releasing Kamailio v4.4.0 (Daniel-Constantin Mierla) > 8. Re: SQlite DB Lock with Presence (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Tue, 29 Mar 2016 09:27:56 -0400 >From: Al S <alirsm(a)outlook.com> >To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Destination Timeout (ACK message) >Message-ID: <BAY177-W1513432E7A3AFDAAB1DE8DD4870(a)phx.gbl> >Content-Type: text/plain; charset="iso-8859-1" > >Hi, >The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below). > >failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} >However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. >Thank, >AS >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/b91e2df…> > >------------------------------ > >Message: 2 >Date: Tue, 29 Mar 2016 14:24:01 +0000 >From: Jignesh Gandhi <Jignesh.Gandhi(a)MoviusCorp.com> >To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Proxy configuration >Message-ID: <011222F3-5820-41FB-A0D8-8D727EA121FA(a)moviuscorp.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Is it possible to use kamailio to be a stateful outbound proxy ? > >The use case is kamailio is an outbound proxy for multiple instances of Asterisk. >Thus all responses need to go back to the same instance of Asterisk that generated them. > >After the call is established. >How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ? > >Any help is appreciated. > >Thanks, >--Jignesh >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/5081718…> > >------------------------------ > >Message: 3 >Date: Tue, 29 Mar 2016 10:29:19 -0400 >From: Alex Balashov <abalashov(a)evaristesys.com> >To: sr-users(a)lists.sip-router.org >Subject: Re: [SR-Users] Proxy configuration >Message-ID: <56FA913F.6080903(a)evaristesys.com> >Content-Type: text/plain; charset=windows-1252; format=flowed > >Jignesh, > >Everything you're asking for is part of Kamailio's core functionality: > >1. Most useful applications of Kamailio involve stateful relay, and the >stock included configurations are stateful. > >2. The loose-routing section of the configuration, and, more broadly, >RFC 3261 in general, take care of routing in-dialog requests (ACKs, >reinvites, BYEs, etc.) to the appropriate dialog party. > >3. Load-balancing/distribution among Asterisk instances can be done with >the 'dispatcher' module; there are many examples floating around, as >it's one of the most common applications for Kamailio. > >-- Alex > >-- >Alex Balashov | Principal | Evariste Systems LLC >1447 Peachtree Street NE, Suite 700 >Atlanta, GA 30309 >United States > >Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) >Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > > >------------------------------ > >Message: 4 >Date: Tue, 29 Mar 2016 16:32:36 +0200 (CEST) >From: S?bastien Brice <sebastien.brice(a)jvs.fr> >To: sr-users <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Local Kamailio share a hosted Kamailio/Asterisk > sip server >Message-ID: <2011796741.12246629.1459261956126.JavaMail.zimbra(a)jvs.fr> >Content-Type: text/plain; charset=utf-8 > > >Hi everyone >At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. >As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. > >ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) > >when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). >to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. > >This works well, but i am wondering if that makes sense if i get several phones behind B. >Should stateless forward works better in this kind of scenario? > >In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality >Is that possible ? >Do you have some sort of example Kamailio do it good On-premise/localy ? >How to deal with nat since all hardphones have rfc1918?address? > >thx you > >PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. > > > >S?bastien BRICE VoIP, Support et Int?gration > > > >------------------------------ > >Message: 5 >Date: Tue, 29 Mar 2016 20:50:14 -0400 >From: Kristian Kielhofner <kris(a)kriskinc.com> >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] SQlite DB Lock with Presence >Message-ID: > <CAKDfjge4kLhdLjV5V7gsOOsLsdByeLgtwnqE-EEXHhxbVeV8Ag(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >Hello everyone, > > I'm trying to use BLF with the config here (except with SQlite): > >http://kb.asipto.com/kamailio:presence:k43-blf > > BLF seems to work perfectly except after the call is ended I get >"sqlite commit failed: database is locked" error messages as seen >here: > >http://pastebin.com/xVijj98H > >BLF is then broken at this point. > >kamailio -v >version: kamailio 4.3.5 (x86_64/linux) 950657 >flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, >USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, >SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, >FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, USE_DNS_CACHE, >USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES >ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB >poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >id: 950657 >compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 > >Has anyone ever seen this before? > >Thanks! > >-- >Kristian Kielhofner > > > >------------------------------ > >Message: 6 >Date: Tue, 29 Mar 2016 14:38:40 +0200 (CEST) >From: S?bastien Brice <sebastien.brice(a)jvs.fr> >To: sr-users <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Kamailio local and hosted >Message-ID: <1194659979.12170808.1459255120868.JavaMail.zimbra(a)jvs.fr> >Content-Type: text/plain; charset=utf-8 > >Hi everyone >At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. >As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. > >ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) > >when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). >to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. > >This works well, but i am wondering if that makes sense if i get several phones behind B. >Should stateless forward works better in this kind of scenario? > >In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality >Is that possible ? >Do you have some sort of example Kamailio do it good On-premise/localy ? >How to deal with nat since all hardphones have rfc1918?address? >, >thx you > >PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. > > > >S?bastien BRICE VoIP, Support et Int?gration > > > >------------------------------ > >Message: 7 >Date: Wed, 30 Mar 2016 10:02:02 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org>, sr-dev <sr-dev(a)lists.sip-router.org>, > "business(a)lists.kamailio.org" <business(a)lists.kamailio.org> >Subject: Re: [SR-Users] Releasing Kamailio v4.4.0 >Message-ID: <56FB87FA.8080304(a)gmail.com> >Content-Type: text/plain; charset=utf-8 > >Hello, > >the final packaging for 4.4.0 is expected to start after 14:00UTC today. >Backports to branch 4.4 can be done up to 12:00UTC, after that let's >sync on sr-dev mailing list, because I will start testing how it builds >on different platforms and it is good to have the a consistent source code. > >Cheers, >Daniel > >On 29/03/16 09:04, Daniel-Constantin Mierla wrote: >> Hello, >> >> refreshing about the major release 4.4.0 planned for tomorrow -- if any >> dev has patches to backport to branch 4.4 should do it as soon as possible. >> >> Cheers, >> Daniel >> >> On 18/03/16 12:16, Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> with no major issues reported lately, I think we are on a good track for >>> releasing stable v4.4.0. Next week a lot of people prepare for Easter, >>> therefore I propose to do the release on Wednesday, March 30, 2016. If >>> new things pop up, the exact date can be adjusted a bit before or after >>> the current proposal. >>> >>> Any new discovered issue should be reported to bug trucker: >>> >>> * https://github.com/kamailio/kamailio/issues >>> >>> Help to complete the upgrade guide would be very appreciated: >>> >>> * https://www.kamailio.org/wiki/install/upgrade/stable-to-devel >>> >>> A draft list of what is new in 4.4 is already available at: >>> >>> * https://www.kamailio.org/wiki/features/new-in-4.4.x >>> >>> Cheers, >>> Daniel >>> > >-- >Daniel-Constantin Mierla >http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > > >------------------------------ > >Message: 8 >Date: Wed, 30 Mar 2016 11:37:39 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> >Subject: Re: [SR-Users] SQlite DB Lock with Presence >Message-ID: <56FB9E63.7040104(a)gmail.com> >Content-Type: text/plain; charset=windows-1252 > >Hello, > >not using sqlite here, but could be related to presence modules doing db >transactions and not releasing the locks. Do you get the issue after the >first call, or it runs for a while and at some point it breaks? > >Cheers, >Daniel > >On 30/03/16 02:50, Kristian Kielhofner wrote: >> Hello everyone, >> >> I'm trying to use BLF with the config here (except with SQlite): >> >> http://kb.asipto.com/kamailio:presence:k43-blf >> >> BLF seems to work perfectly except after the call is ended I get >> "sqlite commit failed: database is locked" error messages as seen >> here: >> >> http://pastebin.com/xVijj98H >> >> BLF is then broken at this point. >> >> kamailio -v >> version: kamailio 4.3.5 (x86_64/linux) 950657 >> flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, >> USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, >> SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, >> FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, USE_DNS_CACHE, >> USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> id: 950657 >> compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 >> >> Has anyone ever seen this before? >> >> Thanks! >> > >-- >Daniel-Constantin Mierla >http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > > >------------------------------ > >Subject: Digest Footer > >_______________________________________________ >sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >------------------------------ > >End of sr-users Digest, Vol 130, Issue 29 >*****************************************

2 1

timeout function
by Al S 31 Mar '16

31 Mar '16

Hi, I am routing call traffic thru proxy to destination A. If destination A is down, we need to reroute to destination B. What function we can use to process timeout so we can reroute the calls to different node? Thanks,ARM

2 1

TCP connection select problem
by 张顺通 31 Mar '16

31 Mar '16

my program structure is LinPhone------TCP------KA_EDGE------UDP------KA_REGISTER------UDP------FS KA_EDGE:all clients TCP connection is in this server KS_REGSITER:all register info is process here when User A want to use program,Linphone send SIP REGISTER to KA_EDGE by TCP(and TCP is always present), and KA_EDGE send to KA_REGISTER by UDP. when FS want to Call A, FS generate SIP: INVITE A@KA_REGISTER_IP KA_REGISTER recive INVITE, lookup the register list, send to KA_EDGE INVITE TCP_IP:TCP_PORT Route:KA_EDGE_IP:KA_EDGE_PORT and LinPhone recive that INVITE. *my question is:* LinPhone A and LinPhone B both register in KA_REGISTER. FS send INVITE: INVITE B: KA_REGISTER_IP and KA_REGISTER lookup correct，send to KA_EDGE INVITE TCP_IP_B:TCP_PORT_B Route:KA_EDGE_IP:KA_EDGE_PORT but KA_EDGE is not correct.sip is correct,* but KA_EDGE send the INVITE to A' TCP connnection* INVITE TCP_IP_B:TCP_PORT_B Route:KA_EDGE_IP:KA_EDGE_PORT Someone had this problem?Is this a bug?

