- sr-users - kamailio.org

Problem with my kamailio installation
by Ramazan Yilmaz 06 Nov '12

06 Nov '12

Hi, I installed kamailio 3.2.x a few times before and they were really fine. I decided to install a fresh one and followed the instructions on http://www.kamailio.org/wiki/install/3.3.x/git. However there is a problem: clients cannot register to kamailio. Each registration attempt produces the following error in the syslog: Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [tcp_main.c:3578]: BUG: handle_ser_child: fd -1 for 0 (pid 32414) Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [pass_fd.c:103]: ERROR: recv_all: 1st recv on 56 failed: Bad file descriptor Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [tcp_main.c:3363]: ERROR: handle_tcp_child: read from tcp child 0 (pid 0, no 0) Bad file descriptor [9] Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 17 on 56: Bad file descriptor (9) Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: ERROR: <core> [tcp_main.c:4006]: ERROR: send2child: send_fd failed for 0x7f510bc41588 (flags 0x4018), fd 17 What could be the problem? Any suggestion? In case you need, the server is ubuntu server 11.10, and I installed including mysql and tls modules.

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ERROR: presence not working correctly with Kamailio 3.3.2-1.1 and Postgresql 9.1
by Robert Verspuy 05 Nov '12

05 Nov '12

Hi All, I've got a problem with Kamailio 3.3.2-1.1 on CentOS 6.3 with locally postgresql 8.4.13-1.el6_3 libraries, connected to a remote postgresql91-9.1.6-1PGDP.rhel6 server The presence module tries to insert a record in the presentity table, with some xml in the body column. But the body value in the postgresql table is not represented as a string '<?xml ...etc' But encodes as '\x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462...etc' This gives problems when the presense module is trying to send out a notify based on the value in the database. With debugging, this gives the error: Entity: line 1: parser error : Start tag expected, '<' not found x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462d38273f3e3c The \x in front of the hex encodes string is not properly interpreted by postgresql or kamailio. The body column of the table presentity is a bytea column, and I believe postgresql 9 outputs these bytea columns a bit differently then postgresql 8. When I try to run the sql statement: insert into presentity (domain,username,event,etag,sender,body,received_time,expires ) values ('newsip.lifexs.nl','00086','presence','a.1352107949.18632.27.0','','<?xml version=''1.0'' encoding=''UTF-',1343534532,1345213723) On the postgresql 9.1 server locally, I still get the '\x3c3f786d6c... etc' value in the database. The bytea_output setting in postgresql 9 is now standard set to 'hex' in stead of escape. I've changed the bytea_output setting in postgresql.conf to 'escape', and then the presence is working without any issues. So maybe the database module of postgresql has to be changed to pick up these bytea encodings properly? Because i think kamailio should work correctly with the default postgresql settings. With kind regards, Robert Verspuy -- *Exa-Omicron* Eenspan 8-K 3897 AL Zeewolde http://exa.nl

2 1

Kamailio sending re invite to the next proxy
by Maurice Vistan 05 Nov '12

05 Nov '12

Hi All, I am using kamailio as dispatcher and set up kamailio just to send 200 okay back for the method register, I would like check if its normal for the kamailio send another invite on the next proxy listed on my dispatcher.lst after the first proxy process the invite and auth. Below is how the call is being process. user kamailio 1proxy 2proxy user2 invite |----------> | 100 | <---------- | invite | | ---------> | 100 |<--------- | auth |<-----------| auth | <------------------------| ack |------------->| ---------->| Invite |-------------->| --------------------------> | then process Is there any I can limit this? or it is realy the behavior? Thanks a lot! Maurice

2 2

ERROR: presence not working correctly with Kamailio 3.3.2-1.1 and Postgresql 9.1
by Robert Verspuy 05 Nov '12

05 Nov '12

Hi All, I've got a problem with Kamailio 3.3.2-1.1 on CentOS 6.3 with locally postgresql 8.4.13-1.el6_3 libraries, connected to a remote postgresql91-9.1.6-1PGDP.rhel6 server The presence module tries to insert a record in the presentity table, with some xml in the body column. But the body value in the postgresql table is not represented as a string '<?xml ...etc' But encodes as '\x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462...etc' This gives problems when the presense module is trying to send out a notify based on the value in the database. With debugging, this gives the error: Entity: line 1: parser error : Start tag expected, '<' not found x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462d38273f3e3c The \x in front of the hex encodes string is not properly interpreted by postgresql or kamailio. The body column of the table presentity is a bytea column, and I believe postgresql 9 outputs these bytea columns a bit differently then postgresql 8. When I try to run the sql statement: insert into presentity (domain,username,event,etag,sender,body,received_time,expires ) values ('newsip.lifexs.nl','00086','presence','a.1352107949.18632.27.0','','<?xml version=''1.0'' encoding=''UTF-',1343534532,1345213723) On the postgresql 9.1 server locally, I still get the '\x3c3f786d6c... etc' value in the database. The bytea_output setting in postgresql 9 is now standard set to 'hex' in stead of escape. I've changed the bytea_output setting in postgresql.conf to 'escape', and then the presence is working without any issues. So maybe the database module of postgresql has to be changed to pick up these bytea encodings properly? Because i think kamailio should work correctly with the default postgresql settings. With kind regards, Robert Verspuy -- *Exa-Omicron* Eenspan 8-K 3897 AL Zeewolde http://exa.nl

1 0

Entries in PUA List and RLS Watchers List
by Sangeeta Shah 03 Nov '12

03 Nov '12

Hello All, I am hoping the authors of the RLS/PUA modules can answer this question since I am not sure what's going on. I have the rls module enabled, with the following config spec: # ------rls module params ------ modparam("rls", "db_url", DBURL) modparam("rls", "db_mode", 2) modparam("rls", "integrated_xcap_server", 1) modparam("rls", "to_presence_code" ,10) modparam("rls", "server_address", "sip:rls@ip:5060") #!endif and modparam("pua_reginfo", "default_domain", "ip") modparam("pua_reginfo", "server_address", "sip:reginfo@ip") modparam("pua", "db_mode", 2) So for both I have DB_MODE set to DB ONLY. For whatever reason, even though all I am doing is bringing up my client I see over 250+ entries in the rls_watchers table for the same watcher and over 100 entries in the pua list table for the same presentity? Anyone have any idea what would be triggering this. There is definitely not anymore more than the normal messaging going on between the client and the server. Only error I see in the syslog: Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: db_mysql [km_dbase.c:122]: driver error on query: Duplicate entry '1350937406' for key 'expires_idx' Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11744]: DEBUG: presence [subscribe.c:1216]: subs->contact= sip:rls@ip:5060 - len = 25 Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11761]: DEBUG: <core> [db_val.c:117]: converting STRING [8475551001] Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11746]: DEBUG: <core> [parser/msg_parser.c:626]: method: <SUBSCRIBE> Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: <core> [db_query.c:210]: error while submitting query Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11744]: DEBUG: rls [rls.h:241]: did= 41826031568472f9-11752@127.0.0.1;533cb9e91f4b999cf76861cbb9ed54ed-93a3;a6a1c5f60faecf035a1ae5b6e96e979a-464d Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11761]: DEBUG: <core> [db_val.c:117]: converting STRING [10.50.251.12] Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11746]: DEBUG: <core> [parser/msg_parser.c:628]: uri: <sip:8475551001@ip> Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: pua [pua_db.c:1149]: DB insert failed -- Sangeeta Shah

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Error while "Make All"
by Techie Sup 02 Nov '12

02 Nov '12

Hello, I followed the instruction mentioned in Kamailio website. When I do a make all, I get the error. Please let me know how to fix this. I am new to Kamailio. ------------------- Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found CC (gcc) [M app_mono.so] app_mono_api.o In file included from app_mono_api.c:37: app_mono_api.h:27:26: warning: mono/jit/jit.h: No such file or directory app_mono_api.h:28:39: warning: mono/metadata/environment.h: No such file or directory In file included from app_mono_api.c:37: app_mono_api.h:35: error: expected specifier-qualifier-list before ‘MonoDomain’ app_mono_api.c:39:39: warning: mono/metadata/mono-config.h: No such file or directory app_mono_api.c:49: error: expected specifier-qualifier-list before ‘MonoDomain’ app_mono_api.c: In function ‘sr_mono_load_script’: app_mono_api.c:93: error: ‘sr_mono_load_t’ has no member named ‘next’ app_mono_api.c: In function ‘mono_sr_init_load’: app_mono_api.c:139: warning: implicit declaration of function ‘mono_config_parse’ app_mono_api.c:141: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:148: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:148: warning: implicit declaration of function ‘mono_jit_init’ app_mono_api.c:149: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:157: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:157: warning: implicit declaration of function ‘mono_domain_assembly_open’ app_mono_api.c:157: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:158: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:162: error: ‘sr_mono_load_t’ has no member named ‘next’ app_mono_api.c: In function ‘mono_sr_initialized’: app_mono_api.c:208: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c: In function ‘app_mono_exec’: app_mono_api.c:230: error: ‘sr_mono_env_t’ has no member named ‘msg’ app_mono_api.c:237: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:246: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:246: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:247: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:256: warning: implicit declaration of function ‘mono_jit_exec’ app_mono_api.c:256: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:256: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:257: warning: implicit declaration of function ‘mono_environment_exitcode_get’ app_mono_api.c:261: warning: implicit declaration of function ‘mono_jit_cleanup’ app_mono_api.c:261: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c: In function ‘app_mono_run’: app_mono_api.c:282: error: ‘sr_mono_env_t’ has no member named ‘msg’ app_mono_api.c:284: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:284: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:285: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:285: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:286: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:291: warning: implicit declaration of function ‘mono_domain_set’ app_mono_api.c:291: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:298: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:298: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c: At top level: app_mono_api.c:310: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ token app_mono_api.c:314: error: expected declaration specifiers or ‘...’ before ‘MonoString’ app_mono_api.c: In function ‘sr_mono_log’: app_mono_api.c:316: warning: implicit declaration of function ‘mono_string_to_utf8’ app_mono_api.c:316: error: ‘text’ undeclared (first use in this function) app_mono_api.c:316: error: (Each undeclared identifier is reported only once app_mono_api.c:316: error: for each function it appears in.) app_mono_api.c:316: warning: assignment makes pointer from integer without a cast app_mono_api.c:318: warning: implicit declaration of function ‘mono_free’ app_mono_api.c: At top level: app_mono_api.c:321: error: expected ‘)’ before ‘*’ token app_mono_api.c:328: error: expected ‘)’ before ‘*’ token app_mono_api.c:338: error: expected ‘)’ before ‘*’ token app_mono_api.c:408: error: ‘sr_mono_api_version’ undeclared here (not in a function) app_mono_api.c:410: error: ‘sr_mono_err’ undeclared here (not in a function) app_mono_api.c:411: error: ‘sr_mono_dbg’ undeclared here (not in a function) app_mono_api.c:412: error: ‘sr_mono_modf’ undeclared here (not in a function) app_mono_api.c: In function ‘sr_mono_load_class_core’: app_mono_api.c:420: warning: implicit declaration of function ‘mono_add_internal_call’ app_mono_api.c: At top level: app_mono_api.c:427: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ token app_mono_api.c:478: error: expected ‘)’ before ‘*’ token app_mono_api.c:530: error: expected ‘)’ before ‘*’ token app_mono_api.c:579: error: expected ‘)’ before ‘*’ token app_mono_api.c:632: error: expected ‘)’ before ‘*’ token app_mono_api.c:679: error: expected ‘)’ before ‘*’ token app_mono_api.c:729: error: ‘sr_mono_pv_gets’ undeclared here (not in a function) app_mono_api.c:729: error: initializer element is not constant app_mono_api.c:729: error: (near initialization for ‘_sr_M_export_pv[0].method’) app_mono_api.c:730: error: ‘sr_mono_pv_geti’ undeclared here (not in a function) app_mono_api.c:730: error: initializer element is not constant app_mono_api.c:730: error: (near initialization for ‘_sr_M_export_pv[1].method’) app_mono_api.c:731: error: ‘sr_mono_pv_seti’ undeclared here (not in a function) app_mono_api.c:731: error: initializer element is not constant app_mono_api.c:731: error: (near initialization for ‘_sr_M_export_pv[2].method’) app_mono_api.c:732: error: ‘sr_mono_pv_sets’ undeclared here (not in a function) app_mono_api.c:732: error: initializer element is not constant app_mono_api.c:732: error: (near initialization for ‘_sr_M_export_pv[3].method’) app_mono_api.c:733: error: ‘sr_mono_pv_unset’ undeclared here (not in a function) app_mono_api.c:733: error: initializer element is not constant app_mono_api.c:733: error: (near initialization for ‘_sr_M_export_pv[4].method’) app_mono_api.c:734: error: ‘sr_mono_pv_is_null’ undeclared here (not in a function) app_mono_api.c:734: error: initializer element is not constant app_mono_api.c:734: error: (near initialization for ‘_sr_M_export_pv[5].method’) app_mono_api.c:750: error: expected ‘)’ before ‘*’ token app_mono_api.c:801: error: expected ‘)’ before ‘*’ token app_mono_api.c:850: error: expected ‘)’ before ‘*’ token app_mono_api.c:895: error: expected ‘)’ before ‘*’ token app_mono_api.c:930: error: ‘sr_mono_hdr_append’ undeclared here (not in a function) app_mono_api.c:930: error: initializer element is not constant app_mono_api.c:930: error: (near initialization for ‘_sr_M_export_hdr[0].method’) app_mono_api.c:931: error: ‘sr_mono_hdr_remove’ undeclared here (not in a function) app_mono_api.c:931: error: initializer element is not constant app_mono_api.c:931: error: (near initialization for ‘_sr_M_export_hdr[1].method’) app_mono_api.c:932: error: ‘sr_mono_hdr_insert’ undeclared here (not in a function) app_mono_api.c:932: error: initializer element is not constant app_mono_api.c:932: error: (near initialization for ‘_sr_M_export_hdr[2].method’) app_mono_api.c:933: error: ‘sr_mono_hdr_append_to_reply’ undeclared here (not in a function) app_mono_api.c:933: error: initializer element is not constant app_mono_api.c:933: error: (near initialization for ‘_sr_M_export_hdr[3].method’) make[1]: *** [app_mono_api.o] Error 1 make: *** [modules] Error 1 Regards Selvam

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Kamailio WebSocket support
by Owen Lynch 02 Nov '12

02 Nov '12

Hi, I see Peter Dunkley's websocket work is not in the latest 3.3 release (3.3.2). What is the road map for this? Thanks, Owen Lynch

2 1

Re: [SR-Users] Kamailio and sylkserver integration
by Vijay Thakur 01 Nov '12

01 Nov '12

Thanks for your reply. Will you please explain that how can i forward calls from kamailio to sylkserver. Regards, VJ++ Message: 7 Date: Wed, 31 Oct 2012 08:26:50 +0100 From: "Olle E. Johansson"<oej(a)edvina.net> Subject: Re: [SR-Users] Kamailio and sylkserver integration To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List"<sr-users(a)lists.sip-router.org> Message-ID:<D7F883EA-4231-4D21-A79F-BB3449F972B1(a)edvina.net> Content-Type: text/plain; charset=us-ascii 31 okt 2012 kl. 06:49 skrev Vijay Thakur<vijay.thakur(a)loopmethods.com>: > Dear Sir, > > Is there any one in the list who have set up (integrated) Kamailio with sylkserver for conference. There is very less available information on the internet on this topic. All the required changes are made in sylkserver config files. > Kindly advise for required changes in Kamailio. As far as I understand, Sylkserver is just a SIP server. Forward calls from kamailio to sylkserver using any of the forwarding commands. /O -- * Olle E. Johansson -oej(a)edvina.net * Kamailio & SIP Masterclass Miami FL December 2012 *http://edvina.net/training/

2 1

Two kamailio servers using same DB but different version table
by Uri Shacked 31 Oct '12

31 Oct '12

Hi, can two kamailio servers, one running 3.2 and the other 3.3 run using the same DB? one will turn to version table X and the other to version table Y (i am not using the modules that the DB table changed... BR, Uri

2 1

statefull proxy Q
by Alex Solt 31 Oct '12

31 Oct '12

Hi, Is there any function in kamailio that could differentiates between a new incoming INVITE message and a re-INVITE message for an existing active call? Thanks, AS

3 6

2600Hz selects Kamailio
by Olle E. Johansson 31 Oct '12

31 Oct '12

Friends, The 2600Hz project that builds an API-driven telephony service have selected to switch to Kamailio as a SIP proxy for their SBC. They blog about the switch, comparing Kamailio with their previous choice and coming to a conclusion: "To our minds, a more active development community means that other users are likely to help solve bottlenecks for us, so we spend less time worrying about our SBC, and more time writing fun new applications! Conclusion: We’re switching to Kamailio because: • Many of our Major Clients already use it • The Naming Conventions are more sensible • The Pace of Innovation on Kamailio is faster" Join me in welcoming 2600Hz to the Kamailio project! /O Links: * http://www.kamailio.org/w/2012/10/2600hz-selects-kamailio/ * http://blog.2600hz.com/post/34639934434/our-other-border-brother-kamailio

1 0

Kamailio and sylkserver integration
by Vijay Thakur 31 Oct '12

31 Oct '12

Dear Sir, Is there any one in the list who have set up (integrated) Kamailio with sylkserver for conference. There is very less available information on the internet on this topic. All the required changes are made in sylkserver config files. Kindly advise for required changes in Kamailio. With Regards, Vijay Thakur

2 1

additional user data for SIP gateway usage?
by Jeremy Ardley 31 Oct '12

31 Oct '12

Hi, I'm after advice. I'm developing a script to provide a degree of PBX functionality for a small to medium office using Kamailio & SEMS. This includes mapping internal extensions to external PSTN numbers via the SIP trunk provider. The incoming mapping is easy using the db_alias functionality. The outgoing mapping is more difficult as the SIP trunk provider requires both the Full National Number and an account number matching the FNN in the outgoing From header. I'd appreciate suggestions on the best way to store, retrieve, and manage the additional information required for outgoing PSTN calls. I think avpops will work but is there some other way to do this? Thanks,

2 1

SIP connection tracking
by Daniil Baturin 31 Oct '12

31 Oct '12

Hi all, We have two softswitch hosts that don't support TLS. We have a peer that requires TLS, they terminate all our connections on a single address. We want to use kamailio as a proxy for it. Could you point me to the documentation and/or working examples that describe how to implement connection tracking to ensure replies to connections originated from host A are relayed back to host A, and connections from host B get back to host B? Thanks in advance. -- #!/usr/bin/env perl @a=split(//, "daniil @ baturin . org" );# Daniil Baturin @b=split(//,q/Px%!+o0Q6lh*7dp$.@8#%|y{/);while($i<24){$_.= chr((ord(@b[$i])-ord(@a[$i])+62)%94+32);$i++};print"$_\n"#

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Re: [SR-Users] Blacklist initialization
by Javi Gallart 30 Oct '12

30 Oct '12

Hi Olle What about doing the init stuff in event-route[htable:mod-init] ? Regards Javi ------Original Message------ From: Olle E. Johansson Sender: sr-users-bounces(a)lists.sip-router.org To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List ReplyTo: SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List Subject: [SR-Users] Blacklist initialization Sent: Oct 30, 2012 21:58 Hi! I am looking for a way to add addresses to the IP blacklist in Kamailio at startup. Is that possible? I know I can do it with sercmd, but I fail to find a documented way to do it in the configuration... Must be missing something simple. Thank you for your help! /O _______________________________________________ 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

Blacklist initialization
by Olle E. Johansson 30 Oct '12

30 Oct '12

Hi! I am looking for a way to add addresses to the IP blacklist in Kamailio at startup. Is that possible? I know I can do it with sercmd, but I fail to find a documented way to do it in the configuration... Must be missing something simple. Thank you for your help! /O

1 0

Default database named now kamailio
by Daniel-Constantin Mierla 30 Oct '12

30 Oct '12

Hello, just short note to let everyone know that the default database for Kamailio is now named 'kamailio' - the change happened over the weekend as part of migrating from last bits from openser to kamailio namespace. Also, the default usernames and passwords to access the database are now: - kamailio with password kamailiorw for read-write operations - kamailioro with password kamailioro for read-only operations If you set db_url parameters in the config file, there should be no impact on existing configurations. You also have to make sure you set the right value in kamctlrc file if you update kamctl. This change impacts development version (to be released as 3.4.0). Two other components are in the to-do list for this change: perl and snmpstats Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Question about acc module
by Thorsten Willand 30 Oct '12

30 Oct '12

3 2

Unknown status in presence_dialog
by David 29 Oct '12

29 Oct '12

Hello, I am using the presence_dialoginfo module on a stand alone presence server. I have this : if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; If the server does not have any status information, it will send out a blank notify with a 0 content length. Is there anyway to tell Kamailio to send an xml doc with status "terminated" if the server does not know the status that was requested by the SUBSCRIBE? Thanks, David Server : version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 14:19:23 Apr 19 2012 with gcc 4.4.3

1 0

siptrace vs. sipcapture
by Mino Haluz 29 Oct '12

29 Oct '12

Hi, we are thinking about switching from siptrace to sipcapture + HEP encapsulation. Do you know some reasons why we should switch to sipcapture or why we should not ? ;) We have multiple SIP proxies and Cisco gateways. In earlier versions of kamailio was siptrace blocking, that's why we started to looking for some other capturing mechanism. I think now siptrace is buffered, so we are hesitating .. Mino

2 1

rtpproxy ipv4/ipv6 bridging question
by Juha Heinanen 29 Oct '12

29 Oct '12

is it possible to make rtpproxy to include both ipv4 and ipv6 address in sdp? if not, i don't see how ipv4/ipv6 bridging can be made to work when $rd is domain name. -- juha

1 0

NEW: new module: xhttp_pi
by Ovidiu Sas 28 Oct '12

28 Oct '12

Hello all, A new module is available in trunk. It provides a web provisioning interface: http://kamailio.org/docs/modules/devel/modules/xhttp_pi.html A sample config example is provided in the source tree: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… An additional xml file is required for web config framework (what to configure, where and how). A sample xml file is provided in the source tree for dispatcher and dialplan tables: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… When kamailio is up and running, a web provisioning interface for managing db records will be available. No supplementary web server needs to be installed in order to manage the db records. Also, it provides a tight control over what can be modified and how. Testing and feedback is appreciated. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

