- sr-users - kamailio.org

Misrouting ACK packet for 200 Ok messages.
by Yasin CANER 13 Apr '16

13 Apr '16

4 3

Stable DEB
by Serge S. Yuriev 13 Apr '16

13 Apr '16

Hi, Shouldn't http://deb.kamailio.org/kamailio point to 4.4 version? And question about recent Ubuntu version support: Trusty is way to far behind latest Willy -- Serge S. Yuriev Lead VoIP engineer

2 2

Detecting calls with missing ACK (Lazy SIP scanners)
by Marrold 13 Apr '16

13 Apr '16

Hi, I have been running a couple of Asterisk honey pots to get a better understanding of the tools and methods potential hackers are using to exploit SIP servers. I have observed many attacks from the 'sipcli' user agent that don't send ACKs. At this stage I'm not sure what they're trying to achieve, whether it's a successful call to one of their test numbers, or maybe they will brute force anything that returns a 401 later, or maybe they're waiting for a 18X response. Below are three typical scenarios- ------ INVITE ------ > <--- 100 Trying --- <----- 200 OK ----- <----- 200 OK ----- <----- 200 OK ----- ( No ACK) ------ INVITE ------ > <-------- 503 -------- <-------- 503 -------- <-------- 503 -------- ( No ACK) ------ INVITE ------ > <-------- 401 -------- <-------- 401 -------- <-------- 401 -------- ( No ACK) Please could anyone point me in the right direction to detect these non completed calls with a missing ACK in Kamailio? I am unsure on the terminology I should be using to search the online documentation. Thanks

5 11

Kamailio 5.0 - Routing blocks in embedded scripting languages - Lua
by Daniel-Constantin Mierla 12 Apr '16

12 Apr '16

Hello, based on some of the plans for Kamailio 5.0 (https://www.kamailio.org/wiki/devel/kamailio-5.0-design), I started the implementation for the support to allow writing the active routing logic at runtime (respectively what are now the routing blocks) in other embedded languages than the native kamailio.cfg interpreter. Of course, besides a lot more extensive set of functions and expressions coming with the well established scripting languages, this way will allow reloading the logic without kamailio restart. I already committed the foundation for it, currently the request_route{} and reply_route{} can be implemented in Lua, relying on app_lua module. Lua was selected for first experiments given that I have some past experience with it, but other languages such as Perl, Python will hopefully follow soon, once the framework gets to a stable design. I wrote some initial docuentation on the wiki: - https://www.kamailio.org/wiki/devel/config-engines Would be helpful if people with experience on the other supported embedded languages will join the effort to design the framework to suit those languages, as well as other developers and community members to start exporting C functions. Also, there are some other things to decide on, listed in the to-do section, where suggestions/assistance are appreciated: - https://www.kamailio.org/wiki/devel/config-engines#to-do Many of them probably will be approached during the planned IRC devel meeting next week: - https://www.kamailio.org/wiki/devel/irc-meetings/2016a Any feedback is welcome! 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 0

Strange behaviour in working with CNXCC
by Dirk Teurlings 12 Apr '16

12 Apr '16

Hi, Working with CNXCC for quite some time now with success. Though running into the following problem. Running Kamailio 4.4 (Debian Wheezy) from the Kamailio repository. Setting up CNXCC with a dialog start event route, a dialog end event route and a cnxcc:shutdown event route. In the start event I set some variables in $var_dlg, because I will need them later on in the dialog, could be the end event, or the shutdown event. Now the strange thing is, that in the shutdown event, when CNXCC runs out of credit for instance, I no longer have access to the $var_dlg pool. All of the variables have been lost. Another strange thing is, if I cut off the call with CNXCC after just 1 second, the variables all are still available. But after 5 seconds cutoff, all variables are lost. On another system, running a 4.2 Kamailio with a CNXCC ported from 4.1, this still works fine. I've also tested this with Kamailio 4.2 and 4.3, with the same CNXCC code from 4.4. Any insight would be greatly appriciated! Cheers, Dirk

1 0

4.4 crash at start
by Juha Heinanen 12 Apr '16

12 Apr '16

I build kamailio from latest 4.4 git source and got crash at start: (gdb) where #0 0x0000000000449af8 in add_interfaces ( if_name=0x7f0e770106e8 "192.98.102.10", family=0, port=13880, proto=1978, ai_l=0x1277600) at socket_info.c:1210 #1 0x000000000044ad77 in fix_socket_list (list=0x5, type_flags=0xffffffff) at socket_info.c:1406 #2 0x000000000044de70 in fix_all_socket_lists () at socket_info.c:1807 #3 0x000000000041b535 in main (argc=0, argv=0x7fff8cfaf1e8) at main.c:2390 -- Juha

3 9

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

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