4 19

db_cassandra status and plans
by Andreas Granig 31 Mar '16

31 Mar '16

Hi all, I was checking db_cassandra with Kamailio 4.3 recently to try making it work as a usrloc back-end, and my observation is that the module has quite some issues. In general, the module states that it's only tested with cassandra 1.0.6, which is really out-dated. I tried with the latest version of the 1.0 series (1.0.12) and even with lots of fiddling I didn't manage to make it work, could be my fault though. There has not been any work for 4 years other than cosmetical documentation fixes two years ago. Documentation is missing important information how to set up the version table, that could be fixed though. The schema definition in the documentation is missing new columns which have been introduced, that could be fixed as well. The module uses the legacy thrift interface, whereas nowadays (since cassandra 1.2+) a native cql driver is available as open-source. The module can not be used together with nathelper for nat pings, since the queries don't provide a key, which is required for cassandra. Our plan is to revamp the whole module and make it work with recent cassandra versions and remove the nat ping limitations. Main question is whether to go for a new module or replacing the old one, since it doesn't seem to work anymore anyways. Feedback whether you got it working with Kamailio 4.3+ and your views on future plans is much appreciated! Andreas

3 5

acc_cdrs empty
by Nelson Migliaro 31 Mar '16

31 Mar '16

Hello, I am not able to set up cdr. I see acc table is updated with 2 new entries after each call (INVITE and BYE) but acc_cdrs is empty. There are errors in the log. I think that I set up everything but is not working, - load the module, set module params and set the flag? Am I missing something, should I use a store procedure or something? ------------------------------------------------------------------------------------------------------------ loadmodule "acc.so" modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } route[WITHINDLG] { if (has_totag()) { if (loose_route()) { route(DLGURI); if (is_method("BYE")) { CREATE TABLE `acc_cdrs` ( `id` int(10) unsigned NOT NULL AUTO_INCREMENT, `start_time` varchar(32) NOT NULL DEFAULT '', `end_time` varchar(32) NOT NULL DEFAULT '', `duration` varchar(32) NOT NULL DEFAULT '', PRIMARY KEY (`id`), KEY `start_time_idx` (`start_time`) ) ENGINE=MyISAM AUTO_INCREMENT=2 DEFAULT CHARSET=latin1$$ Thank you.

1 1

UAC Register Asterisk
by Alberto Sagredo 31 Mar '16

31 Mar '16

Hi Im trying to modify example for Asterisk/Kamailio integration that miconda did, and trying to handle Forwarded register to Asterisk in an authenticated way instead removing secret in asterisk. I have made this changes route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $var(astip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $var(astport); $uac_req(furi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(turi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; xlog("Dentro de REGFWD"); t_on_failure("REG_AST"); uac_req_send(); } failure_route[REG_AST] { if (t_is_canceled()) { exit; } xlog("Dentro de REG AST"); if(t_check_status("401|407")) { xlog("Dentro de 401"); $avp(auser) = "test"; $avp(apass) = "test"; uac_auth(); t_relay(); exit; } } What i see on traces: U x.x.x..240:5060 -> x.x.x..56:5060 REGISTER sip:x.x.x.56:5060 SIP/2.0. Via: SIP/2.0/UDP x.x.x.240;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0. To: <sip:200@x.x.x.56>. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. CSeq: 10 REGISTER. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. Max-Forwards: 70. Content-Length: 0. User-Agent: ANY. Contact: <sip:emp_001_200@x.x.x.240:5060>. Expires: 3600. . # U x.x.x.56:5060 -> x.x.x..240:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP x.x.x..240 ;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0;received=x.x.x..240 ;rport=5060. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. To: <sip:200@x.x.x.56>;tag=as62b47138. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. CSeq: 10 REGISTER. Server: FPBX-2.11.0(11.17.1). Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59e39692". Content-Length: 0. . But Kamailio is not sending authentication . Any idea what could be wrong? Or any tip to do what i try. The idea is that a sip endpoint register with authentication on kamailio and it sends register with authentication to asterisk. Thanks in advance

4 7

Kamailio v4.4.0 Released - new major version is out
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Kamailio v4.4.0 is out – it comes with 9 new modules and a consistent set of improvements touching more than 60 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.4.0! The development will start to focus on 5.0.0 - join the discussion on sr-users mailing list to design and decide what should Kamailio 5.0.0 look like - some notes are already collected at: * https://www.kamailio.org/wiki/devel/kamailio-5.0-design Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

Releasing Kamailio v4.4.0
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Hello, with no major issues reported lately, I think we are on a good track for releasing stable v4.4.0. Next week a lot of people prepare for Easter, therefore I propose to do the release on Wednesday, March 30, 2016. If new things pop up, the exact date can be adjusted a bit before or after the current proposal. Any new discovered issue should be reported to bug trucker: * https://github.com/kamailio/kamailio/issues Help to complete the upgrade guide would be very appreciated: * https://www.kamailio.org/wiki/install/upgrade/stable-to-devel A draft list of what is new in 4.4 is already available at: * https://www.kamailio.org/wiki/features/new-in-4.4.x Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

Kamailio local and hosted
by Sébastien Brice 29 Mar '16

29 Mar '16

Hi everyone At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). to route INVITE from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. This works well, but i am wondering if that makes sense if i get several phones behind B. Should stateless forward works better in this kind of scenario? In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality Is that possible ? Do you have some sort of example Kamailio do it good On-premise/localy ? How to deal with nat since all hardphones have rfc1918 address? , thx you PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. Sébastien BRICE VoIP, Support et Intégration

1 1

Proxy configuration
by Jignesh Gandhi 29 Mar '16

29 Mar '16

Hello, Is it possible to use kamailio to be a stateful outbound proxy ? The use case is kamailio is an outbound proxy for multiple instances of Asterisk. Thus all responses need to go back to the same instance of Asterisk that generated them. After the call is established. How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ? Any help is appreciated. Thanks, --Jignesh

2 1

Destination Timeout (ACK message)
by Al S 29 Mar '16

29 Mar '16

Hi, The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below). failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. Thank, AS

1 0

SIP / WebRTC / GSoC proposals
by Daniel Pocock 29 Mar '16

29 Mar '16

Hi all, We had a few GSoC applications[1] from students who would like to do work relating to SIP and WebRTC this summer. The students have a range of skills. Some are interested in the server side (C, C++) code. Some are interested in Android and Java projects (e.g. the Lumicall and CSipSimple dialers) and some are web developers, interested in making practical use of WebRTC. Would anybody else from the community be interested in helping to mentor one of these projects? Prospective mentors can review the proposals privately before making any firm commitment, please contact me for details. Mentors can have some input in the direction of the students' work, just as long as all code is released publicly. Regards, Daniel 1. http://danielpocock.com/gsoc-2016-voice-video-chat

2 2

XML-RPC reply doesn't contain any data
by Arsen 29 Mar '16

29 Mar '16

Hi guys! We have an issue with XML-RPC module. >From this doc: http://kamailio.org/docs/modules/4.3.x/modules/xmlrpc.html "Success replies always contain at least one return value. In our case the simplest success replies contain single boolean with value 1" We are trying to reload domain module after adding/removing new domain, so we send the following request: XX - web server; YY - kamailio T XX.XX.XX.XX:55028 -> YY.YY.YY.YY:8080 [AP] POST /RPC2 HTTP/1.1. Host: YY.YY.YY.YY:8080. Accept: */*. Accept-Encoding: deflate, gzip. User-Agent: cURL. Accept-Charset: UTF-8. Content-Length: 120. Content-Type: application/x-www-form-urlencoded. . <?xml version="1.0" encoding="iso-8859-1"?> <methodCall> <methodName>domain.reload</methodName> <params/> </methodCall> T YY.YY.YY.YY:8080 -> XX.XX.XX.XX:55028 [AP] HTTP/1.1 200 OK. Sia: SIP/2.0/TCP XX.XX.XX.XX:55028. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 108. . <?xml version="1.0"?> <methodResponse> <params> <param> <value></value> </param> </params> </methodResponse> As you can see in the response from the kamailio there is no any data, so our script is waiting until timeout... Please advice. Thanks, -- Regards, Arsen.