2 1

web homer not displaying data
by Christopher Holland 26 Oct '12

26 Oct '12

I have sipcapture and siptrace setup on the same server using kamailio 3.4.0-dev4 and webhomer 3.3.0 setup to capture sip traffic from a mirriored port on the LAN. Interface ETH1 is to capture and ETH0 is for managment. The sipcapture database is getting data put into it. I can log into webhomer, but see no data in the graphes nor can I search. I have double checked my database permissions and they seem to be set up correctly. webserver is set up with: Apache/2.2.16 (Debian) PHP5 MYSQL Json phpmyadmin Here is a copy of my kamailio.cfg and configuration.php rev_dns=no auto_aliases=no debug=1 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=5 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* IP and port for HEP capturing) */ listen=udp:127.0.0.1:9060 mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" loadmodule "pv.so" loadmodule "db_mysql.so" loadmodule "sipcapture.so" loadmodule "textops.so" loadmodule "siptrace.so" # ----- mi_fifo params ----- ####### Routing Logic ######## modparam("sipcapture", "db_url", "mysql://openser:******@localhost/homerdb") modparam("sipcapture", "capture_on", 1) /* activate HEP capturing */ modparam("sipcapture", "hep_capture_on", 1) /* IP to listen. Port/Portrange apply only on mirroring port capturing */ modparam("sipcapture", "raw_socket_listen", "10.0.0.0:5060-5080") /* Name of interface to bind on raw socket */ modparam("sipcapture", "raw_interface", "eth1") /* activate IPIP capturing */ #modparam("sipcapture", "raw_ipip_capture_on", 1) /* My table name*/ modparam("sipcapture", "table_name", "sip_capture") /* children for raw socket */ modparam("sipcapture", "raw_sock_children", 4) /* insert delayed */ #modparam("sipcapture", "db_insert_mode", 1) /* activate monitoring/mirroring port capturing. Linux only */ modparam("sipcapture", "raw_moni_capture_on", 1) /* Promiscious mode RAW socket. Mirroring port. Linux only */ modparam("sipcapture", "promiscious_on", 1) /* activate Linux Socket Filter (LSF/BPF) on mirroring interface. Linux only */ #modparam("sipcapture", "raw_moni_bpf_on", 1) /* sipcapture node */ modparam("sipcapture", "capture_node", "homer01") # check IP and port of your capture node modparam("siptrace", "duplicate_uri","sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag",22) modparam("siptrace", "trace_on", 1) # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { sip_capture(); drop; } onreply_route { sip_capture(); drop; } ################################## configuration.php <?php /*********************************************************************************/ /* Access db of homer */ define('HOST', "localhost"); define('PORT', 3306); define('USER', "webhomer"); define('PW', "******"); define('DB', "web_homer"); /* Homer connection * this user must have the same password for all Homer nodes * please define all your nodes in homer_nodes table */ define('HOMER_HOST', "localhost"); /* DEFAULT. Don't forget insert this host to your DB nodes table */ define('HOMER_PORT', 3306); define('HOMER_USER', "openser"); define('HOMER_PW', "******"); define('HOMER_DB', "homerdb"); define('HOMER_TABLE', "sip_capture"); /*********************************************************************************/ /* webHomer Settings * Adjust to reflect your system preferences */ define('PCAPDIR',"/var/www/webhomer/tmp/"); define('WEBPCAPLOC',"/webhomer/tmp/"); define('APIURL',"http://localhost"); define('APILOC',"/webhomer/api/"); /* INCLUDE preferences */ include("preferences.php"); ?>

2 2

Message Routing To/From Gateway
by Ed Brady 26 Oct '12

26 Oct '12

Hi, I am trying to route SIP messages to a gateway. I am using SIP MESSAGE over UDP as well as the MSILO module for storing messages for users that aren't currently logged in. I have read through the Dynamic Routing module info but am not able to determine the best way to accomplish the routing I require. Our scenario is outlined below. User A sends a message addressed to "sip:12345678900@mydomain.com" Kamailio at mydomain.com checks if 12345678900 is a registered user in the system. If not, the message header is translated to "sip:12345678900@gatewaydomain..com" and forwarded. Incoming messages need to be accepted from their IP as well. It would also be useful if the address came as "sip:2345678900@mydomain.com" could be translated to "sip:*1*2345678900@gatewaydomain.com". Is there any built in functionality I can configure for this, or will this need to be scripted? Thanks, Ed

2 9

[OT] the role of SBCs
by Daniel-Constantin Mierla 26 Oct '12

26 Oct '12

Hello, based on the outcome of the discussion carried in the thread: http://lists.sip-router.org/pipermail/sr-users/2012-August/074480.html I am looking again the clarify some aspects out there in the VoIP world. So, as mainly dealing with proxy/sip signaling deployments, it's very often to be the first one hit by support issues claiming things don't work. Then you investigate and end up in conclusions like in the above thread: "The problem was at the SBC, __where I did not expect it__." The underlined part heats me up a bit, because I never understood from where it comes this perception that SBC is a MUST-TO-HAVE and the PERFECT (never mistaken or breaking things) node in a VoIP networks. To some extent, the SBC is just a very costly SIP ALG, and a SIP ALG is there to break the things. I don't want to start like a flame war, but is it something that I am obviously missing in regards to what benefits a SBC can bring? I see only inconveniences: - another point of failure - it is a b2bua, therefore very unlikely to offer the same performances of a proxy - if transcoding is needed, a media server can be used behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - if topology hinding is wanted in the b2bua fashion (not the proxy fashion with encoding headers), then the b2bua can be behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - nat traversal was solved long time ago in proxy environment, being scalable by deploying a farm of rtp proxy I don't want to go to other features, including the transport layer, it's a clear win of the proxy in my experience (ok, being deep involved in this project). Then, what makes the SBC so desirable in many companies/voip deployments? If any SBC user here that can share, what was the reason to buy such a device? Any conceptual functionality that cannot be achieved with the proxy as the first hop in front of the (wild) clients? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

10 14

MSRP: Generation of REPORTs on the relay
by Peter Dunkley 26 Oct '12

26 Oct '12

Hello, One of my colleagues at Crocodile has been doing some testing with the msrp module. With the auth change that was made yesterday, and the updated configuration I put in the README example it seems to mostly work. However, there is one set of scenarios relating to failure REPORTs that don't seem to be possible with the current module. RFC 4976 section 6.4.1 paragraph 3 states: If the Failure-Report header is "yes" or "partial", and if there is a problem processing the SEND request or if an error response is received for that SEND request, then the relay MUST respond with an appropriate error response in a REPORT back to the original source of the message. This means that when a SEND request that contains a Failure-Report: header with value of "yes" or "partial" is relayed by Kamailio we need to maintain some state so that when the TCP send fails, or we receive a failure response from the next hop, Kamailio can generate and send a REPORT. At the moment I don't think the msrp module has these capabilities. If the module is getting some re-work before the next release it would be good to get this in there. I am quite happy to help out in any way I can, but I am not sure of the best way to proceed with this. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Kamailio on Twitter: @kamailioproject
by Olle E. Johansson 26 Oct '12

26 Oct '12

The marketing arm of the Kamailio project has without further discussion opened up a Twitter account for Kamailio, to support our web site news, G+ and Facebook pages. Feel free to contribute with case stories and news we can publish on the web site and follow up in the social media flow. The twitter account is @kamailioproject - please follow it. And yes, Kamailio is THE best thing since the invention of sliced bread! This week's EXTRAORDINARY offer: Download TWO licenses of Kamailio for the same licensing cost as we've been offering for ONE license! And get an unlimited number of user licenses for your Kamailio SIP accounts! Kamailio greetings! Have a great SIP weekend! /Marketing

1 0

MSRP relay authentication
by Peter Dunkley 25 Oct '12

25 Oct '12

Hi, MSRP AUTH requests must be authenticated using HTTP Digest authentication. Part of the concatenated data when doing this authentication is the method name. Because of the way MSRP requests are formatted, the part of the request-line that would contain the method name in HTTP or SIP requests is always MSRP. The MSRP method name (AUTH in this case) is at the end of the request-line. When Kamailio converts an MSRP request to SIP it has a method name of MSRP. This means that when Kamailio authenticates an MSRP request it uses "MSRP" as the method name, not the actual MSRP method name (AUTH). Is this correct? Should MSRP requests be authenticated with a method name of "MSRP" (which kind-of makes sense as this is the string at the start of the MSRP request line - which is where the method names are in HTTP and SIP) or should the MSRP method name of "AUTH" be used? If it is the later, does anyone know of an easy way to add this to Kamailio? Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 4

dns queries on ipv6 addresses
by Juha Heinanen 25 Oct '12

25 Oct '12

from the mailing list archives i have learned that dns_try_ipv6=yes in config file is supposed to prevent dns query on ipv6 address. if i have it "no", i get to syslog: Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [ut.h:314]: ERROR: uri2dst: failed to resolve "[2002:c062:670a::10]" :unresolvable A or AAAA request (-7) Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches and sip proxy naturally does not send anything out. if i have it "yes", request is sent out properly, but still sip proxy makes dns queries on the ipv6 address: Oct 24 14:08:35 siika /usr/sbin/sip-proxy[18196]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail why is that? how to completely prevent dns queries on ipv6 addresses? -- juha

6 16

rtpproxy delete fails
by Juha Heinanen 25 Oct '12

25 Oct '12

i made rtpproxy test in setup where two sip phones have registered the same AoR test(a)test.fi. one is behind nat and the other is not. when i call this AoR, my sip proxy executes rtpproxy_manage("FROW3"); in branch route of the branch that is behind nat. syslog shows: Oct 19 15:49:59 siika /usr/sbin/sip-proxy[18596]: INFO: Calling rtpproxy_manage on INVITE <sip:test@192.168.0.31:5074;transport=tcp> Oct 19 15:49:59 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:48714: 18596_16 US ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm 192.98.103.10 8000 wnzdf;1 Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Creating new call Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50104/50105 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50106/50107 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Returning to SIP proxy: 18596_16 50104 192.98.103.10 4 then the uas hehind nat replies with 480 and sip proxy executes in onreply route rtpproxy_manage("FROW3"); that fails and i get to syslog: Oct 19 15:50:15 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:56183: 18594_12 D ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm wnzdf shsqn Oct 19 15:50:15 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Tags didn't match for delete message, ignoring looks like this mismatch is due to the missing ";1" in from tag param that was there when rtpproxy_manage was executed on invite. if my theory is correct, why is ";1" missing? if not correct, what did i do wrong? -- juha

4 17

udp transport in rtpproxy sockets
by Juha Heinanen 25 Oct '12

25 Oct '12

rtpproxy module readme gives impression that udp transport is the only possibility: 4.1. rtpproxy_sock (string) Definition of socket(s) used to connect to (a set) RTPProxy. It may specify a UNIX socket or an IPv4/IPv6 UDP socket. if that is the case, what is the justification for it, i.e, why is tcp (and better yet tls) transport not supported? -- juha

3 3

rtpproxy module flags 1 and 2 question
by Juha Heinanen 25 Oct '12

25 Oct '12

i'm new to rtpproxy (have been using mediaproxy for years) and now try to understand if it would make sense to switch over. rtpproxy readme has this on flags value 1: + 1 - append first Via branch to Call-ID when sending command to rtpproxy. ... This is especially useful if you have serially forked call scenarios where rtpproxy gets an "update" command for a new branch, and then a "delete" command for the previous branch, which would otherwise delete the full call, breaking the subsequent "lookup" for the new branch. how it would be possible in SERIAL forked scenario to get anything for a new branch before previous branch has first failed and gone? -- juha

4 7

Manually add Via Header
by Vassilis Radis 25 Oct '12

25 Oct '12

I have a situation where a far end SIP provider doesn't behave properly when sending 487 replies. The scenario is this: I have a registered user calling into my kamailio which ,using lcr module, routes the call to a SIP provider. When the caller Cancels the call, my kamailio forwards the cancel message to the provider (along with a 200 Cancelling message to the caller). The problem is with the 487 sent by the provider to my kamailio: It does not contain all the two Via headers. It only contains the via header that names my kamailio. So when kamailio gets the 487, it removes its Via header and sends the 487 back to the caller, which now doesnt find any via header. Note that the far end provider, correctly sends all the Via Headers in other replies (183 etc). So now I am trying to "intercept" the 487 from this specific provider in the onreply route, and patch it with the header that the provider should have included. How can I do it? The problem is twofold: 1.I need to detect which header I should manually add (maybe store the Via Header set from the initial invites or the 183 replies, and get it from there? .I dont know how to do that) 2.I need to add it. (How can I manually add a Via header? .This seems easier Can anyone help me, maybe with another solution?

4 5

ip address fields
by Juha Heinanen 25 Oct '12

25 Oct '12

lcr_gw table has: ip_addr varchar(47) trusted table has: src_ip varchar(50) address table has: ip_addr varchar(48) why is it that they are not all the same? also, why the common length is not 8 * 4 + 7 = 39 (http://en.wikipedia.org/wiki/IPv6#Address_format)? -- juha

4 19

Kamailio & OMA Presence Compliance
by Thomas Hoffman 24 Oct '12

24 Oct '12

My colleague and I have been trying to test Kamailio configured as a Presence Server using OMA standards/messages. It appears that Kamailio does not support the OMA standards as it pertains to Presence. This is primarily in the area of the XCAP messages. Can someone please confirm if this is the case? If this is the case, it is on the roadmap to support it? If not, we'd like to know that too and we'll continue trying to get it to work. Thank you, Tom ______________________________________________________________________________________________________________________________ This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify so to the sender by e-mail and delete the original message. In such cases, please notify us immediately at sanchar-sadhan(a)infinite.com . Further, you are not to copy, disclose, or distribute this e-mail or its contents to any unauthorized person(s) .Any such actions are considered unlawful. This e-mail may contain viruses. Infinite has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachments. Infinite reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infinite e-mail system. ***INFINITE******** End of Disclaimer********INFINITE********

1 0

RFC5626 SIP Outbound Support of Flow-Timer Header
by Jonathan Hunter 24 Oct '12

24 Oct '12

Hi, Testing with RFC5626, can you confirm that the Flow-Timer header field is supported and configurable with Kamailio 3.3 ? Thanks Jon

2 3

sercmd compiles and installs as kamcmd for kamailio
by Daniel-Constantin Mierla 24 Oct '12

24 Oct '12

Hello, short note to let everyone know that sercmd tool compiles and installs as kamcmd for flavour kamailio as of today in the master git branch. There was a conflict when trying to install both flavours from packages, because of overlapping sercmd file. I added a basic manpage for the tool, a recommendation for each binary app in many linux distros. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Registering Phones
by Re Alvarez 24 Oct '12

24 Oct '12

Hi I am setting Kamailio as SBC in my organization using Freeswitch as PBX. The integration between the two systems is working perfectly and now I am trying to add some CISCO 7970 phones to the mix. Here's a summary of the setup 1. Kamailio has 2 NICs (1 public, 1 private), RTPproxy is running 2. Freeswitch is on private network. 3. All the CISCO phones are also on the private network. We will never have a SIP phone outside our network, My question, in this case. Should I register my phones to Kamailio or Freeswitch. I am new to Kamailio and still getting down with the SIP concepts. I would prefer to register the phones with Freeswitch and only want Kamailio to get involved when calls are leaving/entering the network. If two users, registered to the same freeswitch server call each other, Kamailio should not have to do anything (ideally, not sure if its possible with SBC setup). Thanks Re

1 0

getting siptrace error on start up
by Christopher Holland 23 Oct '12

23 Oct '12

getting this error: kamailio: ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter <trace_to_database> not found in module <siptrace> parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 63, column 46: Can't set module parameter I have set this to modparam("siptrace", "trace_to_database", "0") in my .cfg /I didn't compile siptrace when I compiled kamailio. I don't know how to add a module after compiling and installing kamailio. I haven't been able to find any documentaion on how to add a module after installing only compiling before installing. Thanks all for your help Christopher /

3 3

Kamailio Hosted Solutions
by Satyajeet Thokchom 23 Oct '12

23 Oct '12

I am looking for hosted Kamailio Solutions for my VOIP calling app in android and iphone. I have the app ready and have even setup kamailio server in our rackspace cloud but there is many problems like one way calling, noisy calls, no call connection etc. (application VOIP library , we are using IMSdroid and Idoubs , based on doubango SIP) I am looking for a hosted solutions affordable for me - with a support for 300 concurrent calls at a time. Best Regards, Satyajeet Thokchom Business Manager, NGBM Solutions Ltd Phone: India:+91-11-25541320 Mobile: 91 7428565264 Website: www.ngbm.net Web Support: www.ngbmsolutions.com Note: "This e-mail is confidential and may also be privileged. If you are not the intended recipient, please notify us immediately; you should not copy or use it for any purpose, nor disclose its contents to any other person"

1 0

turning off rev. dns
by Christopher Holland 22 Oct '12

22 Oct '12

I'm setting up Homer with kamailio for sip capturing. I've got it all set up and am getting this error. I have one interface for packet capture on a mirroired port and one interface for managment. I'm thinking that I have to only have kamailio listen on the packet capture interface and not the managment interface. Would that resolve this issue? loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(1832) WARNING: <core> [socket_info.c:1406]: WARNING: fix_hostname: could not rev. resolve 192.168.111.88 Listening on udp: 127.0.0.1:5060 udp: 192.168.111.88:5060 Aliases: udp: localhost:5060 config file ok, exiting... sense it's a local address it's not possible to do a rev. resolve of that address. Is there a way to turn off the rev. resolve. I've looked at this in the trouble shooting faq. How do I avoid Reverse-DNS-Lookups? A: Simply not use src_ip = "11.22.33.44" and use instead src_ip = 11.22.33.44! We had in our installation some serious trouble with requests coming from systems, where the hostname could not be reverse resolved (the DNS-Server responsible for this IP-Range was poorly configured and would not answer at all).

3 3

Remote Commands
by Ed Brady 22 Oct '12

22 Oct '12

Hi, I am trying to get a company who handles our billing and account provisioning connected to our Kamailio instance. I understand that XMLRPC is likely the best method for doing this, but I am unable to find direct commands equivalent to "kamctl add" or "kamctl rm". Is there a resource that lists the MI or XMLRPC commands, or do I need to create even simple commands? Thanks, Ed

2 4

Hangup call on leg B side
by pars3c 22 Oct '12

22 Oct '12

Hi , i don't know how to realize this: A --- INVITE---->proxy ----INVITE----> B <--- 200 OK ---BYE----> ---INVITE ----->C <--- 200 OK proxy <--- 200 OK -------ACK ---> -------ACK ---> When B answer the call, the proxy should immediately hang up the call and make a new invite. For user A is always the same call. I tried this : onreply_route { ################################ connect if (is_method("INVITE") && status == "200") { dlg_bye("callee"); drop(); } } But it doesn't work. When arrive 200 , the proxy drop the message to the user a , wait for a while, and then send a cancel to b (it should send a BYE, instead send a CANCEL, probably because it not have a ACK). How can I try this? Thanks

2 1

Kamailio 3.1.5 crashing during TLS connection setup
by Kamal Palei 22 Oct '12

22 Oct '12

Dear All I have setup Kamailio 3.1.5 with TLS enabled. Whenever I receive the REGISTER request from endpoint, trying to forward to next proxy. For that I have added below code in kamailio.cfg * if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:5061"); exit(); } *I see once REGISTER received at Kamailio proxy, it is sending TCP synch packet to next proxy, next proxy sends back a synch+reset packet. Then kamailio proxy is crashing with log as below. *root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting * The complete log is given as below. Please let me know how to get out of this issue. I used -m 64 option while starting proxy, but same result. Thanks, Kamal, NECS, Bangalore * 0(2336) INFO: <core> [tcp_main.c:4730]: init_tcp: using epoll_et io watch method (config) 0(2338) INFO: usrloc [hslot.c:53]: locks array size 512 0(2338) INFO: tls [tls_init.c:511]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, compression: on 0(2338) INFO: tls [tls_init.c:519]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, zlib compression: off compiler: gcc -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -DL_ENDIAN -DTERMIO -O3 -fomit-frame-pointer -Wall -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM 0(2338) INFO: tls [tls_init.c:373]: tls: init_tls: disabling compression... 0(2338) WARNING: tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 5242880 and 2621440 bytes 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 5242880 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2621440 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: tls [tls_domain.c:176]: TLSs<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSs<default>: certificate='/etc/certs/pocserver.com/cert.pem' 0(2338) INFO: tls [tls_domain.c:195]: TLSs<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSs<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSs<default>: require_certificate=0 0(2338) INFO: tls [tls_domain.c:213]: TLSs<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSs<default>: private_key='/etc/certs/pocserver.com/key.pem' 0(2338) INFO: tls [tls_domain.c:224]: TLSs<default>: verify_certificate=0 0(2338) INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:544]: TLSs<default>: No client certificate required and no checks performed 0(2338) INFO: tls [tls_domain.c:176]: TLSc<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSc<default>: certificate='(null)' 0(2338) INFO: tls [tls_domain.c:195]: TLSc<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSc<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSc<default>: require_certificate=1 0(2338) INFO: tls [tls_domain.c:213]: TLSc<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSc<default>: private_key='(null)' 0(2338) INFO: tls [tls_domain.c:224]: TLSc<default>: verify_certificate=1 0(2338) INFO: tls [tls_domain.c:227]: TLSc<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:529]: TLSc<default>: Server MUST present valid certificate 2(2340) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 1(2339) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 4(2342) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 3(2341) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 7(2345) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_et io watch method (config) root@B2BUA:/usr/local/src/scripts# 9(2347) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 8(2346) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting 0(2338) ALERT: <core> [main.c:742]: child process 2347 exited by a signal 6 *

2 2

Listen on all IPv6 addresses
by Anton Kvashenkin 22 Oct '12

22 Oct '12

Hello guys. In the Kamailio v.3.3.0 release notes http://www.kamailio.org/w/kamailio-v3-3-0-release-notes/ it says: all IPv6 network interfaces are auto-detected and Kamailio start listening on them (if no strict listening rules are set) But it doesn't happen. version: kamailio 3.3.1 (x86_64/linux) d75cb5 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d75cb5 listen is commented out @ cfg

2 1

Re: [SR-Users] rtpproxy delete fails
by Juha Heinanen 22 Oct '12

22 Oct '12

the patch below adds 't' flag to rtpproxy_destroy flags. if 't' flag is present when rtpproxy_destroy is called, to tag is not included in D command sent to rtpproxy thus causing destroy of the full call. 't' flag can be used to avoid piling up of unused rtpproxy calls by making it possible to call rtpproxy_destroy when 200 ok is received from a branch where rtpproxy is not used. is it ok for me to commit this patch or do long term rtpproxy users have a better idea? it would still make sense to include a timer in rtpproxy that cleans up un-used calls at an interval that corresponds to sip proxy's invite timeout. -- juha *** /usr/src/orig/sip-router/modules/rtpproxy/rtpproxy.c 2012-10-19 14:31:27.658485534 +0300 --- modules/rtpproxy/rtpproxy.c 2012-10-20 09:52:10.308389743 +0300 *************** *** 1678,1683 **** --- 1678,1684 ---- str callid, from_tag, to_tag, viabranch; char *cp; int via = 0; + int to = 1; int ret; struct rtpp_node *node; struct iovec v[1 + 4 + 3 + 2] = {{NULL, 0}, {"D", 1}, {" ", 1}, {NULL, 0}, {NULL, 0}, {NULL, 0}, {" ", 1}, {NULL, 0}, {" ", 1}, {NULL, 0}}; *************** *** 1700,1705 **** --- 1701,1710 ---- via = 1; break; + case 't': + case 'T': + to = 0; + break; case 'a': case 'A': case 'i': *************** *** 1736,1742 **** return -1; } to_tag.s = 0; ! if (get_to_tag(msg, &to_tag) == -1) { LM_ERR("can't get To tag\n"); return -1; } --- 1741,1748 ---- return -1; } to_tag.s = 0; ! to_tag.len = 0; ! if ((to == 1) && get_to_tag(msg, &to_tag) == -1) { LM_ERR("can't get To tag\n"); return -1; }

2 1

mediaproxy-ng fails to parse start_recording() command
by Juha Heinanen 22 Oct '12

22 Oct '12

i tried what happens when i call start_recording() after calling rtpproxy_manage() and got this kind of error to syslog: Oct 20 19:57:43 siika mediaproxy-ng[8423]: Failed to properly parse UDP command line '8554_13 R ngsgznnwgzpuovt(a)siika.tutpro.com mgpwe' from 127.0.0.1:51497, using fallback RE why the error? it is so that mediaproxy-ng does not support recording? -- juha