4 6

Timeout Handling (update)
by Al S 28 Mar '16

28 Mar '16

Hi, I am trying to re-route to a different $ru after timeout and right before Kamailio reply with a 408.I made the changes right before sl_reply_error(), however it does not work as I expected. Any help will be appreciated. Also, $rm represents the request message received by Kamailio. Is there a variable represents the messages leaving Kamailio?. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { // re-route to a new $ru before sending 408 sl_reply_error(); } exit;} Thanks,AS

2 2

using DMQ , Kamailio shutsdown
by Sunil More 28 Mar '16

28 Mar '16

Hello All, I am running two instances of kamailio and using dmq. Kamailio starts good. However one of the node shutsdown unexpectedly. Unsure what is going wrong. Please let me know what could be wrong. below is a the conf detail. loadmodule "dmq.so" modparam("dmq", "server_address", "sip:sunilmore.in:5060") modparam("dmq", "notification_address", "sip:sunilmore.in:5064") loadmodule "usrloc.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", DBURL2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "nat_bflag", FLB_NATB) modparam("usrloc", "db_insert_null", 1) loadmodule "dmq_usrloc.so" modparam("dmq_usrloc", "enable", 0) Below is the backtrace of the logs. ar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: kex Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (11) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [srid-56f4d496-36d7-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: mi_rpc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: ndb_redis Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14013]: DEBUG: <core> [main.c:1723]: main_loop(): Expect maximum 2177 open fds Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: htable [htable.c:237]: child_init(): rank is (12) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tm Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: tm [callid.c:134]: child_init_callid(): DEBUG: callid: ' 0e4b331663a1f7ae-14041(a)sunilmore.in' Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tmx Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: tmx [tmx_mod.c:272]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: sl Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tls Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: ndb_mongodb Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dmq Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: usrloc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [ulcx-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa762c8 and 0xaba2c8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dmq_usrloc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa76100, 40, 1, (nil)), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: htable [htable.c:237]: child_init(): rank is (11) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: dmq_usrloc [dmq_usrloc.c:110]: child_init(): child_init PROC_INIT Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: registrar Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [uloc-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: nathelper Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: nat_traversal Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_fifo Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: avpops Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: rtimer Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dialog Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: domain Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: uac Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: presence Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: kex Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [srid-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_rpc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: ndb_redis Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: htable [htable.c:237]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa762c8 and 0xaba2c8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa76100, 38, 1, (nil)), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa32028 and 0xa76028 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 8, 1, 0x7fab28a78438), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 9, 1, 0x7fab28a786a8), fd_no=1 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 10, 4, 0x7fab082bb21c), fd_no=2 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 11, 4, 0x7fab082bb2a8), fd_no=3 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 12, 4, 0x7fab082bb334), fd_no=4 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 13, 4, 0x7fab082bb3c0), fd_no=5 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 14, 4, 0x7fab082bb44c), fd_no=6 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 15, 4, 0x7fab082bb4d8), fd_no=7 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 16, 4, 0x7fab082bb564), fd_no=8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 17, 4, 0x7fab082bb5f0), fd_no=9 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 18, 4, 0x7fab082bb67c), fd_no=10 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 19, 4, 0x7fab082bb708), fd_no=11 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 20, 4, 0x7fab082bb794), fd_no=12 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 21, 4, 0x7fab082bb938), fd_no=13 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 22, 4, 0x7fab082bb9c4), fd_no=14 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 23, 4, 0x7fab082bba50), fd_no=15 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 24, 4, 0x7fab082bbadc), fd_no=16 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 27, 4, 0x7fab082bbb68), fd_no=17 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 25, 4, 0x7fab082bbbf4), fd_no=18 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 26, 4, 0x7fab082bbc80), fd_no=19 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 29, 4, 0x7fab082bbd0c), fd_no=20 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 31, 4, 0x7fab082bbd98), fd_no=21 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 28, 3, 0x7fab28d647f0), fd_no=22 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 30, 3, 0x7fab28d64810), fd_no=23 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 36, 3, 0x7fab28d64830), fd_no=24 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 39, 3, 0x7fab28d64850), fd_no=25 Mar 25 06:03:03 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:03 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:04 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:04 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:05 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:05 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:06 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:06 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14027]: DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fab0853a9d8 from timer.c:648 Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:08 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:08 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:09 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:09 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 21 Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [tcp_main.c:3448]: handle_ser_child(): dead child 14, pid 14031 (shutting down?) Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa31e60, 21, -1, 0x0) fd_no=26 called Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT: <core> [main.c:739]: handle_sigs(): child process 14031 exited by a signal 11 Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT: <core> [main.c:742]: handle_sigs(): core was generated Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: DEBUG: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14038]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14040]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14037]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14035]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received There were no UAs that tried to register while this error occured. Thanking You, Sunil More Ph : 9503338275

2 4

Kamailio Dispatcher error msg 407
by Rex Lin (林昱頡) 28 Mar '16

28 Mar '16

Hi admins, I'm kinda new in this group and I'm not quite sure if I'm posting in a correct way, advise if I'm wrong. below is my post. ------------------------------------------------------------------------------------------------------------------------------------ I would like to activate Dispatcher module for apply Load Balance functionality purpose. but it's been hard to complete so far, I wonder if you can give me some advise in below situation. we have 2 Kamailios, the version is 4.3.3 ; we use domain for the SIP registration. we want to make Kamailios work in double active mode for SIP load balancing, we have a F5 to do the domain name round robin to make two UAC regitering under two UAS. I have put the "route(DISPATCH) b4 PSTN in request_route. when I try to make UAC-A call to UAC-B, it appears 407 proxy authentication required. we wiresharked, UAS has sent INVITE to UAS-B, but UAS-B sent the error code 407 to UAS-A Is the configuration has to be set to "UAS-B accepting INVITE of UAS-A" ? or the UAS-A should carrying the auth info into UAS-B ? detail configuration as below. ------------------------------------------------------------------------------------------------------------------------------------ dispatcher params modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 1) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_ping_from", "sip:kamailio1@awesomedomain.com") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) DISPATCHER Routing route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Re-route in case of failure failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next node - only for 500 or local timeout if (t_check_status("500") || (t_branch_timeout() && !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } Best Regards, Rex Lin

2 3

Custom Authentication module with Java Code
by Jungtae Kim 28 Mar '16

28 Mar '16

Hello, I'm new to Kamailio. I'm trying to make my own auth module to check if incoming SIP messages are sighed with a correct key. I want to do this by calling a Java code snippet. I found app_java module but I can't figure out if this will support my usecase. Does app_java allow me to call Java code from another module? If not, what can I do to meet my use case? My token verification code is written in Java, so I'd like to reuse my code if possible. Thanks

2 1

Kamailio return response
by Shiv Patidar 28 Mar '16

28 Mar '16

i run client socket program for send SIP request to kamailio server kamailio registration is done successfully with 200OK and i want to store 200OK in client.c program in structure like struct reg_db { char *to char *from char ..... ....... } please help me how to store ?

2 1

Two different Database at the same time
by Barış Şekerciler 28 Mar '16

28 Mar '16

Hello,I want to use Cassandra and PGSQL at the same time in Kamailio. Is that possible?For example, for CDR records I want to use Cassandra and for other things I want to use PostgreSQL. Regards.

2 4

Re: [SR-Users] Kamailio PostgreSQL Asking Password Everytime
by Barış Şekerciler 26 Mar '16

26 Mar '16

Hello Daniel,Thanks for your reply. So I can't see any password for root user option in the kamctlrc file by default.How can I add? Regards.Barış. To: sr-users(a)lists.sip-router.org From: miconda(a)gmail.com Date: Wed, 23 Mar 2016 08:26:58 +0100 Subject: Re: [SR-Users] Kamailio PostgreSQL Asking Password Everytime Hello, I haven't used the postgres in a long while, but I think it should ask only once. Can you try by setting the password inside the kamctlrc file and see if works straight away? Cheers, Daniel On 22/03/16 11:52, Barış Şekerciler wrote: Hello, I have one postgres server at one server and Kamailio at another server. So I believe I did true configuration to the Kamailio. Actually I made some tests and I could create user and authenticated etc. But I have a problem. When I tried to present "kamdbctl create" Kamailio asks password 50 times (almost, much or less idk certainly). After 50 times Db created successfully and works. And it looks like this: root@bs-kamailio:/home/bs-kamailio# kamdbctl create INFO: creating database kamailio ... Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: So how can I solve this situation? I want Kamailio to asks password one time and handle it's job. Regards. _______________________________________________ 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 -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com _______________________________________________ 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

2 3

Kamailio PostgreSQL Asking Password Everytime
by Barış Şekerciler 25 Mar '16

25 Mar '16

Hello,I have one postgres server at one server and Kamailio at another server.So I believe I did true configuration to the Kamailio. Actually I made some tests and I could create user and authenticated etc. But I have a problem. When I tried to present "kamdbctl create" Kamailio asks password 50 times (almost, much or less idk certainly).After 50 times Db created successfully and works. And it looks like this: root@bs-kamailio:/home/bs-kamailio# kamdbctl create INFO: creating database kamailio ... Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: So how can I solve this situation? I want Kamailio to asks password one time and handle it's job. Regards.