4 3

Re: [SR-Users] dangers of excluding 100rel?
by Alex Balashov 19 Oct '12

19 Oct '12

You can strip 100rel as long as it's Supported. However, not so good things happen if you strip it from Required. :-) -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Andrew Mortensen <admorten(a)isc.upenn.edu> wrote:Can I safely strip 100rel support from INVITEs sent by callers registered to my proxy? We're using proxy-generated 302 redirects for call forwarding in our deployment. In general this works better for us than appending branches, as we rapidly ran into branch limits when dealing with large multiple appearance lines. However, we've run into a case where the caller, a Polycom 550, seems to get confused when a parallel forked call is redirected, 100rel is in use and the redirection target is also a multiple appearance line. The caller is correctly responding with PRACKs to all 180s, before and after the redirection. However, hanging up the call on the caller after the redirection, but before any remote party has answered, does *not* cause the caller to send a CANCEL to tear down the call. In fact, the caller doesn't send anything at all. The remote parties continue to ring until the ring timer expires, and the proxy sends CANCELs to all callee endpoints. That's clearly a Polycom bug, and we'll need to work with them to get it resolved. In the meantime, I'm looking for workarounds. One quick, crude option is to strip 100rel from the Supported header in the caller's INVITE. I've tested this, and it does solve the problem, but I'm concerned about side effects. I'm aware of the value 100rel has for calls to the PSTN, but in this case the calls never leave the local network. Are there situations where 100rel is critical for a parallel forked call? andrew _______________________________________________ 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

dangers of excluding 100rel?
by Andrew Mortensen 19 Oct '12

19 Oct '12

Can I safely strip 100rel support from INVITEs sent by callers registered to my proxy? We're using proxy-generated 302 redirects for call forwarding in our deployment. In general this works better for us than appending branches, as we rapidly ran into branch limits when dealing with large multiple appearance lines. However, we've run into a case where the caller, a Polycom 550, seems to get confused when a parallel forked call is redirected, 100rel is in use and the redirection target is also a multiple appearance line. The caller is correctly responding with PRACKs to all 180s, before and after the redirection. However, hanging up the call on the caller after the redirection, but before any remote party has answered, does *not* cause the caller to send a CANCEL to tear down the call. In fact, the caller doesn't send anything at all. The remote parties continue to ring until the ring timer expires, and the proxy sends CANCELs to all callee endpoints. That's clearly a Polycom bug, and we'll need to work with them to get it resolved. In the meantime, I'm looking for workarounds. One quick, crude option is to strip 100rel from the Supported header in the caller's INVITE. I've tested this, and it does solve the problem, but I'm concerned about side effects. I'm aware of the value 100rel has for calls to the PSTN, but in this case the calls never leave the local network. Are there situations where 100rel is critical for a parallel forked call? andrew

1 0

rtpproxy_destroy([flags]) does not accept flags
by Juha Heinanen 19 Oct '12

19 Oct '12

according to rtpproxy module readme, rtpproxy_destroy function takes optional flags param: 5.4. rtpproxy_destroy([flags]) however, the code does not support it: {"unforce_rtp_proxy", (cmd_function)unforce_rtp_proxy_f, 0, 0, 0, ANY_ROUTE}, {"rtpproxy_destroy", (cmd_function)unforce_rtp_proxy_f, 0, 0, 0, ANY_ROUTE}, what has happened? -- juha

2 3

MSILO Configuration
by Ed Brady 19 Oct '12

19 Oct '12

Hi, I am having trouble configuring MSILO. I copied the example config into my kamailio.cfg file and edited the areas that apply to my domain, but when restarting Kamailio I am faced with an error. Below is the error followed by my complete config file. Please advise what I am doing wrong. Is there some sort of resource for HowTos on Kamailio? The module documentation seems very thorough but the examples at the end seem to only be useful if that module is the only one you need to configure. I am trying to do a fairly straight forward configuration I think (SIP messaging with offline message storing and a single incoming/outgoing gateway that is used for all accounts that do not locally exist), but I am unable to find straight forward documentation on how to configure something like this. Please let me know what I am missing. Thanks, Ed Not starting Kamailio: invalid configuration file! > -e > 0(30863) INFO: tls [tls_init.c:377]: tls: init_tls: disabling > compression... > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: syntax error > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: bad command > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: bad command > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 878, column 12: syntax error > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 878, column 12: > ERROR: bad config file (5 errors) > *Config* #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_TLS #!define WITH_NAT #!define WITH_ACCDB # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:R4U8oO8Er872sNHr@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "cfgutils.so" loadmodule "msrp.so" loadmodule "htable.so" loadmodule "msilo.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # -- msilo params -- modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@MYDOMAIN.com") modparam("msilo","contact_hdr","Contact: registrar(a)MYDOMAIN.com :5060;msilo=yes\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","*** User $rU is offline!") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=1800;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } 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()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # Route to MSILO 10-18-12 route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay(); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } event_route[msrp:frame-in] { xdbg("============#[[$msrp(method)]]===========\n"); xdbg("============*[[$si:$sp]]\n"); xdbg("============ crthop: [$msrp(crthop)]\n"); xdbg("============ prevhop: [$msrp(prevhop)]\n"); xdbg("============ nexthop: [$msrp(nexthop)]\n"); xdbg("============ firsthop: [$msrp(firsthop)]\n"); xdbg("============ lasthop: [$msrp(lasthop)]\n"); xdbg("============ prevhops: [$msrp(prevhops)]\n"); xdbg("============ nexthops: [$msrp(nexthops)]\n"); xdbg("============ srcaddr: [$msrp(srcaddr)]\n"); xdbg("============ srcsock: [$msrp(srcsock)]\n"); xdbg("============ sessid: [$msrp(sessid)]\n"); msrp_reply_flags("1"); if(msrp_is_reply()) { msrp_relay(); exit; } # handle AUTH MSRP requests if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } # frame for local server - send Use-Path # -- passwd can be loaded from DB based on $au $var(passwd) = "xyz123"; if(!pv_www_authenticate("myrealm", "$var(passwd)", "0")) { if(auth_get_www_authenticate("myrealm", "0", "$var(wauth)")) { msrp_reply("401", "Authorization Required", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } $var(cnt) = $var(cnt) + 1; pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)"); $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr); $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock); # - Use-Path: the MSRP address for server + session id $var(UsePath) = "Use-Path: msrp://127.0.0.1:5060/" + $var(sessid) + ";tcp\r\n"; msrp_reply("200", "OK", "$var(UsePath)"); exit; } if($msrp(method)=="SEND") { if($msrp(nexthops)>1) { msrp_reply("200", "Received"); msrp_relay(); exit; } $var(sessid) = $msrp(sessid); if($sht(msrp=>$var(sessid)::srcaddr) == $null) { # one more hop, but we don't have address in htable msrp_reply("481", "No Such Session"); exit; } msrp_relay_flags("1"); msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)", "$sht(msrp=>$var(sessid)::srcsock)"); msrp_relay(); exit; } msrp_relay(); }

3 2

Re: [SR-Users] Kamailio v3.3.2 Released
by Paul Cupis 18 Oct '12

18 Oct '12

On 16/10/12 22:29, Jon Bonilla (Manwe) wrote: > El Tue, 16 Oct 2012 17:00:03 +0200 > Daniel-Constantin Mierla<miconda(a)gmail.com> escribió: >> >> Kamailio SIP Server v3.3.2 stable release is out. > > Debian and Ubuntu packages are ready. Please find the instructions > below: > > http://www.kamailio.org/dokuwiki/doku.php/packages:debs I'm seeing a problem with the 3.3.2 packages on Debian at the moment. There is a file conflict between the kamailio package and kamailio-mysql-modules preventing the latter from installing. $ dpkg-deb -c /var/cache/apt/archives/kamailio_3.3.2+squeeze1_amd64.deb | grep srdb -rw-r--r-- root/root 74296 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1.0 -rw-r--r-- root/root 28360 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1 -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1 -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so -> libsrdb2.so.1.0 $ dpkg-deb -c /var/cache/apt/archives/kamailio-mysql-modules_3.3.2+squeeze1_amd64.deb | grep srdb -rw-r--r-- root/root 28360 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1.0 -rw-r--r-- root/root 74296 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1 -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1 -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so -> libsrdb1.so.1.0 Regards,

2 3

Kamailio- Redirect message 300 with LCR reroute
by Fatima Chahrour~Vanrise Support 18 Oct '12

18 Oct '12

Dear List, I'd like to get help regarding my case. I have the following script where many thanks to Daniel has helped me in if (is_method("INVITE")) { if (!load_gws(1, $rU, $fu)) { sl_send_reply("502", "Unable To lOad GatEwAyS"); exit;} if(!next_gw()){ sl_send_reply("503", "Unable To fInD a gateWaY"); exit;} while(next_gw()){ km_append_branch(); } sl_send_reply("302","Moved Temporary"); exit; } The problem am facing is that the call is rerouting from the first gateway to the next gateway successfully when I have two gateways. BUT when I have three gateways, and the first two gateways are off, the call is not rerouted to the third gateway. It keeps hitting the seconds gateway and gives request time out at the end Why is that? Thanks in advance, F Chahrour

1 0

Kamailio v3.3.2 RPM Packages
by Daniel-Constantin Mierla 18 Oct '12

18 Oct '12

Hello, for the record, RPM packages of Kamailio v3.3.2 were built for Centos 5.5, Centos 6, RedHat 5, RedHat 6, Fedora 15, Fedora 16, Fedora 17, OpenSuse 11.4 and OpenSuse 12.1, on i586 and x86_64 architectures. They are available on opensuse build service repos, see the links at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Question about compliance with RCS/OMA Simple Presence
by Sangeeta Shah 17 Oct '12

17 Oct '12

Hello All, I need some guidance on whether the Kamailio presence/presence_xml and integrated xcap module implementations are RCS/OMA Simple presence compliant. Based on a few things I've seen in the code I don't see any handling for the oma presence rule xml extensions. Is there any plan to be RCS/OMA compliant in the new future? I can get SIMPLE PRESENCE with XCAP to work but not with OMA presence rules and resource lists. Thanks, Sangeeta -- Sangeeta Shah

1 0

has_sdp()??
by Juha Heinanen 17 Oct '12

17 Oct '12

while reading rtpproxy module readme, i found this kind of example: if (is_method("INVITE")) { if (has_sdp()) { if (rtpproxy_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } if (is_method("ACK") && has_sdp()) rtpproxy_answer(); does has_sdp() function really exist? it is not listed on wiki page: http://www.kamailhttp://www.kamailio.org/wiki/alphaindexes/3.4.x/modfunctio… -- juha

4 13

make option to clean the installed binaries
by Ajay Sabat 17 Oct '12

17 Oct '12

Hi, I am a newbie here and recently started using kamailio/ser. I am wondering is there a makefile option to cleanup the installed binaries and library files. Thanks, Ajay

2 3

how to search previous messages in the forum
by Ajay Sabat 17 Oct '12

17 Oct '12

Hi, Many times I get doubts and questions about Kamailio that I want to search in the SR-Users forum before posting a new message, as it might have already been answered in the past. Is there a easier way to do that than looking up the subject lines in the monthly archive? Thanks, Ajay

2 1

Errors issued by Kamailio
by Sangeeta Shah 17 Oct '12

17 Oct '12

Hi, I am seeing the following errors in syslog: Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence_xml [notify_body.c:635]: while extracting tuple node Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence_xml [notify_body.c:75]: while constructing offline body Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence [presentity.c:1220]: preparing body Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence [publish.c:171]: Marking presentity Oct 8 16:21:55 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event message-summary Oct 8 16:21:55 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event presence.winfo Oct 8 16:21:57 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event presence I am using Kamailio 3.3 with Presence and RLS with integrated XCAP. The only significant change i made today is update the presence and RLS config to DB ONLY mode. Thanks, Sangeeta -- Sangeeta Shah

2 9

$fs syntax
by Juha Heinanen 17 Oct '12

17 Oct '12

wiki tells: $fs - reference to the forced socket for message sending (if any) in the form proto:ip:port should it be [proto:]ip:port instead? -- juha

3 7

Kamailio 3.2.4 presence
by Vitaliy Aleksandrov 17 Oct '12

17 Oct '12

Hi all, I'm facing problems with active subscription restoration after sipproxy restart. db_mode is 0 (DB_MEMORY_ONLY) and according to comments in the presence.h kamailio holds subscription in memory and periodically updates to db, but retrieves from db only at startup. And it's true, I see that kamailio tries to update the active_watchers table, but it never inserts new subscriptions there. There are no problems with access to the database since when db_mode is 2 (DB_ONLY) kamailio inserts new records into the active_watchers without any problems. Here are presence module settings: modparam("presence", "db_url", "sqlite:////var/db/presence.db") modparam("presence", "db_mode", 0) modparam("presence", "timeout_rm_subs", 1) modparam("presence", "pres_htable_size", 8) modparam("presence", "subs_htable_size", 8) modparam("presence", "max_expires", 3600) modparam("presence", "db_update_period", 60) modparam("presence", "clean_period", 60) Does anybody have success with db_mode = DB_MEMORY_ONLY and active_watchers db table ?

2 2

is kamilio support RFC5626 now ?
by mike 16 Oct '12

16 Oct '12

dear kamilio developers : i read the kamilio 3.3.0' s new features , it say it support outbound , and i have place to handle reg-id and sip.instance . so , i wonder if the kamilio3.3.0 support RFC5626 now，so that the kamilio can be used as a edge proxy . i ask why because i can't find where kamilio add flow-token and path for the register request. thanks

5 8

IPv6 addresses in dispatcher on IPv4 only proxy
by Alex Hermann 16 Oct '12

16 Oct '12

Hello, I am using the dispatcher module with a database table shared among multiple proxies. Some proxies do both IPv4 and IPv6, others only do IPv4. The problem is when i use an IPv6 address in the dispatcher table. Then the IPv4-only proxies fail to load the table. ERROR: dispatcher [dispatch.c:325]: could not resolve [IPv6 address] WARNING: <core> [mem/f_malloc.c:474]: WARNING:fm_free: free(0) called ERROR: dispatcher [dispatcher.c:321]: could not initiate a connect to the database ERROR: <core> [sr_module.c:889]: init_mod(): Error while initializing module dispatcher (/usr/lib/kamailio/modules_k/dispatcher.so) Is there any setting i can use on the IPv4-only proxies to make it skip the IPv6 records? (apart from listening on an IPv6 address) -- Greetings, Alex Hermann

2 9

Error while running TLS enabled Kamailio proxy
by Kamal Palei 16 Oct '12

16 Oct '12

Dear All While executing TLS enabled Kamailio proxy (3.1.5), getting below error. * 0(15895) INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 0(15895) ERROR: tls [tls_domain.c:393]: TLSs<default>: Unable to load certificate file './modules/tls/ser-selfsigned.pem' 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:02001002:system library:fopen:No such file or directory 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:20074002:BIO routines:FILE_CTRL:system lib 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:140DC002:SSL routines:SSL_CTX_use_certificate_chain_file:system lib 0(15895) ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module tls (/usr/local/lib/kamailio/modules/tls.so) 0(15895) ERROR: <core> [main.c:1532]: ERROR: main: error in init_child(PROC_INT) -- exiting 0(15895) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting 0(15893) ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe * I performed below steps to compile and install. 1. I had a working Kamailio-3.1.5 code base without TLS 2. Downloaded latest openssl source code 3. Compiled and generated libssl.a and libcrypto.a 4. Next, I recompiled kamailio proxy, compilation did not go through. I had to modify Makefile in modules/tls module, added path for openssl/bio.h and path for libssl.a 5. Then executed make all clean and make all include_modules=tls , this time compilation succeeded 6. Next I did make install 7. Generated certificates using openssl library that comes by default with ubuntu 10.04 lte 8. Added below entries in kamailio.cfg #!define WITH_TLS #!ifdef WITH_TLS loadmodule "tls.so" modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") modparam("tls", "private_key", "/etc/certs/localb2bua.com/key.pem") modparam("tls", "certificate", "/etc/certs/localb2bua.com/cert.pem") modparam("tls", "ca_list", "/etc/certs/demoCA/cert.pem") enable_tls=yes #!endif 9. Then I tried running kamailio proxy It complained not getting tls.so and tls.cfg Copied these files from modules/tls to /usr/local/etc/kamailio/ path 10. Then I was able to run Kamailio proxy and got the errors pasted at top of email (* Unable to load certificate file './modules/tls/ser-selfsigned.pem') * Please guide me what should I do to avoid this error. Thanks a lot. -Kamal * *

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unsubscribe
by Pablo R. Digonzelli 16 Oct '12

16 Oct '12

Ing. Pablo Digonzelli Sofware Solutions IP Soluciones SRL 25 de Mayo 521. Entrepiso A email: pdigonzelli(a)softsargentina.com email: pdigonzelli(a)gmail.com twitter: @pdigonzelli Tel: 0381 4227378 Cel: 0381 155982714

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Re: [SR-Users] Kamailio 3.3.0 CDR logging call "duration"
by Daniel-Constantin Mierla 16 Oct '12

16 Oct '12

Hello, do not reply to private email only when it is about a discussion on the mailing list - such messages will be ignored. Try to store the value in an avp when processing the BYE, after loose_route(), and store that avp in acc. Daniel On 10/15/12 1:13 PM, Nord7 wrote: > Hello, > no I'm not getting error messages. > Problem: > I can not get kamailio record duration dialogue created field "duration". > I make a test call, answer it, wait 5-6 seconds and hang up. > Now I have to get 5-6 seconds duration in the "duration", but the > field is blank. The rest of the field information about the call is > entered correctly. > So the variable $ DLG_lifetime information about the duration of the > call does not get. > But most of all I am wrong somewhere. > > > 2012/10/15 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > do you get any error message or what is the problem you encounter? > > Cheers, > Daniel > > > On 10/15/12 10:14 AM, Nord7 wrote: >> Hello >> >> Kamailio 3.3.0 >> All default "extra fields" created in database. I create in >> Database new field "duration" and trying write in this field >> variable $DLG_lifetime (from dialog module) , but can't do this. >> What i'm doing wrong? >> >> installed modules: ACC, DIALOG >> # ----- acc params ----- >> /* what special events should be accounted ? */ >> modparam("acc", "early_media", 0) >> modparam("acc", "report_ack", 0) >> modparam("acc", "report_cancels", 0) >> modparam("acc", "detect_direction", 0) >> /* account triggers (flags) */ >> modparam("acc", "log_level", 1) >> modparam("acc", "log_flag", FLT_ACC) >> modparam("acc", "log_missed_flag", FLT_ACCMISSED) >> modparam("acc", "log_extra", >> "src_user=$fU;src_domain=$fd;src_ip=$si;" >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" >> "duration=$DLG_lifetime") >> modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) >> /* enhanced DB accounting */ >> modparam("acc", "db_flag", FLT_ACC) >> modparam("acc", "db_missed_flag", FLT_ACCMISSED) >> modparam("acc", "db_url", DBURL) >> modparam("acc", "db_extra", >> "src_user=$fU;src_domain=$fd;src_ip=$si;" /* extra fields >> created in DataBase */ >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" >> "duration=$DLG_lifetime") >> modparam("acc", "cdr_enable", 1) >> modparam("acc", "cdr_start_on_confirmed", 0) >> modparam("acc", "cdr_facility", "LOG_DAEMON") >> >> # ----- dialog params ----- >> modparam("dialog", "enable_stats", 1) >> modparam("dialog", "hash_size", 4096) >> modparam("dialog", "rr_param", "did") >> modparam("dialog", "dlg_flag", DLG_FLAG) /* DLG_FLAG=4 */ >> modparam("dialog", "default_timeout", 21600) >> modparam("dialog", "timeout_avp", "$avp(i:10)") >> modparam("dialog", "dlg_match_mode", 0) >> modparam("dialog", "detect_spirals", 1) >> modparam("dialog", "db_url", DBURL) >> modparam("dialog", "db_mode", 1) >> modparam("dialog", "db_fetch_rows", 500) >> >> >> ########Routing Logic######## >> >> 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"); >> } >> setflag(DLG_FLAG); # flag 4 duration >> setflag(FLT_ACC); # do accounting ... >> setflag(FLT_ACCFAILED); # ... even if the transaction fails >> if (!t_relay()) { >> sl_reply_error(); >> } >> exit; >> } >> >> >> >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@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://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

segfault in siptrace module using setflag?
by samuel 16 Oct '12

16 Oct '12

Hi folks, I was making some tests with kamailio (both 3.3.0 and 3.3.1 tgz) with siptrace module sending "homer" packets to a remote central agent where the data is stored and presented to the user. The related config is: # ----- siptrace params from HOMER ----- modparam("siptrace", "duplicate_uri", "sip:A.B.C.D:9060") modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #HEPv2 == timestamp will be included to HEP header modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_capture_id", 1) basically for all incoming messages, a route is called with the following actions: route[HOMER] { setflag(22); sip_trace(); } I've been making some tests and if I disable the setflag action, there is no problem (well, no transaction messages are sent to the central agent), but as soon as I activated setflag, kamailio crashes with the following error: <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 28 There's no concrete point in the config file where it fails so I think the problem is more tm-internals related. I can provide further information in case you need it. Just for adding information, and without any intention to start a flame-war, with opensips 1.8.0-tls the server does not crash... Thanks in advance! Samuel.

3 14

Supporting TLS and DTLS in RTP Proxy
by Kamal Palei 16 Oct '12

16 Oct '12

Hi All I am planning to enhance RTP proxy to support TLS and DTLS. We have some requirements where we need to send RTP packets either over TLS or over DTLS. I really do not know where to start :-) Few things that I have thought over is 1. I will take opnssl code base and integrate with rtpproxy code. 2. Will refer kamailio proxy code how tls connection framework is there and try to copy the code from kamailio code base to rtp proxy code base, OR is there a better alternative here. If somebody has a better idea, kindly share. Thanks a lot. -kamal

5 6

Guidance on using Kamailio as Presence Server and Scalability issues for millions of subsribers
by Thomas Hoffman 16 Oct '12

16 Oct '12

We're trying to determine how best to design/architect a solution where we use Kamailio as a Presence/XCAP server and be scalable to support millions of subscribers. I see the Dispatcher module, but it appears that requires having dedicated instances of the server running just in Dispatcher mode - then you can load balance across the actual instances of the presence servers. It's not clear if a server can be configured to work as a dispatcher and a presence server at the same time. Does anyone have any high-level guidance how to approach this problem? Thank you, Tom Hoffman ______________________________________________________________________________________________________________________________ This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify so to the sender by e-mail and delete the original message. In such cases, please notify us immediately at sanchar-sadhan(a)infinite.com . Further, you are not to copy, disclose, or distribute this e-mail or its contents to any unauthorized person(s) .Any such actions are considered unlawful. This e-mail may contain viruses. Infinite has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachments. Infinite reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infinite e-mail system. ***INFINITE******** End of Disclaimer********INFINITE********

2 1

ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1)
by MARINA SERRANO MONTES 16 Oct '12

16 Oct '12

Hi, My kamailio installation usually works right, but sometimes, it crashes and it shows the following error: "ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) " Do you know why it happens randomly or what's the cause ? Many thanks. Br, Marina ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx

2 1

preparing kamailio 3.3.2
by Daniel-Constantin Mierla 16 Oct '12

16 Oct '12

Hello, I am considering packaging Kamailio v3.3.2 at the beginning of next week (Monday or Tuesday). If any of the devs have backports to be done, do them before the end of this week. Also, if someone spotted some fixes in the master that were not backported, write to the lists so we can care of them. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 2

Delay message
by pars3c 15 Oct '12

15 Oct '12

Hi, how can i delay sending of a message ? A ----> proxy (after x seconds) ---> B If I use the sleep function of the cfgutils module,I block all the calls for the sleep time or the single one? Thanks

3 4

Error on compiling Kamailio with TLS support in Ubuntu 10.04LTE
by Kamal Palei 15 Oct '12

15 Oct '12

Hi All I am trying to compile Kamailio - 3.1.5 in Ubuntu 10.04LTE . Tried below comands one by one make -C modules/tls make modules modules=modules/tls make all include_modules=tls But always getting the error *In file included from tls_bio.c:31: tls_bio.h:32:25: warning: openssl/bio.h: No such file or directory * It looks that openssl is not installed in machine. I tried installing openssl as belowB sudo apt-get install libssl-dev But it fails always. The error it gets is *root@B2BUA:/usr/local/src/kamailio-3.1.5# sudo apt-get install libssl-dev sudo: unable to resolve host B2BUA Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: libssl-dev: Depends: libssl0.9.8 (= 0.9.8k-7ubuntu8) but 0.9.8k-7ubuntu8.8 is to be installed E: Broken packages* * root@B2BUA:/usr/local/src/kamailio-3.1.5# * Has anybody faced this issue previously. Whats the workaround here, if somebody knows please share. Thanks Kamal

2 1

Re: [SR-Users] R: Delay message
by Alex Balashov 15 Oct '12

15 Oct '12

You can't, exactly. But you can t_continue() inside an rtimer route that fires at a given interval. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/pars3c <pars3c(a)gmail.com> wrote:With t_suspend of tmx module, how can I set seconds to sleep? -----Messaggio originale----- Da: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Per conto di Alex Balashov Inviato: lunedì 15 ottobre 2012 15:52 A: sr-users(a)lists.sip-router.org Oggetto: Re: [SR-Users] Delay message It sounds like you're after the recent asynchronous TM innovations: http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2535968 http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2525882 On 10/15/2012 09:49 AM, pars3c wrote: > Hi, > > how can i delay sending of a message ? > > A ----> proxy (after x seconds) ---> B > > If I use the sleep function of the cfgutils module,I block all the > calls for the sleep time or the single one? > > Thanks > > > > _______________________________________________ > 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 > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.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 ----- Nessun virus nel messaggio. Controllato da AVG - www.avg.com Versione: 2012.0.2221 / Database dei virus: 2441/5332 - Data di rilascio: 14/10/2012 _______________________________________________ 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 3.3.0 CDR logging call "duration"
by Nord7 15 Oct '12

15 Oct '12

Hello Kamailio 3.3.0 All default "extra fields" created in database. I create in Database new field "duration" and trying write in this field variable $DLG_lifetime (from dialog module) , but can't do this. What i'm doing wrong? installed modules: ACC, DIALOG # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_level", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$DLG_lifetime") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" /* extra fields created in DataBase */ "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$DLG_lifetime") modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 0) modparam("acc", "cdr_facility", "LOG_DAEMON") # ----- dialog params ----- modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", DLG_FLAG) /* DLG_FLAG=4 */ modparam("dialog", "default_timeout", 21600) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 0) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "db_fetch_rows", 500) ########Routing Logic######## 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"); } setflag(DLG_FLAG); # flag 4 duration setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails if (!t_relay()) { sl_reply_error(); } exit; }

2 1

DTLS support in Kamailio
by Kamal Palei 15 Oct '12

15 Oct '12

Hi List Is DTLS supported in Kamailio? If not, can somebody give me initial idea/pointer how can we support it with minimal code change. Best Regards Kamal

2 1

Re: [SR-Users] Get sip clients to work with sip:user@domain and ldap
by Christophe ROY 15 Oct '12

15 Oct '12

Hello I have created a SRV record for sip.domain.tld pointing to our kamailio server, the domain by itself resolves to our DNS IP. Regarding the sockets: root@kamailio1:~# netstat -lnp | grep kamailio tcp 0 0 192.168.14.25:5060 0.0.0.0:* LISTEN 4752/kamailio unix 2 [ ACC ] STREAM LISTENING 17100 4745/kamailio /tmp/kamailio_ctl Thanks 2012/10/11 Daniel-Constantin Mierla <miconda(a)gmail.com> > Hello, > > I recommend using 3.3.x if you start a new deployment. > > Is the domain resolvable by DNS? What are the sockets kamailio is > listening on? > > Cheers, > Daniel > > On 10/10/12 5:21 PM, Christophe ROY wrote: > > Hello > > I'm quite new to SIP and TOIP so I may not understand the concepts > correctly. > I'm supposed to make work SIP clients (for now, linphone on android and > windows) with a Kamailio 3.2 server and ldap authentication. > > Ldap authentication seems to be working fine, as the login/pass is > validated against the Active Directory before registering the SIP client. > However, Linphone seems to register the SIP client as > sip:user@true_ip_address instead of sip:user@domain. > > If I try to make a call to sip:anotheruser@domain it doesn't work, and I > can find in /var/log/syslog the following lines: > > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [ut.h:333]: > no corresponding socket for af 2 > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm > [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding > listening socket) > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm > [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: sl > [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server > error occurred (7/SL) > > However, if I try to make a direct call to sip:anotheruser@his_true_IP, > it works > > How can I make the registered users to be reachable with user@domain with > Kamailio? > And by the way, is there a way to get a list of registered users? > > Thank you for the help > > Christophe > > > _______________________________________________ > 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 Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu > > > _______________________________________________ > 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 > >

3 4

Wesip Netbeans
by pars3c 15 Oct '12

15 Oct '12

Hi , anyone know if it is possible to use Wesip with Glassfish instead of Tomcat? I'm new in using Glassfish and sip servlet, and I don't know how to start the sip servlet. In tomcat there is the file server.xml, and the connector . How "say" to glassfish to start a listener on sip packet ? If it's not possible to use Glassfish, how i can open a sar project with netbeans e debug it? Thanks to all

1 0

update_subs_rlsdb: NO MATCH
by Sangeeta Shah 12 Oct '12

12 Oct '12

I have Kamailio 3.3 with Presence, RLS and integrated XCAP. I see this information message in the log file: Oct 8 14:34:05 RCS-Presence /usr/local/sbin/kamailio[4530]: INFO: rls [rls_db.c:779]: update_subs_rlsdb: NO MATCH Anyone know why the application is calling that method if it doesn't exist in rls_db.c. I still haven't been able to get presence with rls and integrated xcap working so not sure if this is an issue or no. Currently for both presence and RLS i am using "DB-Only" mode. Any help is appreciated. It seems like when client A adds client B as a contact, the kamalio server is not sending out an auth request to client B. Even though it's registered for presence.winfo events. Thanks, Sangeeta -- Sangeeta Shah

2 6

TLS support in Kamailio
by Kamal Palei 12 Oct '12

12 Oct '12

Hi List I have a specific requiremnt where I want Kamailio proxy need to work as a intermideiate proxy . The endpoints will talk to Kamailio using UDP as transport and Kamailio should interact with main proxy using TLS. Is this scenario possible with Kamailio ? If so, can you give me some pointer how can I acheive it. Thanks Kamal Net clooud systems Bangalore

3 5

Kamailio Group voice Call
by Vijay Thakur 12 Oct '12

12 Oct '12

All Experts, I have setup kamailio 3.1.5 server and it is working fine in my production environment. I want to implement group voice calls (audio conferencing). How can i do this. Kindly guide me. With Regards, Vijay Thakur

3 2

Skype-like service in less than one hour
by శ్రీధర్ 12 Oct '12

12 Oct '12

I set up my server following the tutorial at http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour, with the latest kamailio 3.3, and updated the config file accordingly. After the reboot the sip server is not running here are the logs. When I do 'sudo /etc/init.d/kamailio start' the server starts, but has problems communicating with the rtpproxy, the log for the same is also below. Please advice if any changes needed to make this working with the latest version Thanks S $ cat /var/log/syslog | grep kamailio Oct 12 01:07:48 sips kamailio: INFO: tls [tls_init.c:377]: tls: init_tls: disabling compression... Oct 12 01:07:48 sips kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: usrloc [hslot.c:53]: locks array size 512 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: tls [tls_init.c:541]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, compression: on Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: tls [tls_init.c:549]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, zlib compression: on#012 compiler: cc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Wformat-security -Werror=format-security -D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack -Wall -DOPENSSL_NO_TLS1_2_CLIENT -DOPENSSL_MAX_TLS1_2_CIPHER_LENGTH=50 -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: WARNING: tls [tls_init.c:603]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [db.c:320]: could not add connection to the pool Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: usrloc [dlist.c:513]: failed to open database connection Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: registrar [reg_mod.c:472]: failed to register domain Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:623 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:626 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:628 Oct 12 01:54:55 sips kamailio: INFO: tls [tls_init.c:377]: tls: init_tls: disabling compression... Oct 12 01:54:55 sips kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: usrloc [hslot.c:53]: locks array size 512 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_init.c:541]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, compression: on Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_init.c:549]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, zlib compression: on#012 compiler: cc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Wformat-security -Werror=format-security -D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack -Wall -DOPENSSL_NO_TLS1_2_CLIENT -DOPENSSL_MAX_TLS1_2_CIPHER_LENGTH=50 -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: tls [tls_init.c:603]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:187]: TLSs<default>: tls_method=12 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:199]: TLSs<default>: certificate='/etc/kamailio/kamailio-selfsigned.pem' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:206]: TLSs<default>: ca_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:213]: TLSs<default>: crl='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:217]: TLSs<default>: require_certificate=0 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:224]: TLSs<default>: cipher_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:231]: TLSs<default>: private_key='/etc/kamailio/kamailio-selfsigned.key' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:235]: TLSs<default>: verify_certificate=0 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:238]: TLSs<default>: verify_depth=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:576]: TLSs<default>: No client certificate required and no checks performed Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:187]: TLSc<default>: tls_method=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:199]: TLSc<default>: certificate='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:206]: TLSc<default>: ca_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:213]: TLSc<default>: crl='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:217]: TLSc<default>: require_certificate=1 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:224]: TLSc<default>: cipher_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:231]: TLSc<default>: private_key='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:235]: TLSc<default>: verify_certificate=1 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:238]: TLSc<default>: verify_depth=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:561]: TLSc<default>: Server MUST present valid certificate Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily

4 6

log alerting time, progress time connect time
by pars3c 12 Oct '12

12 Oct '12

Hi , i need to register the time when arrive the following messages: invite, alerting, progress , connected and hangup. How can I make it ? it's is possible to store these value in the extra field in the acc module ? Thanks

2 1

TCP error
by Ed Brady 12 Oct '12

12 Oct '12

I am testing connecting an MSRP client to my Kamailio instance, and I keep getting the following error in my syslog: /usr/sbin/kamailio[20055]: ERROR: <core> [tcp_read.c:293]: error reading: Connection reset by peer (104) I believe this must be a misconfiguration on my part, I just don't know where to start. Any ideas what may cause a TCP read error? If this is an uncommon error, I can supply configs to help determine the cause. Thanks, Ed

3 2

Wrong onreply_route is called after serial forking
by Alex Hermann 11 Oct '12

11 Oct '12

Hello, i noticed thet in serial forking, replies to earlier branches arriving after sending out a new branch use the onreply_route of the later branch instead of the onreply_route set before sending the earlier branch.... How to reproduce: 1) set onreply_route to A 2) relay 1st branch 3) 1st branch times out, internal 408 is created 4) tm send CANCEL to 1st branch 5) in failure route, onreply_route is set to B 6) relay 2nd branch 7) 1st branch responds with 487, and goes into reply_route B instead of A I think each branch should take the reply_route which was set before it got relayed and not pick up later changes meant for other branches. How would i fix this? -- Greetings, Alex Hermann

5 7

Question about how the RLS module handle subscribe
by Sangeeta Shah 11 Oct '12

11 Oct '12

Hello All, I am trying to understand how the RLS module handles a subscribe request. My client is sending the following: Subscribe, event = reg, no Allowed header - hence it is getting handled by the presence module (entry make its to active_watchers table) Subscribe, event - presence.winfo, no Allowed header - hence it's getting handled by the presence module (entry makes it to active_watchers table) Publish - handled by presence module (makes it to presentity table) Subscribe, event = presence, Allowed = eventlist - hence it's handled by the rls module. RLS module is having trouble sending a notify per this data in the syslog. Also I don't see any entries in the watcher table or rls_watchers or rls_presentity table. Errors: Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: presence [hash.c:342]: record not found in hash table Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [notify.c:387]: updating in hash table Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [notify.c:273]: in function agg_body_sendn_update Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7fb2a8daa220 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7fb2a8daa598 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7fb2a8da9f90 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x7fb2a8da9b48 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x7fb2a8da9b28 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x7fb2a8daa608 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x7fb2a8daa570 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x7fb2a8da9410 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [subscribe.c:732]: failed sending full state notify Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [subscribe.c:781]: occured in rls_handle_subscribe Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <script>: ========= RLS Handle subscriber returned with ret code: -1 === What's it trying to do here? Any help is appreciated. Thanks, Sangeeta -- Sangeeta Shah

1 0

Kamailio presence with RLS
by Sangeeta Shah 11 Oct '12

11 Oct '12

Hello Everyone, Has anyone had any luck getting RLS working with presence. I am seeing the following errors in my log file: ct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: presence [hash.c:342]: record not found in hash table Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [notify.c:385]: updating in hash table Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [notify.c:271]: in function agg_body_sendn_update Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7fbf415c4178 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7fbf415c48a0 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7fbf415c38a8 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x7fbf415c3bc8 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x7fbf415c4880 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x7fbf415c3c28 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x7fbf415c3648 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x7fbf415c3940 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [subscribe.c:732]: failed sending full state notify Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [subscribe.c:781]: occured in rls_handle_subscribe Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <script>: ========= RLS Handle subscriber returned with ret code: -1 === I think this is my first problem. There is an error later in the log file as well, but it seems Apparently from what i understand we are trying to update the entry in the presence_shtable and we cannot find it and hence we are failing to send the full state notify. Any help is appreciated. If further information is required please let me know. I can try to print out more data in the update_presence_shtable function. I have another htread with a lot of log info but for some reason the thread went cold and I just need to keep moving on getting this to work. Anyone familiar with this please let me know what the likely cause of this issue might be. Thanks, Sangeeta -- Sangeeta Shah

1 0

How can I directly access the domain of additional branches??
by Stephen Dodge (Bistech) 11 Oct '12

11 Oct '12

Hello, Could someone advise if it is possible to directly access the domain of additional branches as $rd references the domain of the main URI only. If it's not possible does anyone have any tips on how I can achieve this. Many Thanks. Steve. ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

4 5

Kamailio direct interconnectivity with PRI
by SamyGo 11 Oct '12

11 Oct '12

Hello, I've a scenario in which I've to deploy a couple Sangoma PRI cards with kamailio. What I wish is that I've some drivers for this purpose and so I don't ned to install FreeSWITCH or Asterisk in between the PRIs and Kamailio. Kindly give any feedback on what are the possibilities and options. Thanks, Sammy

4 7

Get sip clients to work with sip:user@domain and ldap
by Christophe ROY 11 Oct '12

11 Oct '12

Hello I'm quite new to SIP and TOIP so I may not understand the concepts correctly. I'm supposed to make work SIP clients (for now, linphone on android and windows) with a Kamailio 3.2 server and ldap authentication. Ldap authentication seems to be working fine, as the login/pass is validated against the Active Directory before registering the SIP client. However, Linphone seems to register the SIP client as sip:user@true_ip_address instead of sip:user@domain. If I try to make a call to sip:anotheruser@domain it doesn't work, and I can find in /var/log/syslog the following lines: Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) However, if I try to make a direct call to sip:anotheruser@his_true_IP, it works How can I make the registered users to be reachable with user@domain with Kamailio? And by the way, is there a way to get a list of registered users? Thank you for the help Christophe

2 1

kamailio 3.3.1 - uac - remote register - init delay possible?
by René Bormann 11 Oct '12

11 Oct '12

Hello @all, want to use the remote register function of the UAC module. We need to register 42 accounts on one openSER server. At kamailio startup those 42 remote register requests will be startet at the same time. This leads to a kind of FlodAttact on the openSER server of our provider, which will block us :-( We would like to delay each initial remote register by 1 second. We searched the UAC module and taught that at some place the uac.so module has to load the uacreg table and do some for loop to start the remote register process. We found function child_init() in uac.c where the function uac_reg_load_db() is called followed uac_reg_timer(). Our change to uac_reg_timer() is the sleep(1). After compiling it seems to us - that this function is not called. Is there anybody how has a tip for us? Thanks, René /** * */ void uac_reg_timer(unsigned int ticks, unsigned int initial_reg_call) { LM_ERR("----------------------- uac_reg_timer() -- enter \n"); <------>int i; <------>reg_item_t *it = NULL; <------>time_t tn; <------>tn = time(NULL); <------>for(i=0; i<_reg_htable->htsize; i++) <------>{ <------><------>/* free entries */ <------><------>it = _reg_htable->entries[i].byuuid; <------><------>while(it) <------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 1 \n"); <------><------><------>if (initial_reg_call == 1) <------><------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 2 \n"); <------><------><------><------>sleep(1); <------><------><------>} <------><------><------>uac_reg_update(it->r, tn); <------><------><------>it = it->next; <------><------>} <------>} }

3 8

MSRP Config
by Ed Brady 10 Oct '12

10 Oct '12

Hi, I am working on an MSRP project with Kamailio. I have Kamailio installed and configured for SIP with several test accounts created, but am unable to send MSRP messages so far. I am using Blink as a test client, though the final intentions of the project will be to use iDoubs and IMSDroid for client connections. We are having these developed to our needs right now though, so I must use another client to begin testing. I have been unable to find good documentation on the initial configuration of MSRP in Kamailio. I have seen the module documentation, but am unclear how to utilize the information there to configure Kamailio properly. Is there any documentation on getting started with a basic MSRP relay setup so that two clients can chat and send files? I am not new to SIP, but this is my first project with MSRP. Thanks, Ed

4 10

OpenSER/Kamailio Support? - Forwarding SIP INVITE Packets
by Eddie Elric 10 Oct '12

10 Oct '12

Hello fellow OpenSER/Kamailio users, I hope this is the right method of contact - I was unable to find a forum on which to post any queries. In essence, I've set up a Kamailio server of the latest version - logging on to clients works fine, etc. The idea of my setting up was for forwarding SIP messages to our test server, that's where I got stuck. The following link explains everything on Kamailio it seems - http://www.kamailio.org/wiki/cookbooks/3.3.x/core#forward So, I've added to the kamailio.cfg the following - forward("10.0.0.10","5060"); But you see, I'm not sure where I should have added it. I've tried adding to a variety of places, such as - route[SIPOUT] and onreply_route[MANAGE_REPLY] - but none are working. Cheers guys/girls. Best, Eddie

1 1

Kamailio/RTP Proxy issue
by Kamal Palei 10 Oct '12

10 Oct '12

Hi All Facing a strange issue in Kamailio / RTP proxy setup. Using 1. Kamailio - 3.1.5 2. RTP proxy 1.1 I have pne sip client, lets say it is 'A'. 'A' sent INVITE to Kamailio proxy. Kamailio proxy modified the IP address present in SDP to RTP proxy IP address, and sent to main proxy. Main proxy did respond back with 200 OK to INVITE. In 200 OK SDP, Main proxy put the IP address of Media server (media server ip address is different than main proxy server ip address). After the call establishment, it is expected that RTP proxy should send media packets to media server, instead it is sending to proxy server. Pleae do let know, how to put a work around to avoid this problem. Thanks Kamal NECS, Bangalore

2 2

TCP replace alias can cause core
by Jijo 10 Oct '12

10 Oct '12

We found a problem regarding TCP connection alias in the following code at tcp_main.c: int tcpconn_finish_connect( struct tcp_connection* c, union sockaddr_union* from) { : : /* remove all the aliases except the first one and re-add them * (there shouldn't be more then the 3 default aliases at this * stage) */ for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); } c->aliases=1; As TCP_ALIAS_REPLACE flag is set for the default TCP options value, in the function _tcpconn_add_alias_unsafe() a TCP connection alias can be moved from connection A to connection B based on the TCP alias hash. In this case, the number of aliases is incremented in the connection A, and decremented from connection B. However, in the connection B the number of aliases can reach zero (no alias). And the code above can be executed for connection B setting the number of aliases to 1 unconditionally. When this case happens, the connection B keeps an invalid alias (already excluded from connection B by tcpconn_add_alias_unsafe() function called from connection A). When the connection A is released, the aliases are also released, and this memory area can be filled with different data. As connection B has references to an invalid alias it can try to access invalid areas, and can crash Kamailio. This access happens, for example, when another alias is added to connection B. To fix it we include a check before the code: if (c->aliases>0) { for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); memset(a,0xbb,sizeof(struct tcp_conn_alias)); } c->aliases=1; } Please let us know if any comments. Thanks Jijo

2 1

[snmpstats] openserSIPRegRejectedRegistrations not being updated
by Anatoly Ivanov 10 Oct '12

10 Oct '12

Hi, I am observing weird behavior of snmpstats.so - I see openserSIPRegAcceptedRegistrations being updated, but openserSIPRegRejectedRegistrations stays at zero. Could somebody please explain under what conditions this counter is being incremented? I am using auth_radius, so I suspect I simply do not reject failed registartions the 'proper' way, thus causing the counter to stay at 0. Thank you, Anatoly.