3 2

Custom Authentication module with Java code
by Jungtae Kim 25 Mar '16

25 Mar '16

Hello, I'm new to Kamailio. I'm trying to make my own auth module to check if incoming SIP messages are sighed with a correct key. I want to do this by calling a Java code snippet. I found app_java module but I can't figure out if this will support my usecase. Does app_java allow me to call Java code from another module? If not, what can I do to meet my use case? My token verification code is written in Java, so I'd like to reuse my code if possible. Thanks

1 0

Re: [SR-Users] Multiple LCR instances with LCR module
by DanB 25 Mar '16

25 Mar '16

Hey Juha, Many thanks for so quick feedback! Changing the module param did the trick. Have a good one! DanB

1 0

Problem with authdb and view(?)
by Jon Bonilla (Manwe) 25 Mar '16

25 Mar '16

Hi all I've created a "subscriber" table which is a view of a asterisk sippeers table. The view is created as follows: create view subscriber as select id, name as username, fromdomain as domain, secret as password, email as email_address, md5secret as ha1, md5secret as ha1b, NULL as rpid from sippeers where type="friend"; and one of the resulting subscribers looks like this: *************************** 5. row *************************** id: 100022 username: user domain: domain.com password: plain or empty email_address: user(a)domain.com ha1: 3a4f5cc8dc0efb0d899cawe4dec9ea08 ha1b: 3a4f5cc8dc0efb0d899cawe4dec9ea08 rpid: NULL But when I try to register I get ERROR in the logs: New request - M=REGISTER R=sip:domain.com F=sip:user@domain.com ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'md5secret' in 'field list' ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query ERROR: auth_db [authorize.c:187]: get_ha1(): failed to query database Is there any problem with views? Why the logs talks about md5secret column when it should see ha1? cheers, Jon

1 1

Multiple LCR instances with LCR module
by DanB 25 Mar '16

25 Mar '16

Hey Guys, I was wondering if any of you use multiple LCR instances with LCR module since I could see that the documentation allows it, however for some reason the queries to load data (both on restart or kamcmd lcr.reload) somehow hardcode the value of 1 inside: """ select `gw_name`,`ip_addr`,`port`,`uri_scheme`,`transport`,`params`,`hostname`,`strip`,`prefix`,`tag`,`flags`,`defunct`,`id` from `lcr_gw` where `lcr_id`=0 select `gw_name`,`ip_addr`,`port`,`uri_scheme`,`transport`,`params`,`hostname`,`strip`,`prefix`,`tag`,`flags`,`defunct`,`id` from `lcr_gw` where `lcr_id`=1 select `id`,`prefix`,`from_uri`,`stopper`,`enabled`,`request_uri` from `lcr_rule` where `lcr_id`=1 select `rule_id`,`gw_id`,`priority`,`weight` from `lcr_rule_target` where `lcr_id`=1 """ My kamailio version: """ # kamailio -V version: kamailio 4.3.5 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2 """ Many thanks in advance for any kind of tip! DanB

2 1

kamailio memory issue
by Volodymyr Rykhva 25 Mar '16

25 Mar '16

Hi, I have started using kamailio recently. Added to db almost 100K users. And now I am trying to increase shared and private memory with this command when server is running: kamailio -M 12 -m 128. But then I check it with `kamailio -l` and the same values (64 and 8 respectively) are there. Can I change it with config? or what I do wrong. And what's the purpose of child process? - Will kamailio process requests faster if I set child=32 (for example)? Thanks and best regards, Volodymyr

2 1

Re: [SR-Users] htable - banning users
by meres 24 Mar '16

24 Mar '16

Thank u Vasiliy and Daniel, I used -3 condition as you suggested! Regards, meres

1 0

dynamic weighting for dispatcher
by Brooks Bridges 24 Mar '16

24 Mar '16

So I'm building a load balancer solution that needs to monitor remote devices at a specific interval via snmp for certain data (channels and cps) and use that data to create a dynamically updated weight for each of the destinations. I can't find any reasonable mechanism for updating the weight attribute in the dispatcher module itself without having to do some hacky "update the database column then trigger dispatcher.reload" script, which I very much want to avoid. Is this something I should just be thinking of ditching the dispatcher module and rolling my own design or am I missing something obvious? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 2

Dispatching Registrations to Weight based servers
by SamyGo 24 Mar '16

24 Mar '16

Hi All, I'm having a wee little bit of difficulty in trying to load-balance registrations to multiple servers based on their weights. Kind of similar scenario as discussed in this thread: http://lists.sip-router.org/pipermail/sr-users/2014-December/086235.html The problem I'm facing is that dispatcher is always choosing the first destination from the group. I do suspect that dispatcher is not considering registrations as load at all and only counts INVITES as load. If above is correct then is there a way to use dispatcher to count registrations as load ? Thanks, Sammy

2 3

Kamailio 5.0 - B2BUA
by Alex Balashov 24 Mar '16

24 Mar '16

Hi, I wanted to raise the possibility of an inline signalling-only B2BUA component to Kamailio. I know that's an extremely poor fit for Kamailio, and not at all what it's supposed to do. And there are many things about the OpenSIPS B2BUA module that reveal how awkwardly it is situated, as a square peg in a round hole. I myself am philosophically opposed to a B2BUA in Kamailio to the threshold of physical violence. However, the reason I bring it up for discussion is that there are very few viable, high-performance signalling-only B2BUA alternatives that are FOSS. There is enormous demand for a B2BUA, mainly for topology concealment purposes, from the short-duration traffic industry, which many users of Kamailio deal with. It's why a lot of them end up going to OpenSIPS; they need the B2BUA, and prefer to consolidate on one OpenSER brand. Anyone who has tried to run high-CPS traffic through the existing FOSS signalling-only B2BUAs out there situated in front of Kamailio has discovered, sometimes in a very financially painful way, that: - FreeSWITCH falls over at around +/- 300 CPS. This is with RTP relay disabled -- signalling-only. It requires horizontally scaling a large number of FreeSWITCH boxes to meet a capacity requirement of, say, 2000 CPS, which is unfortunate given the favourable proposition offered by Kamailio for the infrastructure unit economics. In other words, it's ironic to have to build a fleet of 10 FreeSWITCH boxes for the 1% problem of topology concealment when Kamailio can otherwise churn through 2000 CPS with no issues. - SEMS' 'sbc' module is a good candidate and can handle the load, but Frafos offers practically no support for it, with all efforts focused on their commercial ABC SBC product. That's very understandable, but just not practical given the high technical knowledge SEMS requires to deploy and maintain in this capacity. - None of the other userspace B2BUA folk traditions can handle the load. So, like I said, I personally recoil in shock and horror at the idea of introducing a B2BUA into something that was designed to be anything but a B2BUA. But, there is a huge market opportunity for this functionality in North America, and at the moment, users who need it are mostly ending up in OpenSIPS land. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

8 15

Presence Publish expires
by Igor Olhovskiy 24 Mar '16

24 Mar '16

Hi! Is there any possibility to set Expires in PUBLISH message to never in Presence module? I want to have BLF’s that indicates states, that driven via PUBLISH, but can be really irregular (let’s say once a week) -- Best regards, Igor

3 5

dialog rpc functions.
by Yasin CANER 24 Mar '16

24 Mar '16

1 0

kamailio in a multi-site configuration
by Sébastien Brice 24 Mar '16

24 Mar '16

Hi everyone i am installing Kamailio to scale asterisk on a remote hosted server in a way to manage calls dialogs in a real-time fashion. (and to manage many customers) Several users (maybe 80) are working at the head-office and i want them register their hardphone on the remote Kamailio Ipbx, to free them of ours end-of-life PABX (alcatel-Lucent). Can kamailio help me to achieve that ? What kind of software have I to install locally (mysql to replicate the master mysql db? a replicated asterisk instance?) ? Taking into consideration disaster recovery plan What could be my options to maintain internal call if my internet provider get down by exemple. thx you for your idea. Sébastien BRICE VoIP, Support et Intégration

3 2

set_advertised_address() / set_advertised_port() + PVs?
by Alex Balashov 23 Mar '16

23 Mar '16

Hello, I have a need to deploy Kamailio in AWS in a scenario of this sort: [Public Internet] <---> Kamailio <---> (Internal AWS servers) In such a scenario, Kamailio would be multihomed. Furthermore, in addition to sending out of two distinct network interfaces, the networking idiosyncrasies of AWS needs to be taken into account. AWS hosts only have an RFC1918 address homed natively, front-ended by 1:1 NAT externally. Normally, this is taken care of by advertised_address, which allows Kamailio to make outward representations about the network address by which it can be reached that are different to the IP to which it is bound. The problem here is that I need to do this conditionally, only on traffic going out the public interface. It should not be done on messages going to the "internal AWS servers". So, the questions that arise are: 1. Is this sane? Any unforeseen effects, e.g. vis-a-vis RR, provided enable_double_rr is enabled and that two genuinely different network interfaces are used? 2. Do set_advertised_address()/set_advertised_port() accept PV arguments, or are they pre-PV "core function folk traditions" in the same way as rewritehostport() and force_send_socket()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 10