2 3

how to use fix_nated_sdp("2") with multiple media types
by MARINA SERRANO MONTES 10 Oct '12

10 Oct '12

Hi, I have a doubt about how to use "fix_nated_sdp("2")" function when we have an scenario with multiple media types (audio and video) and the param "c" in sdp for each media type has a different IP. We are using a SCOPIA 400 MCU to multi-party in a call, and it will set up different IP to send RTP depend on media type. I have tested this function, but it rewrites always all the "c" parameter with the same value. Is there any option to use the function or to rewrite "c" value for each media type independently? Many thanks. Br, Marina ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx

3 11

Nginx dispatching of websocket requests?
by Juha Heinanen 09 Oct '12

09 Oct '12

just to complete the thread, nginx does not currently support proxying of requests that use websocket transport. it is in the roadmap, but no date is set. when nginx receives request over websocket it converts 'Upgrade: Connection: Upgrade' header to 'Connection: close' header and removes Upgrade header. -- juha

1 0

moving all to kamailio namespace
by Daniel-Constantin Mierla 09 Oct '12

09 Oct '12

Hello, some bits were left with the old project name OpenSER, not to disturb too much at that time and see where everything goes. Other voices expressed same idea in the past, now everything is stable and development goes smooth, so it is good to make a decision because openser is too way back and new comers could be confused. In my list is renaming from openser to kamailio next attributes: - default database name (exiting configs should not be impacted that much as most of them have database provided by module parameter) - default database access usernames and passwords, respectively openser, openserro and openserrw should become kamailio, kamailioro and kamailiorw - perl module API is packed as OpenSER - this could affect some people using it, but a search and replace should do it - snmpstats mibs - probably it is going to have some impact as well Eventually we can start doing it gradually, one or two per release, but is probably better to do all at once, so people will do one upgrade with next major release. Another thing is sercmd, it is in conflict for packaging as both ser and kamailio install it. It means that one cannot install both kamailio and ser at same time from packages. One option is to install it as kamcmd. Because these changes affect users and devels, I sent the message to both mailing lists. Opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

6 7

storing and using route-set in dialog module
by Klaus Darilion 09 Oct '12

09 Oct '12

Hi! Are there any means to store the route set of a dialog (on dialog creation) and use the stored route set when forwarding in-dialog requests instead of using the requests' route set? thanks Klaus

1 0

Force to use mediaproxy/rtpproxy
by Mino Haluz 09 Oct '12

09 Oct '12

Hi, I am doing some performance tests and I cannot force SIP INVITEs to use mediaproxy within LAN, without any NAT. I'm using engage_media_proxy or rtpproxy_manage but phones are still exchanging voice data between them, not through the proxy. Is there some trick how to forcibly tell kamailio to use media/rtp proxy ? Maybe kamailio finds out that there is no nat and therefore will not use any nat traversal mechanism. Mino

1 1

get $sht array of values
by Uri Shacked 09 Oct '12

09 Oct '12

Hi, If i have two keys on a HTABLE that are the same. i undestand i need to set the key type to 1 (array). i use the $shtcn and get 3 (size is an entry as well). so far, so good. But, now i am trying to get all the values from the table to an AVP oto say it in another way - i need all values because i parse it afterwards. So, what is the syntax? using $sht() didn't work..... Uri

2 1

kamailio cores on corrupted route header
by Jijo 09 Oct '12

09 Oct '12

Hello, kamailio cores when receives a corrupted route header. For example, this was causing the core. Route: sip:10.236.236.100;transport=tcp;r2=on;lr;ftag=1348218287134-Test-553188;osb-tag=NM;nat=yes;twan=yes?[=& [=<sip:10.236.236.100;transport=tcp;r2=on;lr;ftag=1348218287134-Test-553188;osb-tag=NM;nat=yes;twan=yes?[=&%20[=> I found the problem, the pointer was not initializing to null after freeing it. Please apply this fix in the next version. Here is the diff with the original(3.2.2) and changed version. PGA:/mnt/o/kamailio-3.2.2/parser # diff -u parse_param.c.orig parse_param.c --- parse_param.c.orig 2012-10-09 09:42:58.372003500 -0300 +++ parse_param.c 2012-10-09 21:34:14.556367900 -0300 @@ -545,6 +545,7 @@ error: if (t) pkg_free(t); free_params(*_p); + *_p = 0; return -2; ok: Thanks Jijo

1 0

Kamailio XMPP support
by Vijay Thakur 09 Oct '12

09 Oct '12

Dear All, I have set up my kamailio server successfully and it is working very fine. i have followed the below link for setup: http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to enable xmpp module. At Kamailio wiki page, i have found that there can be two way to implement xmpp module in kamilio. And they are (1) Inside the server (2) Remoter jabber server. But there is no specific "how to" to enable this module. How can i compile and enable this module in kamailio. And after this what will be the setting of my kamailio.cfg. Currently jabberd2 is not available as directed in wiki. Can i use openfire with kamailio for xmpp? How ? Kindly point me any direction. With Warm Wishes, Vijay Thakur information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

1 0

Kamailio XMPP Module Setup
by Vijay Thakur 09 Oct '12

09 Oct '12

Dear All, I have set up my kamailio server successfully and it is working very fine. i have followed the below link for setup: http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to enable xmpp module. At Kamailio wiki page, i have found that there can be two way to implement xmpp module in kamilio. And they are (1) Inside the server (2) Remoter jabber server. But there is no specific "how to" to enable this module. How can i compile and enable this module in kamailio. And after this what will be the setting of my kamailio.cfg. Kindly point me any direction. With Warm Wishes, Vijay Thakur

1 0

Kamailio FAQ Week
by Daniel-Constantin Mierla 09 Oct '12

09 Oct '12

Hello, I'm looking at getting richer content for the FAQ wiki page: * http://www.kamailio.org/wiki/tutorials/faq/main Probably we can get it good enough with a bit of mobilization for one week. If anyone is having some questions fitting the purpose of the FAQ, just add them to the page or send via email. If you don't have the right answer, just leave it as TBA in the wiki, via email is fine to send just the question. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Nginx dispatching of websocket requests?
by Juha Heinanen 09 Oct '12

09 Oct '12

peter suggested using nginx for dispatching xcap requests and that works fine. has someone tried using nginx to dispatch also sip over websocket requests? is there some drawbacks? -- juha

2 1

Kamailio issues errors
by Sangeeta Shah 08 Oct '12

08 Oct '12

I have Kamailio 3.3 with Presence, RLS and integrated XCAP. Now I am noticing the errors in the attached syslog capture. The only thing I have changes is the DB mode to DB Only. Thanks, Sangeeta -- Sangeeta Shah

1 0

implementing call-quota with dialog module
by Aft nix 08 Oct '12

08 Oct '12

Hi, I'm trying to implement a "callquota" based on destination domains. The idea is following: every destation domain has a limit of, say 50 calls. As the sip clients are registered with the "destination domains", i've extracted the domain using "$fd" because it results as same value as destination domain. Problem is, i wanted to have separate quota for each domains. But it seems, that the quota is reached "as total" not as per domain basis. Another thing is, the number of calls reported by it is also unusually large than actual calls. This is implemenated as bellow: #!ifdef WITH_CALL_LIMIT modparam("dialog","enable_stats",1) modparam("dialog","dlg_flag", DLG_FLAG) modparam("dialog","hash_size", 4096) modparam("dialog", "profiles_with_value","callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) #!endif #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$fd", "$var(100)"); if ($var(100) >= 50 ) { xdbg("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$fd"); if (get_profile_size("callquota","$fd", "$var(100)")) { xdbg("DEBUG: there are $var(100) total calls for $fd"); } } #!endif Thanks in advance. -- -aft

2 2

Newbie question about RLS/XCAP/batch'd NOTIFYs
by Dmytro Bogovych 06 Oct '12

06 Oct '12

Greetings. I try to make iOS voip application with presence information. iOS does not let too frequent network updates from server side when runs in background mode. So I cannot rely on plain PUBLISH / SUBSCRIBE scheme - it leads to restart of application by iOS in many cases. Please see this post for details: https://devforums.apple.com/message/565667#565667 The question. Can Kamailio provide way to reduce the number of NOTIFYs? I see it as 1) iOS application creates & uploads RLS via XCAP 2) iOS application asks server about changes in presence information once per 10 minutes or when application goes to foreground state. The best way is to receive single batch'd NOTIFY answer for all watched peers. Is it possible to implement with kamailio? Thank you :)

2 2

compilation warnings
by Vieri 05 Oct '12

05 Oct '12

Hi, On a Gentoo Linux system I'm getting the following compile-time warnings (kamailio-3.3.1 ebuild): * python_msgobj.c:529:5: warning: dereferencing type-punned pointer will break strict-aliasing rules * ld_con.c:241:4: warning: implicit declaration of function 'ldap_simple_bind_s' * perl.c:340:2: warning: null argument where non-null required (argument 2) * perl.c:340:2: warning: null argument where non-null required (argument 1) Hopefully I can just ignore these messages. Thanks, Vieri

3 3

kamailio 3.3 sipcapture and Homer Capture Agent 4
by Juha Heinanen 05 Oct '12

05 Oct '12

is Homer Capture Agent 4 hep protocol incompatible with Kamalio 3.3 sipcapture module? in captagent.xml i set core_hep.conf version to 2, but looks like kamailio is not able to parse the messages: Oct 5 12:40:18 siika sip-proxy[6779]: INFO: <core> [parser/parse_fline.c:145]: ERROR:parse_first_line: method not followed by SP Oct 5 12:40:18 siika sip-proxy[6779]: INFO: <core> [parser/parse_fline.c:242]: ERROR:parse_first_line: bad message Oct 5 12:40:18 siika sip-proxy[6779]: ERROR: <core> [parser/msg_parser.c:705]: ERROR: parse_msg: message=<#002#020#002#006#023ĳ��bg#012�bg#012#002�nP#037#035#010> Oct 5 12:40:18 siika sip-proxy[6779]: ERROR: <core> [receive.c:146]: ERROR: receive_msg: parse_msg failed in kamailio config i have modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) -- juha

2 2

Does Kamailio support ECDHE ciphers for TLS?
by AJ 05 Oct '12

05 Oct '12

Hi, I have Kamailio 3.3.1 working fine with TLS 1.0, running on CentOS 6.3 However, now I want to use some ECDHE ciphersuites, for instance: ECDHE-ECDSA-AES128-SHA I have built an appropriate OpenSSL (1.0.0j) in CentOS to support this, and have generated the appropriate certs: openssl ecparam -out ca-privkey.pem -name secp256r1 -genkey -outform pem openssl req -x509 -new -key ca-privkey.pem -out ca_cert.pem -outform PEM -days 3650 openssl ecparam -out privkey.pem -name secp256r1 -genkey -outform pem openssl req -new -nodes -key privkey.pem -outform pem -out cert_req.pem openssl ca -keyfile ca-privkey.pem -cert ca_cert.pem -in cert_req.pem -out cert.pem However, when I try to connect from my phone I am always getting: Oct 3 20:27:54 /usr/sbin/kamailio[2905]: ERROR: tls [tls_server.c:1190]: TLS accept:error:1408A0C1:SSL routines:SSL3_GET_CLIENT_HELLO:no shared cipher Oct 3 20:27:54 /usr/sbin/kamailio[2905]: ERROR: <core> [tcp_read.c:1127]: ERROR: tcp_read_req: error reading If I use the OpenSSL s_server test-tool, I can connect from my phone just fine with these certs: openssl s_server -accept 8888 -cert cert.pem -key key.pem -pass stdin -CAfile calist.pem -cipher ECDHE-ECDSA-AES128-SHA Does Kamailio support ECDHE ciphersuites? If so, what am I doing wrong? Thanks! AJ

3 2

Kamailio question
by Jostein Aardal 05 Oct '12

05 Oct '12

Hi! I am a student at Oslo and Akershus University college of applied sciences. I'm working on a project where I am supposed to setup a kamailio server. Up until now I have been able to install kamailio and add a user. When I use a sip-client I am able to register and authenticate this user. And I am able to make calls between different users, as long as they both are registered on the same server. But when I try to make a call between users that are registered on different servers/proxies I fail to establish the call. When I look at the sip-messages in wireshark, I see that there is a message "407 proxy authentication required". I thought that enabling "multidomain" in kamailio.cfg would fix this problem, but it doesn't. What I have written in the kamailio.cfg-file is: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_MULTIDOMAIN #!define WITH_ALIASDB #!define WITH_DEBUG yours Jostein Aardal Oslo and Akershus University College of Applied Sciences

3 2

hello i want to active international call
by mark shaw 05 Oct '12

05 Oct '12

hello i want to active international call in my account for important . radioyork2 thedept

2 1

Client info
by Sangeeta Shah 04 Oct '12

04 Oct '12

Hello All, I am trying to use Kamailio 3.3 configured as a presence server and rls server with integrated XCAP. I am trying to test the presence and rls functionality with clients such as Jitsi (doesn't seem to support rls and doesn't let me specify presence rules per contact) and Boghe (doesn't seem to be behaving as expected!). Anyone recommend other clients that work for the functionality I described above. Thanks, Sangeeta -- Sangeeta Shah

1 0

Kamailio LCR - From Uri
by Fatima Chahrour~Vanrise Support 04 Oct '12

04 Oct '12

Dears, Am able to route call using LCR module from anywhere. I need to use certain option in the lcr tables in order to specify the ip of which calls is generated. I suppose that the From_uri column in LCR module meets my requirement, yet am not able to make it work and confused after reading several docs. My question is in what form should the from_uri column be? Is the ip enough? Are the Load params used below missing? modparam("lcr", "db_url","mysql://openser:openserrw@localhost/openser") modparam("lcr","lcr_gw_table","lcr_gw") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","gw_uri_avp", "$avp(i:709)") modparam("lcr","ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr","lcr_id_avp","$avp(s:lcr_id_avp)") What should I change in the routing script to consider the from_uri? if (!load_gws("1")) { sl_send_reply("503", "Unable To lOad GatEwAyS"); exit;} if(!next_gw()){ sl_send_reply("503", "Unable To fInD a gateWaY"); exit;} Thanks in advance, F Chahrour

3 3

sipcapture unsupported family
by Juha Heinanen 04 Oct '12

04 Oct '12

after adding a few module params modparam("sipcapture", "db_url", "MYSQL_SIP_PROXY_USAGE_URL") modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "capture_node", "EXTERNAL_HOSTNAME") i restarted my sip proxy and started to get these kind of errors to syslog: Oct 4 18:09:02 siika /usr/sbin/sip-proxy[22914]: ERROR: sipcapture [sipcapture.c:733]: ERROR: sipcapture:hep_msg_received: unsupported family [71] why they appear and how to turn them off? -- juha

3 4

Presence with RLS and integrated XCAP on Kamailio 3.3
by Sangeeta Shah 04 Oct '12

04 Oct '12

Hello All, Does anyone have presence working on Kamailio 3.3 with integrated XCAP. In theory my setup should work. I have two clients: Client A: Registers with the presence server Subscribe 1 - event = reg Subscribe 2 - event = presence.winfo Subscribe 3 - event = presence Publishes its status I see entries in the foll databases for A: Active Watchers DB: presence, presence.winfo, and reg event entries Watchers DB: No entries Presentity: presence, reg, and presence???? entries PUA list: 1 entry RLS watchers: No entries RLS presentity list: No entries XCAP List: 3 entries - rls services, resource list, and pres rules Client B: Registers with the presence server Subscribe 1 - event = presence.winfo Subscribe 2 - event = presence Subscribe 3 - event = message-summary Publishes its status I see entries in the foll databases for B: Active Watchers DB: message-summary, presence.winfo Watchers DB: 1 entry - with B as the presentity and A as the watcher Presentity: reg, presence PUA list: 1 entry RLS watchers: No entries RLS presentity list: No entries XCAP List: 2 entries resource-lists, pres rules My first question is - Is this all correct or is something going missing? Now the real problem: When Client A adds Client B to it's contact list, on Client A I see the following message: Client B Package (presence), Pending (Subscribe). This information from what I gather is coming back from the periodic notifies for the server because of the presence.winfo event. However, the notify going out to Client B does not have any indication that Client A is a watcher and needs to be authorized. This should be the case per RFC 3857. Excerpts from notifies to client A and client B: Client A NOTIFY sip:8475551001@10.51.2.181:58150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.251.12;branch=z9hG4bKdcef.e5df11e.0 To: sip:8475551001@10.50.251.12;tag=1988526 From: sip:8475551001@10.50.251.12;tag=a6a1c5f60faecf035a1ae5b6e96e979a-c52c CSeq: 3 NOTIFY Call-ID: 19607474-addd-0ffb-a9da-031235052e9d Content-Length: 347 User-Agent: kamailio (3.3.1 (x86_64/linux)) Max-Forwards: 70 Event: presence.winfo Contact: <sip:10.50.251.12:5060;transport=udp> Subscription-State: active;expires=3180 Content-Type: application/watcherinfo+xml <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2" state="partial"> <watcher-list resource="sip:8475551001@10.50.251.12" package="presence"> <watcher id="9e65d804dbc4fc269c5f1d2f6eae0650(a)0.0.0.0" event="subscribe" status="pending">sip:8475551002@10.50.251.12</watcher> </watcher-list> </watcherinfo> Client B: NOTIFY sip:8475551002@10.50.0.113:5060;transport=udp;registering_acc=10_50_251_12 SIP/2.0 Via: SIP/2.0/UDP 10.50.251.12;branch=z9hG4bK7f71.6fca9e81.0 To: sip:8475551002@10.50.251.12;tag=a30db42f From: sip:8475551002@10.50.251.12;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6692 CSeq: 3 NOTIFY Call-ID: f71666432f236f38c7c4dc467b4483e0(a)0.0.0.0 Content-Length: 195 User-Agent: kamailio (3.3.1 (x86_64/linux)) Max-Forwards: 70 Event: presence.winfo Contact: <sip:10.50.251.12:5060;transport=udp> Subscription-State: active;expires=3600 Content-Type: application/watcherinfo+xml <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2" state="full"> <watcher-list resource="sip:8475551002@10.50.251.12" package="presence"/> </watcherinfo> Also the frequency of notifies to client B is a lot less for event: presence.winfo. Any help is greatly appreciated since I've been trying to get presence to work for a few weeks now and keep running into new problems. Thanks, Sangeeta -- Sangeeta Shah

1 0

sip_capture schema?
by Juha Heinanen 04 Oct '12

04 Oct '12

i couldn't find in the source directory database schema for sip_capture table. why it is not included? also, looks like sipcapture module README is missing many of module's params. is this module still supported or has it become obsolete? -- juha

3 9

Kamailio - can't write any logs to file
by Nord7 04 Oct '12

04 Oct '12

Hello everyone. I have installed: CentOS6, Kamailio 3.3, rsyslogd Part of my Kamailio config: debug=9 log_stderror=yes memdbg=9 memlog=9 log_facility=LOG_LOCAL0 loaded modules: debugger.so modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_facility", "LOG_LOCAL0") modparam("debugger", "log_prefix", "from-debugger-with-love:") xlog.so modparam("xlog", "buf_size", 8192) modparam("xlog", "log_facility", "LOG_LOCAL0") rsyslog.conf default config with two added rows: # Save log messages with LOG_LOCAL0 in /var/log/kamailio local0.* /var/log/kamailio rsyslogd and kamailio restarted. When i run command "kamailio -ddddddd" i see debug all my xlog("L_INFO","blabla\n") or xlog("LOG_LOCAL0","L_INFO","blabla\n" ); messages. When i run "kamctl start" i don't have any logs from kamailio in my system. I'm stuck, any suggestions, will be helpful -- Best regards, Kirill Ushakov

2 2

Re: [SR-Users] Error when client tries to put XCAP resource list
by Daniel-Constantin Mierla 03 Oct '12

03 Oct '12

Hello, do not reply privately to discussions on the mailing list. It is important that the thread has a conclusion for the people searching in archive for the same topic. Cheers, Daniel On 10/3/12 10:13 PM, Sangeeta Shah wrote: > Thanks for the response. I was able to figure it out and now it works. > > Appreciate your response. > > On Wed, Oct 3, 2012 at 3:02 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >> Hello, >> >> the http uri is available in available $hu. You can print it with xlog and >> be sure you give it as parameter to xcap server function. >> >> A basic example in the readme shows it: >> * >> http://kamailio.org/docs/modules/stable/modules_k/xcap_server.html#id2532050 >> >> Cheers, >> Daniel >> >> >> >> On 10/3/12 8:34 PM, Sangeeta Shah wrote: >>> Hello All, >>> I am using Kamailio 3.3 with Presence/RLS and integrated XCAP. When >>> my client tries to "PUT" the resource list for the user, the server is >>> returning a 500 Server Error response. Attached is the TCP trace for >>> the put and the corresponding response from the server. I also see the >>> following error in the server logs: >>> Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: >>> <script>: ===== xhttp: put received sip:8475551001@10.50.251.12 >>> Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: >>> xcap_server [xcap_misc.c:109]: invalid http uri - shorter than >>> xcap-root >>> Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: >>> xcap_server [xcap_server.c:560]: cannot parse xcap uri [0] >>> >>> Maybe it's a client/server configuration issue. What is the expected >>> http uri. How do I print out the received HTTP URI And the XCAP root. >>> >>> Any debugging help is appreciated. >>> >>> TCP trace: >>> >>> PUT /xcap-root/resource-lists/users/8475551001/index HTTP/1.1 >>> >>> Host: 10.50.251.12:5060 >>> >>> Content-Length: 2037 >>> >>> Connection: Keep-Alive >>> >>> User-Agent: XDM-client/OMA1.1 >>> >>> X-3GPP-Intended-Identity: 8475551001 >>> >>> Content-Type: application/resource-lists+xml >>> >>> >>> >>> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> >>> >>> <list name="rcs"> >>> >>> <display-name>All Contacts</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <entry uri="8475551001" /> >>> >>> </list> >>> >>> <list name="rcs_blockedcontacts"> >>> >>> <display-name>Blocked Contacts</display-name> >>> >>> </list> >>> >>> <list name="rcs_revokedcontacts"> >>> >>> <display-name>Revoked Contacts</display-name> >>> >>> </list> >>> >>> <list name="oma_allcontacts"> >>> >>> <display-name>OMA All Contacts</display-name> >>> >>> </list> >>> >>> <list name="oma_blockedcontacts"> >>> >>> <display-name>OMA Blocked Contacts</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> </list> >>> >>> <list name="oma_buddylist"> >>> >>> <display-name>OMA BuddyList</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> </list> >>> >>> <list name="oma_grantedcontacts"> >>> >>> <display-name>OMA Granted Contacts</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> </list> >>> >>> <list name="oma_pocbuddylist"> >>> >>> <display-name>OMA POC BuddyList</display-name> >>> >>> </list> >>> >>> <list name="doubango"> >>> >>> <display-name>My Default Contacts</display-name> >>> >>> </list> >>> >>> </resource-lists> >>> >>> HTTP/1.1 500 Server error >>> >>> Via: SIP/2.0/TCP 10.51.2.181:49889 >>> >>> Server: kamailio (3.3.1 (x86_64/linux)) >>> >>> Content-Length: 0 >>> >>> Thanks, >>> Sangeeta >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.com >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - >> http://asipto.com/u/katu >> > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Error when client tries to put XCAP resource list
by Sangeeta Shah 03 Oct '12

03 Oct '12

Hello All, I am using Kamailio 3.3 with Presence/RLS and integrated XCAP. When my client tries to "PUT" the resource list for the user, the server is returning a 500 Server Error response. Attached is the TCP trace for the put and the corresponding response from the server. I also see the following error in the server logs: Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: <script>: ===== xhttp: put received sip:8475551001@10.50.251.12 Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: xcap_server [xcap_misc.c:109]: invalid http uri - shorter than xcap-root Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: xcap_server [xcap_server.c:560]: cannot parse xcap uri [0] Maybe it's a client/server configuration issue. What is the expected http uri. How do I print out the received HTTP URI And the XCAP root. Any debugging help is appreciated. TCP trace: PUT /xcap-root/resource-lists/users/8475551001/index HTTP/1.1 Host: 10.50.251.12:5060 Content-Length: 2037 Connection: Keep-Alive User-Agent: XDM-client/OMA1.1 X-3GPP-Intended-Identity: 8475551001 Content-Type: application/resource-lists+xml <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="rcs"> <display-name>All Contacts</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <entry uri="8475551001" /> </list> <list name="rcs_blockedcontacts"> <display-name>Blocked Contacts</display-name> </list> <list name="rcs_revokedcontacts"> <display-name>Revoked Contacts</display-name> </list> <list name="oma_allcontacts"> <display-name>OMA All Contacts</display-name> </list> <list name="oma_blockedcontacts"> <display-name>OMA Blocked Contacts</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> </list> <list name="oma_buddylist"> <display-name>OMA BuddyList</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> </list> <list name="oma_grantedcontacts"> <display-name>OMA Granted Contacts</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> </list> <list name="oma_pocbuddylist"> <display-name>OMA POC BuddyList</display-name> </list> <list name="doubango"> <display-name>My Default Contacts</display-name> </list> </resource-lists> HTTP/1.1 500 Server error Via: SIP/2.0/TCP 10.51.2.181:49889 Server: kamailio (3.3.1 (x86_64/linux)) Content-Length: 0 Thanks, Sangeeta -- Sangeeta Shah

2 1

Re: [SR-Users] Handling Tel URIs
by Kamal Palei 03 Oct '12

03 Oct '12

Hi Daniel It worked fine. I added the below code. if(is_method("PUBLISH")) { loose_route(); t_relay(); exit; } Thanks Kamal On Wed, Oct 3, 2012 at 5:58 PM, Kamal Palei <palei.kamal(a)gmail.com> wrote: > Thanks Daniel , let me try this out. > > Regards > Kamal > > On Wed, Oct 3, 2012 at 5:33 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> do you have a Route header also for kamailio proxy? If yes, try to do >> loose_route() and then t_relay(). >> >> If not, get the address from Route in $du and relay: >> $du = $(hdr(Route){nameaddr.uri}); >> remove_hf("Route"); >> t_relay(); >> exit; >> >> Cheers, >> Daniel >> >> >> On 10/3/12 1:46 PM, Kamal Palei wrote: >> >> Hi Daniel >> Thanks for your response. >> >> I have a message as below. >> >> PUBLISH tel:+915346586495469 <+915346586495469> SIP/2.0 >> Via: >> To: >> From: >> Route: <sip:main_proxy_ip> >> .. >> .. >> This message comes from phone to Kamailio proxy. >> I just need Kamailio proxy just forward this message to main proxy >> without modifying tel uri present in request uri. >> >> Can this be acheived by tel2sip() api. >> >> Thanks >> Kamal >> >> >> On Wed, Oct 3, 2012 at 4:37 PM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> >>> On 10/3/12 12:36 PM, Kamal Palei wrote: >>> >>> Changing the subject line >>> >>> On Wed, Oct 3, 2012 at 3:54 PM, Kamal Palei <palei.kamal(a)gmail.com>wrote: >>> >>>> Hi All >>>> I am using Kamailio - 3.1.5 version. >>>> >>>> When a message of tel URI comes to Kamailio, Kamailio is not forwarding >>>> the message to next proxy. >>>> If is returning "478 Unresolved response" . >>>> >>>> Can somebody give me some pointer how do I handle this situation? >>>> >>> kamailio cannot forward when there is tel uri -- it requires a domain >>> part (hostname/ip) to know where to send the requests. >>> >>> See this function that should help you convert the tel uri to sip uri: >>> http://kamailio.org/docs/modules/stable/modules_k/siputils.html#id2494464 >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >>> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu >> >> >

1 0

Re: [SR-Users] Handling Tel URIs
by Kamal Palei 03 Oct '12

03 Oct '12

Changing the subject line On Wed, Oct 3, 2012 at 3:54 PM, Kamal Palei <palei.kamal(a)gmail.com> wrote: > Hi All > I am using Kamailio - 3.1.5 version. > > When a message of tel URI comes to Kamailio, Kamailio is not forwarding > the message to next proxy. > If is returning "478 Unresolved response" . > > Can somebody give me some pointer how do I handle this situation? > > Thanks > Kamal > > _______________________________________________ > 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 CDR logging to DB
by Tayeb Meftah 03 Oct '12

03 Oct '12

hi everyone, can someone show me how to save cdr's into db ? i enabled acc in the default config but still no cdr in the acc table thank you -- Tayeb Meftah General director Voice of the blind ™ broadcast freedom http://www.vobradio.org/ Phones: Algeria: +213-660-347746 UK, London: +442032983627 USA, NewYork: +1(347)759-5883 iNum: +883510001288000 we're blind, but not deaf!