Git clone failing?
by Richard Good 23 Mar '16

23 Mar '16

Hi Is anyone else struggling with git clone? richard@richard-laptop-new:~/Smile-dev/git_code$ git clone --depth 1 --no-single-branch git://git.kamailio.org/kamailio kamailio Cloning into 'kamailio'... fatal: read error: Connection reset by peer Tried on a few machines our side and keep getting connection reset. Regards *Richard Good* *Senior Manager: Applications & Services* *Smile Communications Pty (Ltd)* *Mobile:* +27 (0) 72 389 8365 * Skype:* richard.a.good richard.good(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com -- This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/home/email-disclaimer/ <http://www.smilecoms.com/disclaimer>

2 1

Random number generation kamailio
by Cibin Paul 23 Mar '16

23 Mar '16

Hi, How can I generate a random number say of length 10-14 in Kamailio. Can I use cfgutils for this. Please advise Regards Cibin

4 11

Feature request for Dialog terminate
by Yasin CANER 23 Mar '16

23 Mar '16

1 0

Kamailio Bridge
by Nelson Migliaro 23 Mar '16

23 Mar '16

Hello, I am trying to set up a Kamailio in Bridge Mode with two interfaces using rtpproxy. The problem I have is SDP information port and IP is not updated from outside to inside. But on the other hand the SDP is rewrited from inside to outside. In the rtpproxy log I am seeing this error: Mar 21 14:21:21 kamailio rtpproxy[6119]: ERR:create_twinlistener: can't bind to the IPv4 port 23414: Cannot assign requested address Mar 21 14:21:21 kamailio rtpproxy[6119]: ERR:handle_command: can't create listener and in kamailio log: ERROR: rtpproxy [rtpproxy.c:2727]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy Something strange and interesting is the port that shows up in the RTPPROXY error (23414) does not corresponds to any ports used. Thank you Nelson.- --------------------------------------------------------------------------------------------------------------------------------------------- Everything is set up this way: Kamailio Asterisk (asterisk-inside-ip) <-> Kamailio (inside-kamailio-ip) / Kamailio (outside-kamailio-ip) <-> sip-vendor-ip listen=inside-kamailio-ip listen=outside-kamailio-ip mhomed=1 if(src_ip=="asterisk-inside-ip"){ rtpproxy_manage("faie"); } if(src_ip=="sip-vendor-ip"){ rtpproxy_manage("faie"); } rtpproxy OPTIONS="rtpproxy -l inside-kamailio-ip/outside-kamailio-ip -m 20000 -M 30000 -u rtpproxy:rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s udp: 127.0.0.1:7722 -d DBUG:LOG_LOCAL6" ---------------------------------------------------------------------------------------------------------------------------------------------- SIP TRACES 2016/03/21 14:46:52.121472 asterisk-inside-ip:5060 -> inside-kamailio-ip:5060 INVITE sip:999999999@inside-kamailio-ip SIP/2.0 Via: SIP/2.0/UDP asterisk-inside-ip:5060;branch=z9hG4bK7f1ece8e;rport Max-Forwards: 70 From: 111111111<sip:1000@asterisk-inside-ip>;tag=as497457e6 To: <sip:999999999@inside-kamailio-ip> Contact: <sip:1000@asterisk-inside-ip:5060> Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE User-Agent: Abantix Date: Mon, 21 Mar 2016 13:46:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: <sip:888888888@local-domain ;user=phone>;privacy=off;party=calling Content-Type: application/sdp Content-Length: 251 v=0 o=root 478097443 478097443 IN IP4 asterisk-inside-ip ########## ASTERISK IP s=asterisk c=IN IP4 asterisk-inside-ip ########## ASTERISK IP t=0 0 m=audio 17146 RTP/AVP 8 101 ########## ASTERISK PORT a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/03/21 14:46:52.139435 outside-kamailio-ip:5060 ->sip-vendor-ip:5060 INVITE sip:999999999@sip-vendor-ip SIP/2.0 Record-Route: <sip:outside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Record-Route: <sip:inside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Via: SIP/2.0/UDP outside-kamailio-ip;branch=z9hG4bK0bfd.7bd4d3ae01e5850c14a3a939a612a05a.0 Via: SIP/2.0/UDP asterisk-inside-ip:5060;received=asterisk-inside-ip;branch=z9hG4bK7f1ece8e;rport=5060 Max-Forwards: 69 From: 1111111111 <sip:1111111111@sip-vendor-ip>;tag=as497457e6 To: <sip:999999999@sip.sip-vendor-ip> Contact: <sip:1000@asterisk-inside-ip:5060> Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE User-Agent: Abantix Voice Sevices Date: Mon, 21 Mar 2016 13:46:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: <sip:888888888@local-domain ;user=phone>;privacy=off;party=calling Content-Type: application/sdp Content-Length: 271 v=0 o=root 478097443 478097443 IN IP4 outside-kamailio-ip ########## OUTSIDE KAMAILIO IP REWRITED BY RTPPROXY s=asterisk c=IN IP4 outside-kamailio-ip ########## OUTSIDE KAMAILIO IP REWRITED BY RTPPROXY t=0 0 m=audio 24122 RTP/AVP 8 101 ########## OUTSIDE KAMAILIO PORT REWRITED BY RTPPROXY a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/03/21 14:46:57.393518sip-vendor-ip:5060 -> outside-kamailio-ip:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP outside-kamailio-ip;rport=5060;branch=z9hG4bK0bfd.7bd4d3ae01e5850c14a3a939a612a05a.1 Via: SIP/2.0/UDP asterisk-inside-ip:5060;received=asterisk-inside-ip;branch=z9hG4bK7f1ece8e;rport=5060 From: 1111111111 <sip:1111111111@sip-vendor-ip>;tag=as497457e6 To: <sip:999999999@sip.sip-vendor-ip>;tag=fkii1xm4-CC-26 Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE Record-Route: <sip:sip-vendo-ip;lr;ftag=as497457e6;vsf=AAAAAAEECQkCAgsNAXBeLxwZHQxBH11lcw--;did=0e.126;nat=yes> Record-Route: <sip:outside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Record-Route: <sip:inside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Contact: <sip:999999999@sip-vendor-carrier:5060;user=phone;transport=udp> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 71167517 71167517 IN IP4sip-vendor-ip ########## SIP VENDOR IP s=Sip Call c=IN IP4sip-vendor-ip ########## SIP VENDOR IP t=0 0 m=audio 44862 RTP/AVP 8 101 ########## SIP VENDOR PORT a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/03/21 14:46:57.395029 inside-kamailio-ip:5060 -> asterisk-inside-ip:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk-inside-ip:5060;received=asterisk-inside-ip;branch=z9hG4bK7f1ece8e;rport=5060 From: "1000" <sip:1000@asterisk-inside-ip>;tag=as497457e6 To: <sip:999999999@inside-kamailio-ip>;tag=fkii1xm4-CC-26 Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE Record-Route: <sip:sip-vendo-ip;lr;ftag=as497457e6;vsf=AAAAAAEECQkCAgsNAXBeLxwZHQxBH11lcw--;did=0e.126;nat=yes> Record-Route: <sip:outside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Record-Route: <sip:inside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Contact: <sip:999999999@sip-vendor-carrier:5060;user=phone;transport=udp> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 71167517 71167517 IN IP4sip-vendor-ip ########## SIP VENDOR IP NOT REWRITED BY RTPPROXY s=Sip Call c=IN IP4sip-vendor-ip ########## SIP VENDOR IP NOT REWRITED BY RTPPROXY t=0 0 m=audio 44862 RTP/AVP 8 101 ########## SIP VENDOR PORT NOT REWRITED BY RTPPROXY a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=nortpproxy:yes

2 2

Kamailio forward REGISTER return "general error"
by Cockhootec Yahrabee 23 Mar '16

23 Mar '16

Hi everyone, I am trying to forward registration from one kamailio instance to another one eg. UA1 <---> kamailioB <------> kamailioA I am trying to register on `kamailioA` with UA1. I have UA1 registered on kamailioA. I am routing SIP messages by adding this to `request_route` block: if (uri != myself) { forward(otherKamailioIP, 5060); } Where I am using as `otherKamilioIP on kamailioA the kamailioB IP address and vice versa. I can clearly see in Wireshark, that the SIP messages are routed correctly but I am getting 401 Unauthorized which is sent from kamailioA to kamailioB and then from kamailioB to UA1. I printed value of `auth_check` following this http://lists.sip-router.org/pipermail/sr-users/2014-November/085851.html and I am getting return code of `-1` which is according to the documentation http://kamailio.org/docs/modules/4.2.x/modules/auth_db.html#auth_db.f.www_a… the GENERIC ERROR. I don't know what to do next. Does anyone encounter this kind of error before? Thanks in advance

2 3

processing ACK
by jaflong jaflong 23 Mar '16

23 Mar '16

Hi Everyone I am manually processing sdp and handling media so I want to repond to ACK myself instead of forwarding on for media server. I catch the method=="ACK" condition. From within this condition how do I retrieve avp variables that were set in the INVITE. Requesting $avp("myvar") comes out as null. Also what function can I run so the ACK is replied to properly so it stops resending 200 OK of the Invite.