2 1

Locate UA in usrloc table
by Vitaliy Aleksandrov 03 Oct '12

03 Oct '12

Hi all. Is there any easy method to find if somebody with $si:$sp is registered (usrloc->received.ip == $si, usrloc->received.port == $sp) ? reg_fetch_contacts uses uri as search key, but in my case I don't have it. The only way I know is to write a module that will use "get_all_ucontacts" function exported by usrloc module in its API.

2 1

FreeSWITCH | Kamailio MWI forwarding
by Tayeb Meftah 03 Oct '12

03 Oct '12

hello we're using Kamailio / freeswitch freeswitch is the Voicemail server now. please how to forward MWI bethwan kamailio and Freeswitch? if user 1000 have a vm i want kamailio to fetch it from FS and advertise it to the subscriber thank you -- Tayeb Meftah General director Voice of the blind ™ broadcast freedom http://www.vobradio.org/ Phones: Algeria: +213-660-347746 UK, London: +442032983627 USA, NewYork: +1(347)759-5883 iNum: +883510001288000 we're blind, but not deaf!

2 1

WARNING:fm_free at stop
by Juha Heinanen 03 Oct '12

03 Oct '12

at some point i have started to get these kind of warning when i stop my sip proxy: --text follows this line-- Oct 1 10:23:01 siika /usr/sbin/sip-proxy[16970]: WARNING: <core> [mem/f_malloc.c:512]: WARNING:fm_free: free(0) called any idea where they come from or how i could found out where they come from? -- juha

2 20

(no subject)
by Kamal Palei 03 Oct '12

03 Oct '12

Hi All I am using Kamailio - 3.1.5 version. When a message of tel URI comes to Kamailio, Kamailio is not forwarding the message to next proxy. If is returning "478 Unresolved response" . Can somebody give me some pointer how do I handle this situation? Thanks Kamal

1 0

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

I will check. The problem is that this installation is exotic and deviant from my usual order of operations, which creates further confusion. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: On 10/2/12 9:45 PM, Alex Balashov wrote: Perhaps the source of my misunderstanding is something to do with branching. The actual case I am troubleshooting is a 503 generated via t_reply out of a failure route. The complaint is that this 503 contains a modified From header, exposed to the originator of the call. If you created the transaction after calling msg_apply_changes(), then I am quite sure it is how it worked all the time. Try with an older version and see. Cheers, Daniel -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Those originated locally were never using the original value if you called msg apply changes -- you can try with an older version, if it does that, then it is a bug somewhere :-). Btw, if you called msg apply changes after creating the transaction, then SL is out of scope and tm local reply is using the initial values, but that means the changes are only for that branch, a failure route will get the initial request content. Cheers, Daniel On 10/2/12 9:30 PM, Alex Balashov wrote: The expected behaviour is that all replies, both relayed and endogenously originated (via SL or TM), as well as sequential requests directed back to the caller, will preserve the original headers. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Not sure what is 'work as you expect', provided the details I gave in the previous mail related to local replies. Otherwise, if you don't track all dialogs, could be an issue if some of the replies are ok and some are not. I will check the uac module and add a parameter to not use dialog module even if loaded, if such option is not already there. Cheers, Daniel On 10/2/12 9:11 PM, Alex Balashov wrote: So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.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 _______________________________________________ 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 Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

Perhaps the source of my misunderstanding is something to do with branching. The actual case I am troubleshooting is a 503 generated via t_reply out of a failure route. The complaint is that this 503 contains a modified From header, exposed to the originator of the call. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:Those originated locally were never using the original value if you called msg apply changes -- you can try with an older version, if it does that, then it is a bug somewhere :-). Btw, if you called msg apply changes after creating the transaction, then SL is out of scope and tm local reply is using the initial values, but that means the changes are only for that branch, a failure route will get the initial request content. Cheers, Daniel On 10/2/12 9:30 PM, Alex Balashov wrote: The expected behaviour is that all replies, both relayed and endogenously originated (via SL or TM), as well as sequential requests directed back to the caller, will preserve the original headers. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Not sure what is 'work as you expect', provided the details I gave in the previous mail related to local replies. Otherwise, if you don't track all dialogs, could be an issue if some of the replies are ok and some are not. I will check the uac module and add a parameter to not use dialog module even if loaded, if such option is not already there. Cheers, Daniel On 10/2/12 9:11 PM, Alex Balashov wrote: So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.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 _______________________________________________ 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 Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

The expected behaviour is that all replies, both relayed and endogenously originated (via SL or TM), as well as sequential requests directed back to the caller, will preserve the original headers. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:Not sure what is 'work as you expect', provided the details I gave in the previous mail related to local replies. Otherwise, if you don't track all dialogs, could be an issue if some of the replies are ok and some are not. I will check the uac module and add a parameter to not use dialog module even if loaded, if such option is not already there. Cheers, Daniel On 10/2/12 9:11 PM, Alex Balashov wrote: So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.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 _______________________________________________ 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 Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.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 _______________________________________________ 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 Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Alex Balashov <abalashov(a)evaristesys.com> wrote:Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.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 1

uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 0

Kamailio project booth at Astricon 2012
by Daniel-Constantin Mierla 02 Oct '12

02 Oct '12

Hello, Kamailio project will participate at Astricon 2012 with its own booth in the open source area, courtesy of our Asterisk friends organizing the show. The event takes place in Atlanta, GA, USA, between Oct 23-25, 2012: * http://www.astricon.net/2012/default.aspx Special thanks to community members JR Richardson, Nir Simionovich and Eric Klein, which coordinate the activity at the booth. I made a news about it at: * http://www.kamailio.org/w/2012/10/kamailio-booth-at-astricon12/ If you are interested in joining the team at the booth and present how you use Kamailio, reply to me in order discuss further. If you participate at Astricon, stop by booth #20 to chat about use cases of Kamailio and what is new around the project. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Kamailio - VoIP dinner in Berlin, next Monday, Oct 8, 2012
by Daniel-Constantin Mierla 02 Oct '12

02 Oct '12

Hello, we're planning a dinner in Berlin, next Monday, Oct 8, 2012 - exact time and place to be decided soon. Besides me, among people interested to participate are Alex Balashov, Andrei Pelinescu-Onciul, Jiri Kuthan and Raphael Coeffic. If anyone here wants to participate, reply to count the heads for reservation. It's the usual german rule, everyone pays for him/herself. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

XCAP configuration with Presence Server and Resource lists not working on Kamailio 3.3.1
by Sangeeta Shah 01 Oct '12

01 Oct '12

Hello All, I am trying to get the presence module on Kamailio 3.3.1 working with integrated XCAP and resource lists. The clients I am using to test are: Jitsi and Boghe. I am following the configuration as specified by the respective module documentation: http://kamailio.org/docs/modules/devel/modules_k/xcap_server.html http://kb.asipto.com/kamailio:presence:k31-made-simple http://www.kamailio.org/dokuwiki/doku.php/presence:presence-module and http://kamailio.org/docs/modules/devel/modules_k/rls.html The server is up and running but it doesn't seem like it's working for either one of the clients. I checked the archives and someone had a similar issue in Aug and switched back to releases 3.2/3.1. Has anything been resolved? Are there any changes that need to be made to the documentation for 3.3 to support this functionality. Any help is appreciated. If tcpdumps for the respective clients are desired I can provide them. Thanks, Sangeeta -- Sangeeta Shah

2 2

Error starting kamailio with dialog module usage
by Aft nix 01 Oct '12

01 Oct '12

Hi, After implementing a "Call limit" application with the following config: #!ifdef WITH_CALL_LIMIT modparam("dialog","enable_stats",1) modparam("dialog","dlg_flag", 4) modparam("dialog","hash_size", 4096) modparam("dialog", "profiles_with_value","callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) #!endif request_route { ................ #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$rU", "var(100)"); if ($var(100) >= 5 ) { xlog("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$ru"); if (get_profile_size("callquota", "$var(100)")) { xlog("DEBUG: there are $var(100) total calls for $rU\n"); } } #!endif ............... } Now it passes cleanly when i issue : $kamailio -c but it fails to start with following info: Sep 29 19:37:49 108 /usr/local/sbin/kamailio[1865]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:507 Sep 29 19:37:49 108 /usr/local/sbin/kamailio[1865]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:523 line 507 : get_profile_size("callquota", "$rU", "var(100)"); Thanks in advance -- -aft

2 2

kamailio core at qm_status
by Jijo 01 Oct '12

01 Oct '12

Hi All, I'm observing a core intermittently at "qm_status (qm=0x786cd000) at mem/q_malloc.c:763" for kamailio version 3.1.0 looking at the backtrace this is occurring while doing a tcp buffer overrun. Please have a look and let me know if anyone observed this issue or how can we debug it. Please find the backtrace and kamailio version osbprod-V2R0:~ # /usr/sbin/kamailio -V version: kamailio 3.1.0 (i386/linux) 21a375 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, 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, PKG_SIZE 15MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 21a375 compiled on 12:38:44 Apr 26 2012 with gcc 4.5.0 #0 qm_status (qm=0x786cd000) at mem/q_malloc.c:763 #1 0x0819c0f3 in qm_debug_frag (qm=0x786cd000, f=<value optimized out>) at mem/q_malloc.c:167 #2 0x0819c352 in qm_free (qm=0x786cd000, p=0x7a6cb6ac, file=0x821cce2 "<core>: tcp_main.c", func=0x822285f "wbufq_run", line=880) at mem/q_malloc.c:457 #3 0x0815f686 in wbufq_run (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:880 #4 handle_tcpconn_ev (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:4141 #5 0x08169bf2 in io_wait_loop_epoll () at io_wait.h:1092 #6 tcp_main_loop () at tcp_main.c:4606 #7 0x080b0404 in main_loop () at main.c:1655 #8 0x080b1f84 in main (argc=9, argv=0xbfb4c9f4) at main.c:2446 (gdb) l 758 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ", 759 "dumping free list stats :\n"); 760 for(h=0,i=0;h<QM_HASH_SIZE;h++){ 761 unused=0; 762 for (f=qm->free_hash[h].head.u.nxt_free,j=0; 763 f!=&(qm->free_hash[h].head); f=f->u.nxt_free, i++, j++){ 764 if (!FRAG_WAS_USED(f)){ 765 unused++; 766 #ifdef DBG_QM_MALLOC 767 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ",

4 7

How to force tcp connection reuse for NOTIFY
by Owen Lynch 01 Oct '12

01 Oct '12

Hi, I want to configure Kamailio so that it sends a NOTIFY using the tcp connection already established by the client SUBSCRIBE request. The aim is for clients to establish secure TLS connections to Kamailio when they first register or subscribe, then for kamailio to reuse these connections for all further NOTIFY/INVITE requests. I thought that adding force_tcp_alias() in the route script would accomplish this, but it doesn't. The NOTIFY always uses the ip:port tuple specified in the SUBSCRIBE Contact header. Can anyone shed any light on this? Thanks, Owen Lynch

2 2

Registration Authentication Error
by Nathaniel L Keeling III 30 Sep '12

30 Sep '12

Hello, we are testing the new version of Kamailio, 3.3 and are getting authentication error when trying to register a sip account. We have a db backend in which we used kamctl to add the user to the db. We are using the kamailio.cfg script created during the install with changes for our environment. Is there some other changes required besides the normal. I can see in the logs where everything looks ok except for when it compares the calculated values. I have doubled check the subscriber table and password but to no avail. 11(28548) DEBUG: db_postgres [km_dbase.c:393]: PQclear(1008b8560) result set 11(28548) DEBUG: auth_db [authorize.c:124]: HA1 string calculated: e13d569284e76a33ca63e4d6203f844a 11(28548) DEBUG: auth [api.c:211]: check_response: Our result = '18210b5da2b3d27e6ba50977072599ea' 11(28548) DEBUG: auth [api.c:221]: check_response: Authorization failed Thanks Nathaniel

5 18

Kamailio 3.3 presence with resource lists and integrated XCAP
by Sangeeta Shah 28 Sep '12

28 Sep '12

Hello All, I am trying to configure Kamailio v 3.3 to serve as a presence server with resource lists and integrated XCAP. I followed the respective module documentation and am trying to test with the Boghe IMS client. I just can't get it to work! I think I still don't have everything I need in the config file to handle the xcap documents. Attached is the debug log from the server. If someone has had luck setting up the presence server with integrated xcap and resource lists please take a few minutes to take a look at the log file and see if you can help me out. I am also posting the xcap portion of the config file: #!ifdef WITH_XCAPSRV ##!define WITH_XHTTPAUTH event_route[xhttp:request] { xdbg("===== xhttp: request [$rv] $rm => $hu\n"); #!ifdef WITH_XHTTPAUTH if (!www_authorize("10.50.251.12", "subscriber")) { www_challenge("10.50.251.12", "0"); exit; } #!endif if($hu=~"^/xcap-root/") { set_reply_close(); set_reply_no_connect(); # xcap ops $xcapuri(u=>data) = $hu; if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:" + $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; xlog("===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if($xcapuri(u=>auid)=="xcap-caps") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>pres-rules</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "ok", "application/xcap-caps+xml", "$var(xbody)"); exit; } #!ifdef WITH_XHTTPAUTH # be sure auth user access only its documents if ($au!=$(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } #!endif switch($rm) { case "PUT": #xcaps_put("$var(uri)", "$hu", "$rb"); #if($xcapuri(u=>auid)=~"pres-rules") #{ # xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); # pres_update_watchers("$var(uri)", "presence"); # pres_refresh_watchers("$var(uri)", "presence", 1); #} xcaps_put("$var(uri)", "$var(doc_uri)", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } else if ($xcapuri(u=>auid)=~"resource-lists" || $xcapuri(u=>auid)=~"rls-services") { rls_update_subs("$var(uri)", "presence"); } exit; break; case "GET": xlog("===== xhttp: get $var(uri) => $hu\n"); xcaps_get("$var(uri)", "$hu"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$hu"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp del: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } exit; break; } } # http ops xhttp_reply("200", "ok", "text/html", "<html><body>OK: $si:$sp</body></html>"); exit; } -- Sangeeta Shah

1 0

DB accounting missing if t_newtran() is called explicitly
by Andrew Pogrebennyk 28 Sep '12

28 Sep '12

Hi, I have found recently that in order to detect retransmits I have to create a transaction explicitly when the request comes in: force_rport(); if(!t_check_trans()) t_newtran(); sl_send_reply("100", "Trying"); xlog("L_INFO", "New request - $ci\n"); it appears like there are carriers or UAs that do not honor the T1 retransmission interval retransmit the INVITE sooner than proxy creates a transaction in t_relay(). And since we are counting concurrent calls, we count the same call multiple times, which is not good. But with this patch we've faced another sporadic problem - if the transaction is created beforehand the accounting record is lost.. we use acc_db mode and set flag to account the transaction. And there are no errors in kamailio log but no insert into acc in mysql binlog either. I wasn't successful reproducing it in the lab systems with identical setup. Is anybody here perhaps aware of some limitation in acc module or callbacks which makes a transaction created beforehand not accountable? On a related note, it could make sense to create a transaction implicitly if dlg_manage() is called to avoid counting same call many times, I just don't know yet how common this issue is in real life.

2 4

SRV record in via header not resolving.
by Jijo 28 Sep '12

28 Sep '12

Hello, I'm observing that Via Header domain name of type SRV is resolving as DNS A. I looked at the code and it does support only A. It is basically setting to default SIP port check_via_address which is causing DNS A query. Is there any specific reason to do so? I changed the code to support SRV as below and working fine. static int check_via_address(struct ip_addr* ip, str *name, unsigned short port, *short proto*, int resolver) { : : //if (port==0) port=SIP_PORT; if (resolver&DO_DNS){ DBG("check_via_address: doing dns lookup\n"); /* try all names ips */ he=sip_resolvehost(name, &port, *(char *) &proto*); /* don't use naptr */ if (he && ip->af==he->h_addrtype){ for(i=0;he && he->h_addr_list[i];i++){ if ( memcmp(&he->h_addr_list[i], ip->u.addr, ip->len)==0) return 0; } } } : : : } /* check if IP address in Via != source IP address of signaling */ int received_test( struct sip_msg *msg ) { int rcvd; rcvd=msg->via1->received || check_via_address(&msg->rcv.src_ip, &msg->via1->host, msg->via1->port,* msg->via1->proto*, received_dns); return rcvd; } Thanks Jijo

2 2

Kamailio sip domain
by Burak ÖZTÜRK 28 Sep '12

28 Sep '12

Hi, What is the sip domain in kamctlrc file? Can I change it whatever I want? Is it must be a valid sip domain? Can I give a name like "mysipdomain.com", but there is no server with this name?

2 1

invalid version 6 for table starting from git rev 78dae896127ce6762e3fa7c2541e1b5f9b8a9023
by siklub 28 Sep '12

28 Sep '12

Hi All, This is what i have done. * get kamailio from git git clone git://git.sip-router.org/sip-router * Installing kamailio from working rev A * destroy the database * update kamailio code from git * installing kamailio from not working rev B * create the database again (kamdbctl create) * starting kamailio This is the error that i got. ep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: <core> [db.c:440]: invalid version 6 for table location found, expected 5 (check table structure and table "version") Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: usrloc [dlist.c:640]: error during table version check. Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: usrloc [dlist.c:640]: error during table version check. Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: registrar [reg_mod.c:484]: failed to register domain I try to find last working rev Last working rev is (rev A) : ca551f7cb3770a08832758e543587415b3c6d80d (2012-09-11) First not working rev is (rev B) : 78dae896127ce6762e3fa7c2541e1b5f9b8a9023 There is no special reason for destroying the database, i'm just playing with kamailio from https://www.kamailio.org/wiki/install/devel/git, it seems that i have done right procedure. Am i missed something? Thanks

2 2

dispatching xcap requests
by Juha Heinanen 28 Sep '12

28 Sep '12

is it possible to use kamailio based dispatcher to dispatch also xcap requests to one or more kamailio instances that serve them, i.e., is it possible to call t_relay in event_route [xhttp:request]? -- juha

3 2

Kamailio sip user authentication
by Burak ÖZTÜRK 28 Sep '12

28 Sep '12

I installed kamailio from this link: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git And I used Linphone and/or CSipSimple android apps as sip clients. And if I give the correct sip server name (usually server ip address), kamailio does not check the sip username and password and sip clients can register. In short, kamailio does not have authentication machanism for sip clients. Is it true or I'm doing something wrong? I searched the web and saw an authentication module called auth_identity. Is it a module to make authentication and if yes, how to activate it? Thanks Regards

2 1

non-local socket info message
by Juha Heinanen 27 Sep '12

27 Sep '12

when i do registrar save in multiple registrar setup, i get this kind of info message to syslog when socket field value in location table does not match local socket of the registrar: INFO: usrloc [udomain.c:321]: non-local socket <tcp:192.98.103.2:5070>...ignoring looks like location table entries are updated no matter of the message. what does "ignoring" mean? in my case i don't use socket field value for anything, because requests to contacts are always routed to dispatcher. would it make sense to have a module param that tells usrloc to skip handling of socket info? -- juha

1 0

help with kamailio crash
by Vicente Hernando 26 Sep '12

26 Sep '12

Hello, we are using a kamailio 3.3.1 version here and suddenly it crashed with a core. Any point about how to debug this will be appreciated! # /usr/local/kamailio/sbin/kamailio -V version: kamailio 3.3.1 (x86_64/linux) 91e8cb flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 91e8cb compiled on 04:59:57 Sep 13 2012 with gcc 4.6.3 (gdb) bt full #0 0x00007f242514e445 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f2425151bab in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x00000000004740b2 in sig_alarm_abort (signo=<optimized out>) at main.c:695 No locals. #3 <signal handler called> No symbol table info available. #4 0x00007f24252066d7 in syscall () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000542c58 in futex_get (lock=0x7f23e1ceb7a8) at mem/../futexlock.h:110 i = <optimized out> #6 timer_del_safe (tl=0x7f23e1d378e0) at timer.c:729 ret = -1 #7 0x0000000000447ab6 in destroy_dns_cache () at dns_cache.c:245 No locals. #8 0x0000000000474f95 in cleanup (show_status=1) at main.c:572 memlog = <optimized out> #9 0x0000000000475969 in shutdown_children (show_status=1, sig=15) at main.c:712 No locals. #10 0x00000000004767ac in handle_sigs () at main.c:803 chld = <optimized out> chld_status = 139 memlog = <optimized out> #11 0x0000000000478ed5 in main_loop () at main.c:1762 i = <optimized out> pid = <optimized out> si = <optimized out> si_desc = "udp receiver child=5 sock=79.170.68.187:5060\000\177", '\000' <repeats 18 times>, "\001\000\000\000\000\000\000\000\000\000\300>\000\000\000\000\001\000\000\000#\177\000\000Q\251U\000\000\000\000\000\020p\321$$\177\000\000\b\000\000\000\000\000\000\000x\264\316\341#\177\000\000\000gE\345\322\024P\027" nrprocs = <optimized out> #12 0x000000000041add2 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2546 cfg_stream = <optimized out> c = <optimized out> r = <optimized out> tmp = 0x7fff551d8f24 "" ---Type <return> to continue, or q <return> to quit--- tmp_len = 1427999204 options = 0x5df520 ":f:cm:M:dVIhEb:l:L:n:vrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 603487729 rfd = 0 debug_save = <optimized out> debug_flag = <optimized out> dont_fork_cnt = <optimized out> n_lst = 0x0 p = <optimized out> Thanks in advance, Vicente.