5 6

for kamailio registration
by Shiv Patidar 23 Mar '16

23 Mar '16

For registration , we would like to see the complete traces at all nodes whoever processing UE sending REGISTER method. could you please let me know how can we check the logging at all IMS nodes? i am able to successfully register and got 200OK but i want understand what and all processing has been done by server? can't i see default messages with the way install step i followed i need this for test/development

4 4

Kamailio 4.3.x kamailio-redis RPM
by Jay Patel 23 Mar '16

23 Mar '16

Is there any trusted yum repo which builds redis module RPM for EL6/CentOS? Seems like default make file does not build redis and it'snot published on rpm.kamailio.org. cnxcc will be added bonus. Thanks in advance. Regards, Jay

2 3

Kamctl dispatcher reload
by Alberto Sagredo 22 Mar '16

22 Mar '16

I have installed Kamailio 4.4.x and im getting this error when trying to reload dispatcher? Any idea where to look? kamctl dispatcher reload 500 ERROR Reloading data Also kamcmd dispatcher.reload gave me same result.

1 1

Kamailio World 2016 for students
by Daniel-Constantin Mierla 22 Mar '16

22 Mar '16

Hello, continuing the program from last years, based on the roots and the tight relation of Kamailio project with the academic environment, we are offering three seats at Kamailio World Conference, May 18-20, 2016, in Berlin, to students enrolled in universities or research institutes (both bachelor and PhD programs qualify). Last year we had students participating from Austria, Slovakia and Netherlands. If you are a student and want to participate, email me directly (miconda(a)gmail.com) or to registration(a)kamailio.org . Participation to all the content of the event (workshops, conference and social event) is free, but you will have to take care of expenses for traveling and accommodation. Write a short description about your interest in real time communications and what is the university or the research institute you are affiliate to. Also, if you are not a student, but you are in touch with some or have access to students forums/mailing lists, it will be very appreciated if you forward these details. More information about Kamailio World is available on the web site: - http://www.kamailioworld.com Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 18-20, 2016 Berlin, Germany - http://www.kamailioworld.com

1 0

Database engine not found
by Shiv Patidar 22 Mar '16

22 Mar '16

when i run this command kamctl fifo debug 6 then i got WARNING: database engine not found tried 'MYSQL'

2 1

Kamailio 5.0 - Better way of dealing with stateless CANCELs
by Alex Balashov 21 Mar '16

21 Mar '16

Currently (AFAIK), restarting Kamailio amidst production call processing is basically "safe"; except for no availability during the few seconds it takes to restart (which will just result in retransmissions until it is available), most things will happen "correctly" after restart even though TM state has been lost: (1) Initial requests will be routed as initial requests always were; (2) In-dialog requests will be loose-routed as sequential requests always were; (3) Replies to open transactions will fall back to stateless routing but will be delivered correctly to their destinations based on SIP fundamentals (i.e. Via). (4) rtpproxy & rtpengine control messages are grouped by Call-ID, so also stateless. If the proper destroy/remove functions are not called from failure_route[] due to lack of TM state, it's not so bad; rtpproxy & rtpengine will see an RTP timeout after a while and expire the bindings on their own. If dialog state is used, it will be lost, but assuming one is willing to live with that, it's okay. I don't know if there has been any work done to create a persistence layer for dialog that can be re-read completely on startup, and if it actually works - does it? - but it's a relatively small price to pay if it's important to integrate a change into production in the middle of the day. The one exception is CANCEL handling. CANCEL is a special animal, since it's a hop-by-hop (branch-level) request, so CANCELs sent from a caller apply to the 'caller -> Kamailio' branch. Kamailio generates separate CANCELs endogenously for one or more 'Kamailio -> gateway' branches. Stateful CANCEL handling with TM is implemented using t_check_trans() or t_relay_cancel(). For example, in the stock config[1] # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } Or, as in our case, more folklorically: if(is_method("CANCEL")) { if(!t_relay_cancel()) { # Corresponding INVITE transaction found, but error # occurred. sl_send_reply("500", "Internal Server Error"); exit; } # Corresponding INVITE transaction for CANCEL was not # found. exit; } In both cases, the corresponding INVITE transaction must exist. Unfortunately, there's no good alternative. According to RFC 3261 Section 16.11 ("Stateless Proxy"): Stateless proxies MUST NOT perform special processing for CANCEL requests. They are processed by the above rules as any other requests. So, in other words, route_logic(CANCEL) == route_logic(initial INVITE). Sometimes, this is possible - with considerable config logic labour - but other times the path taken by the CANCEL is not so deterministic, as for example with round-robin load balancing, random distribution, complex LCR, etc. One is basically left in these situations with the choice of implementing one's own Call-ID/branch => destination persistent state database of some kind, which is, to put it mildly, complicated and undesirable. Now, if the INVITE transaction receives a final negative reply, this will get back to the calling UAC, and it will process it correctly. However, some calls get answered with 2xx. Many UACs will behave reasonably in this situation: when they don't receive a 200 OK for their CANCELs but later receive an answer, they will go ahead and send the end-to-end ACK, then BYE the call. However, they cannot be reliably counted upon to do this. Some simply get drop the INVITE transaction after their CANCEL has gone unreplied for a short time, regardless of whether they receive a final negative reply for it. Is there a better way? Perhaps a feature can be devised by which Kamailio keeps some kind of lightweight and restart-persistent map to which to send the CANCELs? Or perhaps TM is due for a feature that allows the shm transaction table to be dumped to disk and persisted across restarts? Comments welcome. Also, if I'm missing something, please let me know! -- Alex [1] https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg#L466 -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 8

Missing pipelimit values in data returned by JSONRPC-S
by Brooks Bridges 21 Mar '16

21 Mar '16

When trying to pull a list of pipes and their current values by using pl.dump via the JSONRPC-S module, I am only getting the last result in the reply as opposed to an array. If I use the CLI RPC interface I get the data I expect. Here's a sanitized example: [root@server1 ~]# kamcmd pl.stats PIPE: id=user_1 load=0 counter=29 PIPE: id=user_2 load=0 counter=108 PIPE: id=user_3 load=0 counter=18 PIPE: id=user_4 load=0 counter=0 PIPE: id=user_5 load=0 counter=1 PIPE: id=user_6 load=0 counter=2 PIPE: id=user_7 load=0 counter=0 PIPE: id=user_8 load=0 counter=84 PIPE: id=user_9 load=0 counter=26 PIPE: id=user_10 load=0 counter=0 PIPE: id=user_11 load=0 counter=23 PIPE: id=user_12 load=0 counter=9 PIPE: id=user_13 load=0 counter=23 PIPE: id=user_14 load=0 counter=0 PIPE: id=user_15 load=0 counter=107 PIPE: id=user_16 load=0 counter=10 PIPE: id=user_17 load=0 counter=28 PIPE: id=user_18 load=0 counter=45 PIPE: id=user_19 load=0 counter=7 PIPE: id=user_20 load=0 counter=0 PIPE: id=user_21 load=0 counter=12 PIPE: id=user_22 load=0 counter=94 PIPE: id=user_23 load=0 counter=14 PIPE: id=user_24 load=0 counter=7 PIPE: id=user_25 load=0 counter=18 PIPE: id=user_26 load=0 counter=46 PIPE: id=user_27 load=0 counter=45 PIPE: id=user_28 load=0 counter=50 PIPE: id=user_29 load=0 counter=2 PIPE: id=user_30 load=0 counter=2 [root@server1 ~]# curl -s --header 'Content-Type: application/json' --data-binary '{"jsonrpc": "2.0", "method": "pl.stats"}' http://1.2.3.4:5060/jsonrpc {"jsonrpc":"2.0","result":"PIPE: id=user_30 load=0 counter=2"} [root@server1 ~]# The jsonrpc config and event route is about as basic as you can get: modparam("jsonrpc-s", "pretty_format", 0) modparam("jsonrpc-s", "transport", 1) event_route[xhttp:request] { xlog("L_INFO", "[R-HTTP-REQUEST:$si] -> Got HTTP request\n"); if($hu =~ "^/jsonrpc") { xlog("L_INFO", "[R-XHTTP-REQUEST:$si] -> Handling JSON RPC request: $hu\n"); jsonrpc_dispatch(); } else { xhttp_reply("404", "Not Found", "text/html", ""); } } Should the JSONRPC-S module be returning this as an array instead of just a single value? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