2 1

Question about CPU
by Ricardo Martinez 26 Sep '12

26 Sep '12

Hello. Is normal this output from the “top” command?! I have a kamailio with 16 children receiving calls at rate of 30cps and using the dialog module to timeout the calls : This is the top command : Tasks: 168 total, 2 running, 166 sleeping, 0 stopped, 0 zombie Cpu0 : 56.0%us, 0.7%sy, 0.0%ni, 43.4%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu1 : 6.3%us, 3.0%sy, 0.0%ni, 87.8%id, 0.0%wa, 0.7%hi, 2.3%si, 0.0%st Cpu2 : 0.3%us, 0.3%sy, 0.0%ni, 99.3%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu3 : 0.7%us, 0.3%sy, 0.0%ni, 99.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 3924912k total, 535700k used, 3389212k free, 20236k buffers Swap: 6291448k total, 0k used, 6291448k free, 226120k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND *2566 root 20 0 1163m 27m 23m R 56.2 0.7 2:08.32 kamailio * 1103 root 20 0 249m 1656 924 S 2.7 0.0 1:32.89 rsyslogd 2525 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.93 kamailio 2527 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.11 kamailio 2528 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.89 kamailio 2529 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.13 kamailio 2535 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.86 kamailio 2537 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.85 kamailio 2538 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.17 kamailio 2541 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.91 kamailio 2542 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.07 kamailio 2543 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.12 kamailio 2544 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.64 kamailio 2549 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.10 kamailio 2551 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.14 kamailio 2552 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.00 kamailio 2495 root 20 0 15088 1240 900 R 0.3 0.0 0:04.10 top 2526 root 20 0 1163m 27m 23m S 0.3 0.7 0:02.83 kamailio 2532 root 20 0 1163m 27m 23m S 0.3 0.7 0:02.87 kamailio 1 root 20 0 19396 1500 1192 S 0.0 0.0 0:02.86 init 2 root 20 0 0 0 0 S 0.0 0.0 0:00.01 kthreadd 3 root RT 0 0 0 0 S 0.0 0.0 0:00.00 migration/0 4 root 20 0 0 0 0 S 0.0 0.0 0:00.00 ksoftirqd/0 5 root RT 0 0 0 0 S 0.0 0.0 0:00.00 migration/0 6 root RT 0 0 0 0 S 0.0 0.0 0:00.00 watchdog/0 The question is : I always see the process PID : 2566 using more CPU than the rest of the childrens. Is this normal? Regards, Ricardo Martinez-.

2 3

LDAP user authentication
by martian＠centrum.sk 26 Sep '12

26 Sep '12

Hi. I have been looking into LDAP way of authenticating users. Reading this guide http://www.kamailio.org/dokuwiki/doku.php/tutorials:kamailio31-auth-ldap and reading the AUTH and LDAP module documentations, it seems to me that currently you can bind to LDAP (using some service account for example) and perform the SEARCH operation for data only. Therefore you need to retrieve user login and password from the LDAP db and than authenticate the user in Kamailio. My question is: What is you can't simply retrieve the password from LDAP db?? Is it possible to do a BIND operation to LDAP, using login name and password provided by user in REGISTER message? (this means not using the ones specified in the external ldap config file). BIND operation kind of authenticates the user. So theoreticaly, if LDAP binding authentication succeeds, the user is trusted and can be replied with 200 OK. This in fact means: using bind operation instead of search operation when a REGISTER message (with Authorization header) arrives. Any opinions on corectness of this approach are welcome, along with clarifying the possibility to do this. Thanks in advance. Martin

1 0

Audiocide
by Cong Le Kha 26 Sep '12

26 Sep '12

Dear Kamailio I have got a problem as you with Audiocode SIP/ SECurity/ MGmt/ DebugRecording/ ControlProtocol/ CONFiguration/ IPNetworking/ TPApp/ BSP/ PING SHow />NOTIC:( sip_stack)(185 ) AcSIPParser: Problem in SIP Message Headers RECOV:( sip_stack)(186 ) !! [ERROR] AcSIPParser: Parse Error. No appropriate delimiter found in multipart body RECOV:( sip_stack)(187 ) !! [ERROR] Message type: INVITE RECOV:( sip_stack)(188 ) !! [ERROR] Source header: RECOV:( sip_stack)(189 ) !! [ERROR] Line: 41. Column: 1 Can you help me?

3 2

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
by Samuel Muller 26 Sep '12

26 Sep '12

Hello all, I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN. I configured it like this : Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062). Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy. I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. But the audiocodes returns some errors about SIP headers sent by Kamailio : ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected Here you have the example of an INVITE from a SIP phone to the PSTN : ** audiocodes debug ** 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from 77.246.81.132:5060 ---- INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0 To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>> Call-ID: 944d8aec-27503ee6(a)192.168.0.113 CSeq: 102 INVITE Max-Forwards: 49 Contact: "Sam" <sip:0123451010@77.246.81.162:15170> Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 281 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, replaces Content-Type: application/sdp P-Asserted-Identity: <0123451010> Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes v=0 o=- 26933860 26933860 IN IP4 192.168.0.113 s=- c=IN IP4 77.246.81.133 t=0 0 m=audio 35038 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes. When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations. I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ... Someone can confirm or already met the problem ? Many thanks all :) .Sam.

5 5

Kamailio and SylkServer
by Gary Shergill 25 Sep '12

25 Sep '12

Hi Kamailio Community, I was wondering how I would integrate Kamailio into SylkServer please? The SIP IM/presence part of Kamailio that is. When I was attempting to get the Presence and IM working I received the following reply; =================== > If you use sylkserver, then you can just forward the >INVITE to sylkserver, which has an MSRP relay integrated, =================== I'm not entirely sure what that means, but I found the following which I believe relates to it; http://sylkserver.ag-projects.com/projects/sylkserver/wiki/Applications I'm not entirely sure if; a) this is the right thing I'm looking at, b) where I would add these lines to Kamailio. Thanks for any help you can provide. Kind Regards, Gary Shergill

1 1

Asterisk and Dispatcher Module
by Ian French 25 Sep '12

25 Sep '12

Hi, I've been working my way throught this tutorial ( http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb) and all it working well thanks. In the other benifits section you mention that this configuration can be used with multiple instances of Asterisk. Could you elaborate a little on this?? Is this achieved using the Dispatcher module? I'm trying to get this working with two instances of Asterisk but I can figure out how to specifc multiple asterisk servers as oppossed to the single. asterisk.bindip = "192.168.178.23" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" Thanks in advance Ian

2 1

Re: [SR-Users] is kamilio support RFC5626 now ?
by Peter Dunkley 25 Sep '12

25 Sep '12

Hi, I have been working on adding full RFC5626 support to Kamailio in my spare time - but it is taking longer than I hoped because I haven't been able to spend as much time as I wanted working on it. There is a branch in git (pd/outbound) where I have put some of the module skeleton code, and I still hope to get this done and tested well before the end of the year (and certainly in time for Kamailo 3.4). Regards, Peter Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >_______________________________________________ >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

Starting kamailio. Error on topoh module
by Miguel Baptista 25 Sep '12

25 Sep '12

Hello, I am testing the "topology hiding" module (topoh) on kamailio-3.2.0 and every-time I start kamailio I get these error messages related with the topoh module: Sep 20 16:51:14 trd-sgw1 /usr/proxy/kamailio/sbin/kamailio[31768]: ERROR: topoh [th_mask.c:79]: +++ ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789.- Sep 20 16:51:14 trd-sgw1 /usr/proxy/kamailio/sbin/kamailio[31768]: ERROR: topoh [th_mask.c:80]: +++ y0GxFtSUVeTOMWzHPam.jKR93gCoNpkicrJBqIEhvYbnd175AfLl4u6wZs-X8Q2D I am only getting those errors when kamailio is starting. Kamailio runs fine and it hides my internal topology, on the SIP Headers, without any error message. I tried to add/modify/remove several topoh parameters but I am always getting those errors starting kamailio. Any idea why I getting it? How can I get rid of it? I don't feel comfortable to apply that configuration on my production servers knowing that kamailio outputs an error when starting. Thanks, - Miguel Baptista

2 3

Dialog Module : "inconsitent dlg timer data"
by Ricardo Martinez 25 Sep '12

25 Sep '12

Hello. Can someone explain to me why i’m getting this error : WARNING: dialog [dlg_req_within.c:173]: inconsitent dlg timer data on dlg 0xb565c500 [2166:2082643875] with clid ' 5ed68b60-fdbe-4aa7-819e-0002a400e27c(a)64.76.155.86' and tags '5e604a9ea4' 'ed505c1da4' This is the output for dlg_list_ctx kamctl fifo dlg_list_ctx dialog:: hash=2995:55475986 state:: 4 ref_count:: 2 timestart:: 1348177242 timeout:: 15084496 callid:: 33d68b60-a09a-b1dc-8198-0002a400e27c(a)64.76.155.86 from_uri:: sip:557100050994@64.76.155.86 from_tag:: 3360b198a4 caller_contact:: sip:557100050994@64.76.155.86 caller_cseq:: 226 caller_route_set:: caller_bind_addr:: udp:64.76.155.82:5060 callee_bind_addr:: udp:64.76.155.82:5060 to_uri:: sip:0267005622408596@64.76.155.82 to_tag:: ba507815a4 callee_contact:: sip:0267005622408596@64.76.155.130 callee_cseq:: 226 callee_route_set:: context:: I can’t find the problem. I’m using this dialog params : # ---- dialog params ------ modparam("dialog", "dlg_flag", 31) modparam("dialog", "profiles_with_value","gwid" ) modparam("dialog", "default_timeout",7200) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) $avp(i:10)=10 $dlg_ctx(timeout_bye) = 4; dlg_manage(); Regards, Ricardo

2 3

kamailio workers and private memory
by Uri Shacked 25 Sep '12

25 Sep '12

Hi, I use "kamctl sercmd pkg.stats" to check the private memory status. I notice that the workers for the proccess "slow timer", "timer", "MI FIFO", use a lot of memory. My system configuration does "mt_reload" for around 5 million lines every hour, it looks like in time, the pkg memory of the workers above is growing...... I set the private memory to 16M instead of 4M. Now the used size (after 3 weeks) is 4M. I will upgrade to 3.3.0 soon and will use mem_join=1, will it help the private memory defrag as well? How can I increase the number of workers for these processes? Will it help increasing it if possible? BR, Uri

2 1

Issue Starting up Kamalio.
by Jay chary 24 Sep '12

24 Sep '12

Hi We are trying to install Kamailio v3.3.1 on CENTOS, I followed the below urls for installation http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Howeever when I try to start Kamailio by issuing kamctl start , I see the below response : INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed and I see the below in the messages log : Sep 20 10:08:49 kamailio kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Sep 20 10:08:49 kamailio /usr/local/sbin/kamailio[21533]: INFO: usrloc [hslot.c:53]: locks array size 512 Sep 20 10:08:49 kamailio /usr/local/sbin/kamailio[21533]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: auth_db [authdb_mod.c:244]: empty parameter 1 not allowed Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:543 Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:552 Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:585 Sep 20 10:09:30 kamailio avahi-daemon[2760]: Invalid query packet. Can u please help us ? Regards Chary

3 2

Kamailio presentations
by Pavel Segeč 24 Sep '12

24 Sep '12

Hello Everybody, I'm an university teacher and founder of the nil.uniza.sk portal with some SIP/Kamailio tutorials. I'd like to use the Kamailio server as the basement for my class lecturers about the IP multimedia communication systems. May anyone help me and provide a link to usable presentation about the kamailio, which should I use and probably extend in my classes? I search the web but nothing appropriate have not found thanks palo73

1 0

Re: [SR-Users] xcap server support for org.openmobilealliance.xcap-directory
by Peter Dunkley 23 Sep '12

23 Sep '12

Hi, org.openmobilealliance.xcap-directory isn't supported by the xcap-server module. Adding support for this shouldn't be difficult though. Regards, Peter Juha Heinanen <jh(a)tutpro.com> wrote: >in xcap module readme there is an example xcap server config file and it >does not include in xcap-caps aiud >org.openmobilealliance.xcap-directory. does it mean that xcap-directory >application is not currently supported or could the aiud just be added >there? > >-- juha > >_______________________________________________ >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

xcap server support for org.openmobilealliance.xcap-directory
by Juha Heinanen 22 Sep '12

22 Sep '12

in xcap module readme there is an example xcap server config file and it does not include in xcap-caps aiud org.openmobilealliance.xcap-directory. does it mean that xcap-directory application is not currently supported or could the aiud just be added there? -- juha

1 0

"Command not found" error during "make install"
by Youngjin Park 22 Sep '12

22 Sep '12

Hi, I faced the following error during installing "sip-router" by command "make install" after successful compilation by command "make all". I will appreciate you so much if you can shed light on. My system is 32 bit Fedora. Thanks in advance. Youngjin install -m 644 README /usr/local/share/doc/ser/ docbook2x-man -s ../../doc/stylesheets/serdoc2man.xsl auth.xml make[1]: docbook2x-man: Command not found make[1]: *** [auth.7] Error 127 make: [modules-man] Error 1 (ignored) docbook2x-man -s ../../doc/stylesheets/serdoc2man.xsl auth_db.xml make[1]: docbook2x-man: Command not found make[1]: *** [auth_db.7] Error 127 make: [modules_s-man] Error 1 (ignored) make[1]: `sercmd' is up to date.

3 6

no socket found for match second RR
by Juha Heinanen 21 Sep '12

21 Sep '12

when proxy gets ack to invite, it spits out this kind of error message: Sep 20 22:03:35 wheezy2 /usr/sbin/sip-proxy[4575]: INFO: processing ACK <sip:jh@192.98.103.10:5074> Sep 20 22:03:35 wheezy2 /usr/sbin/sip-proxy[4575]: WARNING: rr [loose.c:744]: no socket found for match second RR route headers in the request are: Route: <sip:192.98.103.10:5060;transport=tcp;r2=on;ftag=43884fd989cff6dc;lr> Route: <sip:192.98.103.10:5060;r2=on;ftag=43884fd989cff6dc;lr> and when i start the proxy it tells this: Listening on udp: 192.98.103.2 [192.98.103.2]:5070 udp: 127.0.0.1:5072 tcp: 192.98.103.2 [192.98.103.2]:5070 tcp: 127.0.0.1:5072 tcp: 127.0.0.1:6060 Aliases: *: 192.98.103.10:5060 *: sip.test.fi:5060 why does loose_route() not find the first alias? -- juha

4 9

any easy way to access record-route URIs?
by Juha Heinanen 21 Sep '12

21 Sep '12

is there any easy way to access URIs in one or more record-route headers? there is @via[index], but no corresponding select for record- route URIs. -- juha

3 5

Re: [SR-Users] [sr-dev] kamailio core at qm_status ( patch required important)
by Jijo 21 Sep '12

21 Sep '12

Hi Daniel, This patch needs to be applied to avoid core. Thanks Jijo On Wed, Sep 19, 2012 at 10:54 AM, Jijo <realjijo(a)gmail.com> wrote: > > Hi All, > > Finally i found the issue, > > Here is one of the bad trace for SUBSCRIBE(722bytes) and NOTIFY(1282bytes) > which corrupted the memory. The messages came in the order NOTIFY and > SUBSCRIBE. The core is generated in a different place but I believe this > could be the reason for memory corruption. > > Here is the trace UDP Process 27294 processing NOTIFY and Process 27303 processing > SUBSCRIBE . > > > > The explanation and implementation is below > > > > 2012-09-19T02:06:17+01:00 [info] sipserver: [27303] INFO: <script>: > CI=1-3292(a)10.233.20.152 -R39 - Entry R-URI=1234 FD=10.233.20.152 > SI=10.233.20.152 > > 2012-09-19T02:06:17+01:00 [info] sipserver: [27294] INFO: <script>: > CI=1-3292(a)10.233.20.152 -R1 - Force LAN socket: tcp:10.233.20.151:5060<null> > > 2012-09-19T02:06:17+01:00 [info] sipserver: [27303] INFO: <script>: > CI=1-3292(a)10.233.20.152 -R1 - Force LAN socket: tcp:10.233.20.151:5060<null> > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:2357]: tcp_conn_send_put : calling wbufq_add > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:730]: ERROR: wbufq_add(722 bytes): buf:SUBSCRIBE > sip:1234@10.233.20.141:5063;transport=tcp SIP/2.0 > > Record-Route: <sip:10.233.20.151;tran > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:747]: ERROR: wbufq_add(722 bytes): first:b00519f4 last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:774]: ERROR: wbufq_add(2 last free crt_size:722): > first:b00519f4 last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27294] ERROR: <core> > [tcp_main.c:796]: ERROR: wbufq_insert(1282 bytes): buf:NOTIFY > sip:1234@10.233.20.141:5063;transport=tcp SIP/2.0 > > Record-Route: <sip:10.233.20.151;transpo > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27294] ERROR: <core> > [tcp_main.c:801]: ERROR: wbufq_insert(2 last free ): first:b00519f4 > last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27294] ERROR: <core> > [tcp_main.c:820]: ERROR: wbufq_insert(22 last free ): first:b00519f4 > last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27359] ERROR: <core> > [tcp_main.c:887]: ERROR: wbufq_run(3 last free ): first:b00519f4 > last:b00519f4 > > 2012-09-19T02:06:17+01:00 [crit] sipserver: [27359] : <core> > [mem/q_malloc.c:157]: BUG: qm_*: fragm. 0xb00519dc (address 0xb00519f4) end > overwritten(0, 0)! > > 2012-09-19T02:06:18+01:00 [alert] sipserver: [27265] ALERT: <core> > [main.c:755]: child process 27359 exited by a signal 11 > > 2012-09-19T02:06:18+01:00 [alert] sipserver: [27265] ALERT: <core> > [main.c:758]: core was generated > > 2012-09-19T02:06:18+01:00 [info] sipserver: [27265] INFO: <core> > [main.c:770]: INFO: terminating due to SIGCHLD > > > > Process 27294(NOTIFY) created the TCP connection structure for destination > IP and just before calling wbufq_insert(), context switch happened and > process 27303(SUBSCRIBE) got the cpu. Since the connection structure is > already available process 27303 add the SUBSCRIBE message(722 bytes) to > wbufq. Afterwards process 27294 got the CPU and invoked wbufq_insert() > which basically corrupted the memory due to an overflow with the existing > implementation. > > > > > > inline static int _wbufq_insert(struct tcp_connection* c, const char* > data, > > > unsigned int size) > > { > > struct tcp_wbuffer_queue* q; > > struct tcp_wbuffer* wb; > > > > q=&c->wbuf_q; > > if (likely(q->first==0)) /* if empty, use wbufq_add */ > > return _wbufq_add(c, data, size); > > : > > : > > : > > : > > if ((q->first==q->last) && > ((q->last->b_size-q->last_used)>=size)){ > > /* one block with enough space in it for > size bytes */ > > > > memmove(q->first->buf+size, > q->first->buf, size); > > memcpy(q->first->buf, data, size); > > q->last_used+=size; > > } > > > > The above condition shall be true in this case and memmove was moving the > pointer which was causing the overflow. > > memmove(void *dest, const void *src, size_t n); > > > > As per the memmove man page, the src shall be copied with size ‘n’ to a > temporary buffer and then temporary buffer to dest. > > > > dest is q->first->buf+size: which is basically (q->first->buf + NOTIFY > MSG SIZE). so the dst will move my 1282 bytes, so we have remaining space > of only 818 bytes. > > > > src is q->first->buf: which is basically copied to temp buffer with NOTIFY > SIZE(1282bytes). > > > > Finally we are moving the buffer from temporary buffer of size 1282 bytes > to buffer which we left with 818 bytes, This basically corrupt the memory > and on wbufq_run we see the memory corruption > > > > 2012-09-19T02:06:17+01:00 [crit] sipserver: [27359] : <core> > [mem/q_malloc.c:157]: BUG: qm_*: fragm. 0xb00519dc (address 0xb00519f4) end > overwritten(0, 0)! > > > > I think we don’t need memove, so we can change the code as below > > > > if ((q->first==q->last) && ((q->last->b_size-q->last_used)>=size)){ > > /* one block with enough space in it for size bytes */ > > //memmove(q->first->buf+size, q->first->buf, size); > > memcpy(q->first->buf+q->last_used, data, size); > > q->last_used+=size; > > } > > OR > we need only the else part as it always add the block to the first. > > Thanks > Jijo > > On Wed, Jul 18, 2012 at 12:42 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> sorry, I just read the last messages in the thread and I didn't noticed >> it is about a crash, but thought it is about an annoying log message. >> >> The backtrace is no longer matching the latest sources of the 3.1 branch, >> but I expect it is due to a double free issue, so you have to update to >> latest 3.1.6, as suggested by other users here. There were many fixes from >> 3.1.0 to 3.1.6. >> >> Cheers, >> Daniel >> >> >> On 7/17/12 6:03 PM, Jijo wrote: >> >> Hi, >> >> This is not happening at shutdown or status check. Its aborting when >> the system is active. >> >> Thanks >> Jijo >> On Tue, Jul 17, 2012 at 3:06 AM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> does it keep being or it is one time and that's it? >>> >>> That is printed at shut down or if pkg_status() or shm_status() is >>> executed from some part of code or in config via cfgutils module functions. >>> >>> You can get rid of them by setting memdbg and memlog to a value higher >>> than debug global parameter. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 7/16/12 5:28 PM, Jijo wrote: >>> >>> Thanks.. It is not easy to upgrade as it is happening at customer >>> system. >>> Is there any change occurred for this issue.I looked at it, but didn't >>> see anything in q_malloc.c/qm_status() >>> >>> >>> On Mon, Jul 16, 2012 at 11:12 AM, Jon Bonilla <manwe(a)aholab.ehu.es>wrote: >>> >>>> El Mon, 16 Jul 2012 10:27:42 -0400 >>>> Jijo <realjijo(a)gmail.com> escribió: >>>> >>>> > Hi All, >>>> > >>>> > I'm observing a core intermittently at "qm_status (qm=0x786cd000) at >>>> > mem/q_malloc.c:763" for kamailio version 3.1.0 >>>> > >>>> >>>> I'd say that you're using a very old version. You should update your >>>> branch to >>>> 3.1.6 or upgrade to a newer branch. >>>> >>>> >>>> _______________________________________________ >>>> sr-dev mailing list >>>> sr-dev(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>>> >>> >>> >>> >>> _______________________________________________ >>> sr-dev mailing listsr-dev@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu >>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw >> >> >

2 1

Kamailio OS Sysctl Tweaks
by Cenk İlker İzanlı 20 Sep '12

20 Sep '12

Hi, Is there any sysctl tweaks for OS to get the most of Kamailio as performance meaning? Thx

2 2

Kamailio Presence module
by Gary Shergill 20 Sep '12

20 Sep '12

Hi Kamailio Community, Recently installed Kamailio, plan to integrate with SylkServer. Tried the SIP IM feature, but can't seem to get it working. Logged on to Bria and Blink, two seperate accounts. They can call each other, but can't add each other for presence. So I installed the presence module with apt-get and edited the kamailio.cfg file to add the line; #!define WITH_PRESENCE Restarting the server comes up with the following error; ---------------------- root@opensep:~# vi /etc/kamailio/kamailio.cfg root@opensep:~# service kamailio restart Not starting Kamailio: invalid configuration file! -e 0(5114) : <core> [cfg.y:3591]: parse error in config file /etc/kamailio/kamailio.cfg, line 392, column 32-36: syntax error 0(5114) : <core> [cfg.y:3591]: parse error in config file /etc/kamailio/kamailio.cfg, line 392, column 32-36: Invalid arguments 0(5114) : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 392, column 37: ERROR: bad config file (3 errors) ---------------------- Those lines are as follows; ---------------------- #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif ---------------------- The exact place is where it says DBURL on both params. I am using Ubuntu 12.04 and Kamailio 3.3.x. Thank you for your time. Kind Regards, Gary Shergill

3 8

authentication for client applications
by David Thomson 20 Sep '12

20 Sep '12

Hi, I am working on a project where a custom sip client will be integrated into a suite of applications to provide voip. The sip client will be working with Kamailio. The goal is to ensure that the client is authorized for communication with kamailio before allowing any calls to be made. Conventional username/password authentication for individual users will also be used once the client has been authenticated. Currently other applications in the suite use a digital signature in the http headers when communicating with server processes. If the signature is validated by the server process then the applications identity is validated and communication with the server process is allowed. Is it possible to include a public key and digital signature in the register events and have kamailio perform the transformation to validate the client's identity? If so which module provides such functionality? Has something like this been implemented in the past? Thanks for any input. ttyl,Dave

3 2

call forking using dbaliases not working for un-NATed clients
by Yufei Tao 19 Sep '12