3 10

Removing Via and Record-Route headers
by Marrold 21 Mar '16

21 Mar '16

Hi, Is there any way to remove Via and Record-Route headers from requests sent to an endpoint and update the contact header, but have Kamailio statefully remember where the replies need to route? I imagine this would involve mangling the packets slightly to behave similar to a B2BUA. My motivations, in order of priority - 1) Reduce UDP packet size to avoid MTU limitations 2) Improve interoperability, don't expect end points to be able to properly parse / process many via / record-route headers 3) Hide topology. If someone could point me in the right direction in terms of configuration or modules, that would be great. Thanks

4 10

What to read after RFC 3261 to get started with Kamailio.
by Oivvio Polite 21 Mar '16

21 Mar '16

I want to get started with Kamailio. So far I've understood that I really need to grok SIP before tackling Kamailio so I'm reading the book "SIP demystified" as the starting point of my Kamailio journey. After that I will read RFC 3261. After that I think it's time to read something that will give me good understanding of how to actually implement applications with Kamailio. Searching Amazon for Kamailio returns "Building Telephony Systems with OpenSER" as the results that seems to be most specific to Kamailio. This book is from 2008. Is it still relevant or is it better to go with the online docs? regards, Oivvio

3 4

problem with relaying after t_susped() and t_continue()
by Uri Shacked 21 Mar '16

21 Mar '16

I have a problem when using t_suspend() and t_continue(). Let's say I would like to have a 5 sec delay before relaying an INVITE. For example the following code: * route[MYROUTE] {* * t_newtran();* * if (is_method("INVITE")) {* * t_reply("100", "Trying");* * t_reply("180", "In Queue");* * $sht(a=>$ci::t_index) = $T(id_index);* * $sht(a=>$ci::t_label) = $T(id_label);* * $sht(a=>$ci::t_suspended) = 1;* * xlog("L_WARN","SUSPENDING t_index = $sht(a=>$ci::t_index) and t_label = $sht(a=>$ci::t_label)");* * t_suspend();* * sleep(5);* * if (t_is_canceled()) drop;* * $sht(a=>$ci::t_suspended) = 0;* * t_continue("$avp(t_index)","$avp(t_label)","1");* * }* * route(RELAY);* * exit;* * }* * route[1] {* * xlog("L_WARN","After transaction continue\n");* * route(RELAY);* * exit;* * }* When using this code, I get the following error: Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: WARNING: <script>: After transaction continue Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: <core> [resolve.c:1693]: sip_hostport2su(): could not resolve hostname: "mysip" Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: tm [ut.h:319]: uri2dst2(): failed to resolve "mysip" Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: BUG: tm [t_lookup.c:1481]: t_unref(): tm: t_unref: REQ_ERR DELAYED should have been caught much earlier for 0x7fd4e727a468: 27 (hex 1b) If I try to t_continue() without any route: t_continue("$avp(t_index)","$avp(t_label)",""); The call is working great and the following error pops up: Mar 9 17:10:36 mysip /usr/local/sbin/kamailio[5830]: ERROR: tmx [tmx_mod.c:635]: w_t_continue(): empty action lists in route block [] Am I doing something wrong? My other question is regarding CANCEL. Let's assume I suspended the INVITE and in these 5 sec I received CANCEL for it. Is there a short and elegant way to remove the transaction from suspension? For now I do: * if (is_method("CANCEL")) {* * if (t_check_trans()) {* * if($sht(a=>$ci::t_suspended) == 1) {* * t_continue("$sht(a=>$ci::t_index)","$sht(a=>$ci::t_label)","");* * xlog("L_WARN","After transaction continue\n");* * }* * route(RELAY);* * }* * exit;* * }* Thanks, Uri

2 1

LCR prefix based
by Roberto Innaimi 21 Mar '16

21 Mar '16

Hi to all, i want to use prefix based LCR routing, but for some reason, something is not working as expected. Can anyone help me? This is the LCR configuration: #LCR MODULE loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://kamailioro:kamailioro@localhost/kamailio") modparam("lcr", "lcr_gw_table","lcr_gw") modparam("lcr", "id_column", "id") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "hostname_column", "hostname") modparam("lcr", "port_column", "port") modparam("lcr", "params_column", "params") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "strip_column", "strip") modparam("lcr", "tag_column", "tag") modparam("lcr", "flags_column", "flags") modparam("lcr", "defunct_column", "defunct") modparam("lcr", "lcr_rule_table", "lcr_rule") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "request_uri_column", "request_uri") modparam("lcr", "stopper_column", "stopper") modparam("lcr", "enabled_column", "enabled") modparam("lcr", "lcr_rule_target_table", "lcr_rule_target") modparam("lcr", "rule_id_column", "id") modparam("lcr", "gw_id_column", "gw_id") modparam("lcr", "priority_column", "priority") modparam("lcr","weight_column", "weight") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "tag_avp", "$avp(lcr_tag)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") modparam("lcr", "defunct_gw_avp", "$avp(s:defunct_gw_avp)") route[LCR] { if (status=="200") { xlog("LCR: Inside the LCR route\n"); } if(method=="INVITE") { xlog("We got an invite"); if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Couldn't load gateways"); sl_send_reply("500", "Server Internal Error - Cannot load gateways"); exit; } else { xlog("GW Selected '$avp(i:709)'\n"); xlog("Domain of destination: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("Couldn't proceed to next gateway"); sl_send_reply("503", "Service not available, no gateways found"); exit; } else { xlog("Calling the first matched gateway\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); } } } AND this is the error which i have got: 17(7483) ERROR: <script>: We got an invite17(7483) ERROR: <script>: GW Selected '1|sip:|3|||2873019861||5060||;transport=udp|1' 17(7483) ERROR: <script>: Domain of destination: <null> 17(7483) ERROR: <script>: To URI: sip:001326212223313@10.10.0.10 17(7483) ERROR: <script>: Calling the first matched gateway 17(7483) ERROR: <script>: ruri_user_avp: '001326212223313' 17(7483) ERROR: <script>: To URI after next_gw: sip: 001326212223313(a)10.10.0.10 17(7483) ERROR: <script>: Request URI: 326212223313 -- Roberto Innaimi roberto.innaimi(a)gmail.com

2 1

Problem with Freeswitch with IP public and Kamailio behind NAT
by Hai Bui Duc Ha 21 Mar '16

21 Mar '16

Hi all, I work on Kazoo system. I have NAT problem when configure Freeswitch and Kamailio. As the old network diagram: <https://lh3.googleusercontent.com/-dVJit4UPAL8/VuI4z1LBdKI/AAAAAAAABDo/zSG0…> I configure dispatcher to FS 1,2 with IP private and map IP public <-> IP private for each server via Router. This work fine. Then now, I want to improve FS performance so that I set up new FS with IP public: <https://lh3.googleusercontent.com/-YZl8d3HBo8E/VuI7ChgTGzI/AAAAAAAABD0/AQQW…> I configure dispatcher FS3 with IP public, connection between FS3 & ecallmgr via public IP is Okay but have problem with NAT: *When Kamailio foward SIP INVITE message to FS3 then FS3 response 200 OK message but it send to Ip private of Kamailio not IP public.* As Kamailio I have already configure advertise IP public: local.cfg listen=UDP_SIP advertise 125.212.212.40:5060 listen=TCP_SIP advertise 125.212.212.40:5060 View console log on Freeswitch, I see the ecallmgr response to FS3: 1. bridge({outbound_redirect_fatal="false",call_timeout=20, originate_timeout=20,local_var_clobber="true"}[ecallmgr_Authorizing-ID= "71310939684a2c8acc4e0e6dff1be6c2",ecallmgr_Owner-ID= "188cd101dcea060d080261b328fdf1ca",ecallmgr_Account-ID= "3f221d1ce6959ec04acd372923e253b9",sdp_secure_savp_only="false", sip_invite_domain="hahai1412.anttel-pro.ab-kz-02.antbuddy.com", presence_id="sharklasers(a)hahai1412.anttel-pro.ab-kz-02.antbuddy.com <https://happy.antbuddy.com/sharklasers@hahai1412.anttel-pro.ab-kz-02.antbud…> ",sip_h_X-KAZOO-AOR=sip:sharklasers@hahai1412.anttel-pro.ab-kz- 02.antbuddy.com,absolute_codec_string="^^:PCMU:VP8",leg_timeout="20", effective_callee_id_number="sharklasers",effective_callee_id_name= "sharklasers ",origination_callee_id_number="sharklasers", origination_callee_id_name="sharklasers ",ecallmgr_Realm=" hahai1412.anttel-pro.ab-kz-02.antbuddy.com",ecallmgr_Username= "sharklasers"]sofia/sipinterface_1/sharklasers@113.161.89.159:5914 <https://happy.antbuddy.com/sharklasers@113.161.89.159:5914>;transport=ws ;*fs_path=sip:10.127.15.36:5060*;lr;received="sip:113.161.89.159:5914 ;transport=ws") I research fs_path parameter: https://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path I guess this is broken It should be Public IP: *fs_path=sip:125.212.212.40:5060*. But I don't know how I can change this parameter ! Anybody faced that problem ? Any solution ? Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

2 1

Kamailio behind NAT
by Safdar Khan 21 Mar '16

21 Mar '16

Hello I have been running kamailio on local network including rtpengine and freeswitch. Now i am configuring kamailio with my public IP(trying with stock kamailio configuration to get started) #define WITH_NAT listen=udp:192.168.3.32:5060 listen=tcp:192.168.3.32:5060 advertised_address= "122.xx.xx.xx" advertised_port="5060" and also using rtpengine module at udp:192.168.3.32:22222 now the problems i am facing are 1.> UA can't register with UDP transport(not even recieving any packets at server ,testing with IMSdroid). 2.>TCP is registering the user, but users can't make calls . i've also noticed this error in /var/log/syslog when tcp ca't make calls ERROR: <core> [tcp_main.c:4250]: tcpconn_main_timeout(): connect 100.68.181.111:50842 failed (timeout) please help me out

2 1

srv_query question
by Vasiliy Ganchev 18 Mar '16

18 Mar '16

Hi folk! I am trying to implement srv_query usage in my script (server is based on Kamailio+Asterisk). The task: when outgoing REGISTER comes from Asterisk (outgoing trunk to operator), and R-URI consists domain name, need to store all ip-addresses of remote side (for following use) Currently, this works using dns_query, and storing in htable structure with remote IPs. But if remote side use SRV, I do not have all IPs Trying to use srv_query function from ipops, I am stuck with following: - srv_query(srvcname, pvid) to get srvcname - for example, "_sip._tcp.example.com", I need to provide, NAPTR record lookup (that is according to RFC3263, and Kamailio does this in core, while request processing) but I see no way how to make NAPTR lookup from script (not involving non Kamailio tools) Looking in mail list archive, found nothing that could help. What I am missing, is there an easy solution, how to cope with my case? Would appreciate any suggestion or idea. Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/srv-query-question-tp146546.html Sent from the Users mailing list archive at Nabble.com.

2 3

PostgreSQL - How To
by Barış Şekerciler 17 Mar '16

17 Mar '16

Hello everyone,I have Kamailio server which runs at 192.168.2.170 address and have PostgreSQL server which runs at 192.168.2.174.So I tried to arrange Kamailio for working with PostgreSQL but I think I give something miss. I couldnt handle it.Do you have any tutorial for this? Or can you tell me how can I exactly do this?Regards.

2 1

xhttp_rpc commands ?
by Uri Shacked 17 Mar '16

17 Mar '16

Hi, Can any one help me with an example for RPC command sent over http? My configuration works OK, I get the http request fine. I lack the knowledge of how to send commands..... For example, how do I send a command to list dialogs (dlg.list)? Or to profile_get_size ? Http://myserver:myport/http_rpc/dlg/????????? Thanks, Uri

4 10

htable - banning users
by meres gr 17 Mar '16

17 Mar '16

I am using the following script to ban users failing to authenticate with fail2ban: My problem is that it logs attempts of users only existing to the database I would like to modify it in order to log all FAILED authentication attempts Is this possible? # Authentication route route[AUTH] { #!ifdef WITH_AUTH if(is_present_hf("Authorization")) { if($sht(a=>$au::auth_count)==3) { $var(exp) = $Ts - 900; if($sht(a=>$au::last_auth) > $var(exp)) { sl_send_reply("403", "Try later"); exit; } else { $sht(a=>$au::auth_count) = 0; } } if(!www_authenticate("$td", "subscriber")) { switch ($retcode) { case -1: sl_send_reply("403", "Forbidden"); exit; case -2: #fixed null -> $null: if($sht(a=>$au::auth_count) == $null) $sht(a=>$au::auth_count) = 0; $sht(a=>$au::auth_count) = $sht(a=>$au::auth_count) + 1; xlog("User $fu Authenticating from ip $si\n"); if($sht(a=>$au::auth_count) == 3) xlog("User $fU Authenticating for the 3d time from ip $si\n"); $sht(a=>$au::last_auth) = $Ts; break; } www_challenge("$td"/*realm*/,"0"/*qop*/); exit; } $sht(a=>$au::auth_count) = 0; } else { www_challenge("$td","0"); exit; } } #!endif

3 2

SIP reg request not print in kamailio.log
by Shiv Patidar 15 Mar '16

15 Mar '16

i send sip reg request to kamailio then i dump tcp thent i got sip request like(INVITE ,TO,FROM ,via header file) but in kamailio.log file i am not got sip request what i need to do please help me my kamailio.log print this log in my terminal /usr/local/sbin/kamailio[25086]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15085]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:53]:sctp_core_destroy():SCTP API not initialized /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:75]:sctp_core_check_support():SCTP API not enabled-if you want to use it load sctp module /usr/local/sbin/kamailio[23190]:INFO:rr[../outbound/api.h:54]:ob_load_api():failed to import bind_ob /usr/local/sbin/kamailio[23190]:INFO:rr[rr-mod.c:174]:mod_init():outbound module not available /usr/local/sbin/kamailio[23190]:INFO:usrloc[hslot.c:51]:ul_init_locks():locks aaray size 1024 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is initially 163840 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is finally 327680 /usr/local/sbin/kamailio[23198]:INFO:ctl[io_listener.c:210]:io_listen_loop():io_listen_loop:using epoll_lt as the io watch method (auto detected) i want to print INVITE TO FROM Via header file in my kamailio.log

2 5

Kamailio and UCARP with Multiple VIPs defined
by Jonathan Hunter 15 Mar '16

15 Mar '16

Hi Guys, Using Centos 6.7, kamailio 4.2 and Ucarp I am testing multiple VIPs on the same server and having some issues. I have ucarp running, with 1 Public and 1 Private VIP defined, and both are pingable and failover in a cluster. The Public is for Subscribers to Register with, and the Private for Communication to the Asterisk media servers behind it. My problem is kamailio when started states its listening on both the VIPS, however when I try and force_socket over the Private IP address, it states ERROR: <core> [forward.c:218]: get_out_socket(): no socket foundMar 1 20:04:35 bt1proxy1 /usr/sbin/kamailio[20372]: ERROR: <core> [forward.c:220]: get_out_socket(): no corresponding socket found for(udp:10.70.0.72:5080)Mar 1 20:04:35 bt1proxy1 /usr/sbin/kamailio[20372]: ERROR: tm [t_fwd.c:520]: prepare_new_uac(): ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) This is when listening on the VIP which is 10.70.0.75 and I also look to force_socket out of this IP. It just doesnt work, however if I use the physical interface on the server, that being 10.70.0.70, it works fine and forwards requests on the Private LAN. I have added net.ipv4.ip_nonlocal_bind = 1 to sysctl.conf as well but no joy. If 1 VIP is defined I have no issue, its just with 2, has anyone had this issue before ? Any help would be great. Many thanks Jon

2 9

RTPPROXY error message
by Kevin Greene 15 Mar '16

15 Mar '16

Hi: I am sure this has been answered before on this list and I have tried the basic answers I got from Google without any luck. Any suggests would be great. The Kamailio host is running Oracle Linux 6.7 (64 bit) and is fully patched as of today. The Kamailio version is: Server:: kamailio (4.3.4 (x86_64/linux)) Build:: mi_core.c compiled on 14:25:16 Nov 25 2015 with gcc 4.4.7 I have two major issues: I am seeing this on my Kamailio log: Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: ERROR: rtpproxy [rtpproxy.c:1681]: send_rtpp_command(): can't send command to a RTP proxy Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy <udp:127.0.0.1:7722> does not respond, disable it Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP proxy Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: WARNING: rtpproxy [rtpproxy.c:1610]: rtpp_test(): support for RTP proxy <udp:127.0.0.1:7722> has been disabled temporarily So the initial thought is that the proxy is not running, however I can see the proxy is alive: [root@sipserver boot]# ps -All | grep rtpproxy 5 S 498 2843 1 1 80 0 - 32524 hrtime ? 01:29:06 rtpproxy When I do "netstat -a", I can see 24 established connections. This is a small sample: udp 0 0 localhost:47937 localhost:rtpproxy ESTABLISHED udp 0 0 localhost:38746 localhost:rtpproxy ESTABLISHED udp 0 0 localhost:33660 localhost:rtpproxy ESTABLISHED So I am not sure what is going on. The second problem: I see this in the Kamailio log: Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: db_mysql [km_dbase.c:124]: db_mysql_submit_query(): driver error on query: PROCEDURE kamailio.kamailio_rating does not exist (1305) Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [call kamailio_rating('default')] When I tried to create the database originally, the script that was part of the package would not work. I therefore manually created the database so I am thinking that something got left out. I have been over the schema and I am not sure what is missing. The database is MySQL (5.5) running in a separate host. Does anyone have an idea where to look what is missing? I was told that there is some sort of script that can "repair" the Kamailio schema in the database but I have not been able to find this magical script. Thanks. Kevin

2 1

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users