19 Sep '12

Hi I have a strange problem on forking calls to a group of users. For example I have two users y2 and y3 in dbaliases, both with alias_username 'group'. And y2 and y3 both registers with Kamailio fine. When I make a call to 'group' from a third client y1, what my kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to BRANCH_ALIASDB, where a lookup location will be done for each of the username resulting from lookup of dbaliases, something like this: y1--INVITE 'group'-->lookup dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup location---¬ | |-->relay ---->[BRANCH_ALIASDB]--->'y3'-->lookup location---- The all works well as long as all clients are NAT'ed. However when they are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only goes to one of the group members, e.g. y2 only. When checking the log, it seemed to have done the dbaliases lookup fine, and each location lookup successfully. But Kamailio only relayed y2's IP, e.g. to the client, while y3's to itself. When comparing the location table when clients are NAT'ed or not, I find that the 'received' column is only populated when I do fix_nated_register. And group calls only works when 'received' column is populated. That explains why when clients are NAT'ed group calls work, as I only do fix_nated_register if nat_uac_test returns true. But if this is the only reason, if two clients register using the same username, e.g. both as y3, and when 'received' column of location table is empty (no fix_nated_register done), I would expect a call to y3 should also only make 1 client ring. But in fact both of them rang! The flow is like: y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client registered as 'y3' | ---> IP of 2nd client registered as 'y3' While a call to 'group' (thus dbaliases lookup took place) under such un-NAT'ed set up made only 1 client ring. So I can make it work by always doing fix_nated_register. But I'm not clear about these things: - why does a lookup of dbaliases before lookup of location make such difference? - does lookup location work differently depending on whether it is called from trunk or from a route called from a branch route? Following is relevant parts from my config file: ############################################################# route[LOCATION] { if ( alias_db_lookup("dbaliases") ) { t_on_branch("BRANCH_ALIASDB"); # in branch_route[BRANCH_ALIASDB], # call another route that looks up location, # if not existent, call drop() } else { xlog("L_DBG","LOCATION: not alias - go to lookup location trunk\n"); route(LOCATION_TRUNK); # normal look up location and sending of 404 etc } ... ... } ############################################################# branch_route[BRANCH_ALIASDB] { xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; Method:$rm\n"); route(LOCATION_BRANCH); } route[LOCATION_BRANCH] { if (!lookup("location")) { # Drop this branch - it's going nowhere drop(); } } ############################################################# route[RELAY] { xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } ############################################################ route[NAT] { #!ifdef WITH_NAT xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); force_rport(); if (nat_uac_test("2")) { if (method=="REGISTER") { fix_nated_register(); xlog("L_DBG","NAT: Just done fix_nated_register in REGISTER message in NAT route\n"); } else { xlog("L_DBG","NAT: fix_nated_contact\n"); fix_nated_contact(); } setflag(FLT_NATS); } # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force media relay #!endif return; } ############################################################# And in the main route, route LOCATION and RELAY are the last two routes: route { ... route(NAT); ... ... # user location service route(LOCATION); route(RELAY); } Hope I have made it clear. Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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Re: [SR-Users] call forking using dbaliases not working for un-NATed clients
by Yufei Tao 19 Sep '12

19 Sep '12

In LOCATION_BRANCH, after lookup location, I've added: if ($du==$null) $du=$ru; And now everything works! In summary, either of the following would fix the problem: 1. call fix_nated_contact for all clients, so that 'received' column in location table is populated (so that $du will be set), or 2. set $du after lookup if it is empty Wonder if it is worth adding fixes in the source code. Thanks very much Klaus for pointing me to the right direction! Yufei Date: Wed, 19 Sep 2012 15:13:29 +0100 From: Yufei Tao <yufei.tao(a)redembedded.com> Subject: Re: [SR-Users] call forking using dbaliases not working for un-NATed clients To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5059D309.6030500(a)redembedded.com> Content-Type: text/plain; charset="ISO-8859-1" I should have made it clearer: $du is null both before and after lookup location in LOCATION_BRANCH when no fix_nated_register was done (thus 'received' column in location table was null). When fix_nated_register was done, $du for each branch was null before lookup location, but set to 'received' after lookup location as you said. But in this case (fix_nated_register done), everything works fine. So seems now the problem happens when: 'received' in location table is null, causing $du not to be set. I used to think $du is only set by record-route and thanks for clarifying this Even when without fix_nated_register (thus no 'received'), the lookup location for each branch, i.e. y2 and y3, set $ru to the 'contact' field successfully, but $du isn't set. But Kamailio did relay to the first branch (trunk?) and not the others. So the first branch is handled differently? So maybe when $du is null, should set it to $ru on the branches? Yufei On 19/09/12 11:53, Klaus Darilion wrote: > On 19.09.2012 12:24, Yufei Tao wrote: >> Hi Klaus >> >> Thanks for the reply! >> >> I check the $du, it is always null before and after the lookup. Is it >> only set when relaying to a proxy (from record-route), and not to a >> client? > That's strange. For NATed clients, $du must contain the 'received' > URI. Otherwise they can not be contacted as $ru contains the private > IP address. > >> When no fix_nated_register is called, the lookup location for both >> clients y2 and y3 is successful from the log, when printing out $ru >> after lookup. But seems Kamailio only relays to one client's IP while >> not others. I think there must be some differences when branch route is >> executed first time and second time as you said. As it feels like for >> the first branch (trunk?) it used the 'contact' column from the location >> table, and for the other branches, it tries to use 'received'? > It seems db_lookup() creates multiple branches. Is lookup() only > finding 1 contact in table or multiple contacts? > > regards > Klaus > >> Yufei >> >> On 18/09/12 18:49, Klaus Darilion wrote: >>> I suspect that the branch route is first executed for the NATed >>> client. Then the 'received' column is used as destination URI. When >>> executing the branch route again, the destination URI is still the >>> value from the previous branch, and lookup() will not overwrite is as >>> 'received' is not available. Then Kamailio sends the INVITE again to >>> the first client. >>> >>> You can try to set $du to Null before lookup(). ($du=null or >>> $du=$null, not sure what the correct syntax is). >>> >>> Another workaround is to use fix_nated_register() for every client >>> (the pragmatic and more secure approach). >>> >>> regards >>> Klaus >>> >>> On 18.09.2012 15:16, Yufei Tao wrote: >>>> Hi >>>> >>>> I have a strange problem on forking calls to a group of users. For >>>> example I have two users y2 and y3 in dbaliases, both with >>>> alias_username 'group'. And y2 and y3 both registers with Kamailio >>>> fine. >>>> When I make a call to 'group' from a third client y1, what my >>>> kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to >>>> BRANCH_ALIASDB, where a lookup location will be done for each of the >>>> username resulting from lookup of dbaliases, something like this: >>>> >>>> y1--INVITE 'group'-->lookup >>>> dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup >>>> location---? >>>> >>>> | |-->relay >>>> >>>> ---->[BRANCH_ALIASDB]--->'y3'-->lookup >>>> location---- >>>> >>>> The all works well as long as all clients are NAT'ed. However when >>>> they >>>> are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only >>>> goes to one of the group members, e.g. y2 only. When checking the log, >>>> it seemed to have done the dbaliases lookup fine, and each location >>>> lookup successfully. But Kamailio only relayed y2's IP, e.g. to the >>>> client, while y3's to itself. >>>> >>>> When comparing the location table when clients are NAT'ed or not, I >>>> find >>>> that the 'received' column is only populated when I do >>>> fix_nated_register. And group calls only works when 'received' >>>> column is >>>> populated. That explains why when clients are NAT'ed group calls work, >>>> as I only do fix_nated_register if nat_uac_test returns true. >>>> >>>> But if this is the only reason, if two clients register using the same >>>> username, e.g. both as y3, and when 'received' column of location >>>> table >>>> is empty (no fix_nated_register done), I would expect a call to y3 >>>> should also only make 1 client ring. But in fact both of them rang! >>>> The >>>> flow is like: >>>> >>>> y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client >>>> registered as 'y3' >>>> | >>>> ---> IP of 2nd client >>>> registered as 'y3' >>>> >>>> While a call to 'group' (thus dbaliases lookup took place) under such >>>> un-NAT'ed set up made only 1 client ring. >>>> >>>> So I can make it work by always doing fix_nated_register. But I'm not >>>> clear about these things: >>>> - why does a lookup of dbaliases before lookup of location make such >>>> difference? >>>> - does lookup location work differently depending on whether it is >>>> called from trunk or from a route called from a branch route? >>>> >>>> >>>> Following is relevant parts from my config file: >>>> >>>> ############################################################# >>>> route[LOCATION] >>>> { >>>> if ( alias_db_lookup("dbaliases") ) >>>> { >>>> t_on_branch("BRANCH_ALIASDB"); # in >>>> branch_route[BRANCH_ALIASDB], >>>> # call another route that >>>> looks up >>>> location, >>>> # if not existent, call drop() >>>> >>>> } >>>> else >>>> { >>>> xlog("L_DBG","LOCATION: not alias - go to lookup location >>>> trunk\n"); >>>> route(LOCATION_TRUNK); # normal look up location and sending of >>>> 404 etc >>>> } >>>> >>>> ... ... >>>> } >>>> ############################################################# >>>> branch_route[BRANCH_ALIASDB] >>>> { >>>> xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; >>>> Method:$rm\n"); >>>> route(LOCATION_BRANCH); >>>> } >>>> >>>> route[LOCATION_BRANCH] >>>> { >>>> if (!lookup("location")) >>>> { >>>> # Drop this branch - it's going nowhere >>>> drop(); >>>> } >>>> } >>>> ############################################################# >>>> route[RELAY] { >>>> xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); >>>> >>>> >>>> #!ifdef WITH_NAT >>>> if (check_route_param("nat=yes")) { >>>> setbflag(FLB_NATB); >>>> } >>>> if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { >>>> xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); >>>> route(RTPPROXY); >>>> } >>>> #!endif >>>> >>>> /* example how to enable some additional event routes */ >>>> if (is_method("INVITE")) { >>>> t_on_reply("REPLY_ONE"); >>>> t_on_failure("FAIL_ONE"); >>>> } >>>> >>>> if (!t_relay()) { >>>> sl_reply_error(); >>>> } >>>> >>>> exit; >>>> } >>>> >>>> ############################################################ >>>> route[NAT] { >>>> #!ifdef WITH_NAT >>>> xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); >>>> >>>> force_rport(); >>>> if (nat_uac_test("2")) { >>>> if (method=="REGISTER") { >>>> fix_nated_register(); >>>> xlog("L_DBG","NAT: Just done fix_nated_register in >>>> REGISTER >>>> message in NAT route\n"); >>>> } else { >>>> xlog("L_DBG","NAT: fix_nated_contact\n"); >>>> fix_nated_contact(); >>>> } >>>> setflag(FLT_NATS); >>>> } >>>> >>>> # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force >>>> media relay >>>> #!endif >>>> return; >>>> } >>>> >>>> ############################################################# >>>> And in the main route, route LOCATION and RELAY are the last two >>>> routes: >>>> route { >>>> ... >>>> route(NAT); >>>> >>>> ... ... >>>> >>>> # user location service >>>> route(LOCATION); >>>> >>>> route(RELAY); >>>> } >>>> >>>> Hope I have made it clear. Thanks very much! >>>> >>>> Yufei >>>> -- >>>> Yufei Tao >>>> Red Embedded -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

1 0

IM on Kamailio
by Gary Shergill 19 Sep '12

19 Sep '12

Hi Kamailio Community, I've been configuring the presence module on Kamailio and trying to get it working, using the following as a guideline; http://nil.uniza.sk/instant-messaging/simple/configuring-im-and-presence-ka… >From that I assumed that enabling Presence would mean that IM would be enabled too. However, that doesn't seem to be the case. May I ask please how to configure IM on Kamailio? Note that I am testing this with one computer connected by Bria and another computer connected via Blink. I am able to log on to a user on each (test1 and test2) and they are able to call each other. The issue is, with presence enabled, they are unable to IM each other (or add each other as contacts and see online status). Thank you. Kind Regards, Gary Shergill

4 4

Kamailio-Redirect based LCR routing
by Fatima Chahrour~Vanrise Support 19 Sep '12

19 Sep '12

Dear Daniel\List, Following up to a previous post where I was able in my testing lab to redirect call with 1st destination of lcr_gw table to another Kamailio(acting as any SBC) with Daniels assist, I need a hand in redirecting the call with multiple contacts , basically based on the lcr routing rule configured in kamailio LCR module for the rerouted call. Is this possible using code 300 with multiple choices message? Or you advise me with another method? If 300 redirect message works, what should be the variable set as $ru to get all lcr_gw destinations? Example for clarification: My current redirect script used after lcr functions is : .. $ru= $ru; sl_send_reply("302","Moved Temporary"); .. This script is successfully sending the 302 message with Contact: sip:1119613454646@192.168.111.195. (1119613454646 is the dialed number with prefix, 192.168.111.195 is first lcr_gw ip). Now if the call to 192.168.111.195 is failed and I need to redirect the second destination, second lcr_gw ip, how can I achieve this? Thank you for your usual response. Kind Regards, F Chahrour

2 4

AUTOREPLY call forking using dbaliases not working for...
by strobel＠hfo-telecom.de 18 Sep '12

18 Sep '12

******************* AUTOREPLY VON yufei.tao(a)redembedded.com ***************************** Vielen Dank für Ihre Nachricht. Leider bin ich in der Zeit vom 17.09.2012 bis einschließlich 21.09.2012 nicht im Haus. Ihre E-Mail wird in meiner Abwesenheit nicht weitergeleitet. Bitte wenden Sie sich in dringenden Fällen an meine Kollegen Herrn Künzel oder Herrn Schrenk. eMail: technik(a)hfo-telecom.de Phone: +49 9281 / 1448 - 100 Fax: +49 9281 / 1448 - 123 Mit freundlichen Grüßen, Rainer Strobel Network Management HFO Telecom AG Bahnhofstr. 18 D-95028 Hof / Saale Hi I have a strange problem on forking calls to a group of users. For example I have two users y2 and y3 in dbaliases, both with alias_username 'group'. And y2 and y3 both registers with Kamailio fine. When I make a call to 'group' from a third client y1, what my kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to BRANCH_ALIASDB, where a lookup location will be done for each of the username resulting from lookup of dbaliases, something like this: y1--INVITE 'group'-->lookup dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup location---¬ | |-->relay ---->[BRANCH_ALIASDB]--->'y3'-->lookup location---- The all works well as long as all clients are NAT'ed. However when they are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only goes to one of the group members, e.g. y2 only. When checking the log, it seemed to have done the dbaliases lookup fine, and each location lookup successfully. But Kamailio only relayed y2's IP, e.g. to the client, while y3's to itself. When comparing the location table when clients are NAT'ed or not, I find that the 'received' column is only populated when I do fix_nated_register. And group calls only works when 'received' column is populated. That explains why when clients are NAT'ed group calls work, as I only do fix_nated_register if nat_uac_test returns true. But if this is the only reason, if two clients register using the same username, e.g. both as y3, and when 'received' column of location table is empty (no fix_nated_register done), I would expect a call to y3 should also only make 1 client ring. But in fact both of them rang! The flow is like: y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client registered as 'y3' | ---> IP of 2nd client registered as 'y3' While a call to 'group' (thus dbaliases lookup took place) under such un-NAT'ed set up made only 1 client ring. So I can make it work by always doing fix_nated_register. But I'm not clear about these things: - why does a lookup of dbaliases before lookup of location make such difference? - does lookup location work differently depending on whether it is called from trunk or from a route called from a branch route? Following is relevant parts from my config file: ############################################################# route[LOCATION] { if ( alias_db_lookup("dbaliases") ) { t_on_branch("BRANCH_ALIASDB"); # in branch_route[BRANCH_ALIASDB], # call another route that looks up location, # if not existent, call drop() } else { xlog("L_DBG","LOCATION: not alias - go to lookup location trunk\n"); route(LOCATION_TRUNK); # normal look up location and sending of 404 etc } ... ... } ############################################################# branch_route[BRANCH_ALIASDB] { xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; Method:$rm\n"); route(LOCATION_BRANCH); } route[LOCATION_BRANCH] { if (!lookup("location")) { # Drop this branch - it's going nowhere drop(); } } ############################################################# route[RELAY] { xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } ############################################################ route[NAT] { #!ifdef WITH_NAT xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); force_rport(); if (nat_uac_test("2")) { if (method=="REGISTER") { fix_nated_register(); xlog("L_DBG","NAT: Just done fix_nated_register in REGISTER message in NAT route\n"); } else { xlog("L_DBG","NAT: fix_nated_contact\n"); fix_nated_contact(); } setflag(FLT_NATS); } # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force media relay #!endif return; } ############################################################# And in the main route, route LOCATION and RELAY are the last two routes: route { ... route(NAT); ... ... # user location service route(LOCATION); route(RELAY); } Hope I have made it clear. Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ _______________________________________________ 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

record_route_advertised_address question
by Juha Heinanen 18 Sep '12

18 Sep '12

rr readme on record_route_advertised_address function tells: Note: A second Record-Route will be inserted if the transport used on the inbound and outbound interfaces changes. that is not the case. i made call record_route_advertised_address("192.98.103.10:5060"); when request came over udp and went out over tcp and only one rr header was inserted: incoming: U 2012/09/17 17:30:17.467079 192.98.103.10:5060 -> 192.98.103.2:5070 INVITE sip:test@test.fi SIP/2.0. ... outgoing: T 2012/09/17 17:30:17.504631 192.98.103.10:5060 -> 192.98.103.10:38078 [AP] INVITE sip:0x20b7580@192.98.103.10:5050;transport=tcp SIP/2.0. Record-Route: <sip:192.98.103.10:5060;ftag=wumod;lr>. ... i have not set enable_double_rr param and default is 1. is this a bug that i should report on the tracker? -- juha

3 5

Port change 5060
by Vijay Thakur 18 Sep '12

18 Sep '12

All Experts, I want to change my SIP port from 5060 for other one. Before making any change in my live server (Kamailio 3.2.1, i want to be sure. Kindly suggest me that where should i make changes to implement it. This is a security measure for kamailio from port scanning. With thanks in advance. Vijay Thakur

5 6

pidf-lo extraction
by Cenk İlker İzanlı 17 Sep '12

17 Sep '12

Hi, Is there any basic document showing how to extract info from pidf-lo data on kamailio part? Thx.

2 1

rsyslog settings for good kamailio performance
by Albert Petit 17 Sep '12

17 Sep '12

Hi, For load testing I would like to setup rsyslog so Kamailio logs are written asynchronously Would the following configuration be enough for that purpose in * rsyslog.conf*? # # Write log file in asynchronous mode and use an IO Buffer of 64 Kb # IO buffer config http://www.rsyslog.com/doc/rsyslog_conf_global.html # # $OMFileAsyncWriting on $OMFileFlushOnTXEnd off $OMFileFlushInterval 60 $OMFileIOBufferSize 128k #---------------------------------------------------------------------- ## kamailio ##---------------------------------------------------------------------- $outchannel kamailio_log_rotation,/var/log/pocserverplatform/kamailio.log,524288000,/usr/local/bin/kamailio_log_rotation_script :programname, contains, "kamailio" $kamailio_log_rotation local0.* $kamailio_log_rotation Apart of that is there a way to completely disable completely any logs of Kamailio to discard they're causing any performance issue? Thanks Albert

3 2

kamailio prepaid services
by Cenk İlker İzanlı 15 Sep '12

15 Sep '12

Hi, Is there any guide to install/configure kamailio as prepaid service infrastructure? Thx

1 0

Re: [SR-Users] Limiting concurrent calls
by Alex Balashov 15 Sep '12

15 Sep '12

Htable is another popular method for those disposed less favourably to the dialog module. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Uri Shacked <ushacked(a)gmail.com> wrote:in a few words - you can set avp's as the profile params in the dialog module (that module works great for me) that way you wont need to set more than one profile. and for every call use it. or just use $shv() and +1 or -1 every INVITE, CANCEL, BYE or failure reply. hope it helps, Uri Hi, I need to limit the number of concurrent calls go through kamailio. Its a simple set up(Not per user basis, a global limit will suffice). I know i need to employ dialog module. But i'm a little confused about profiles. I thought for simple application like this, i may be able to use profile altogether. Any code snippet to limit number of concurrent calls will be very helpful. Thanks in advance. -- -aft

1 0

Limiting concurrent calls
by Uri Shacked 15 Sep '12

15 Sep '12

in a few words - you can set avp's as the profile params in the dialog module (that module works great for me) that way you wont need to set more than one profile. and for every call use it. or just use $shv() and +1 or -1 every INVITE, CANCEL, BYE or failure reply. hope it helps, Uri > Hi, > I need to limit the number of concurrent calls go through kamailio. > Its a simple set up(Not per user basis, a global limit will suffice). > > I know i need to employ dialog module. But i'm a little confused about > profiles. I thought for simple application like this, i may be able to > use profile altogether. > > Any code snippet to limit number of concurrent calls will be very helpful. > > Thanks in advance. > -- > -aft

1 0

I get no calls?
by gooly 15 Sep '12

15 Sep '12

Hi, I have registered my (Fritz!Box) ATA successfully registered on my account 8305234(a)iptl.org but what is tried I never received a call? Here is what I can see on 'my' web-page with you: > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:58 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:56 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:45 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:45 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-09-10 13:43 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-09-10 13:43 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:40 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:40 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:39 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:39 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-08-22 00:20 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-08-22 00:19 500 Server Internal Error Please can you tell me what I have to do to be able to receive Voip-Calls? Regards, Carl

2 1

Limiting concurrent calls
by Aft nix 15 Sep '12

15 Sep '12

Hi, I need to limit the number of concurrent calls go through kamailio. Its a simple set up(Not per user basis, a global limit will suffice). I know i need to employ dialog module. But i'm a little confused about profiles. I thought for simple application like this, i may be able to use profile altogether. Any code snippet to limit number of concurrent calls will be very helpful. Thanks in advance. -- -aft

1 0

cfg. parser: failed to find command force_rtp_proxy
by Cenk İlker İzanlı 15 Sep '12

15 Sep '12

Hi, I'm installing kamailio and rtpproxy with this guide. http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour Versions of the programs: # aptitude versions rtpproxy i 1.2.1-1 stable 500 root@li304-234:~# kamailio -V version: kamailio 3.3.0 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 22:18:19 Jun 18 2012 with gcc 4.4.5 When I start kamailio, I get this error, Sep 14 17:08:44 (none) kamailio: ERROR: <core> [cfg.y:3455]: cfg. parser: failed to find command force_rtp_proxy Sep 14 17:08:44 (none) kamailio: : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 729, column 19: unknown command, missing loadmodule?#012 Sep 14 17:08:44 (none) kamailio: ERROR: <core> [cfg.y:3455]: cfg. parser: failed to find command force_rtp_proxy Sep 14 17:08:44 (none) kamailio: : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 806, column 19: unknown command, missing loadmodule?#012 Related lines throwing error are below. 724 route[RTPPROXY] { 725 #!ifdef WITH_NAT 726 if (is_method("BYE")) { 727 unforce_rtp_proxy(); 728 } else if (is_method("INVITE")){ 729 force_rtp_proxy(); 730 } 731 if (!has_totag()) add_rr_param(";nat=yes"); 732 #!endif 733 return; 734 } 735 803 #!ifdef WITH_NAT 804 if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) 805 && status=~"(183)|(2[0-9][0-9])") { 806 force_rtp_proxy(); 807 } 808 if (isbflagset("6")) { 809 fix_nated_contact(); 810 } 811 #!endif Does anyone know what's the problem with cfg file? Thx. Cenk

2 2

[Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Joe Hart 15 Sep '12

15 Sep '12

Hi all, For a project on which I'm currently working, I am having some problems figuring out how to correctly configure Kamailio to communicate with RTP Proxy in order to send media into and out of a network with private IP address ranges. I have a proxy set up to send the SIP traffic, and all of this is working fine. However, I'm having some trouble getting the RTP Proxy set up. Currently, when the call is connected, the offer/answer is made and RTP Proxy seems to be taking over, but I'm having trouble getting my audio to flow in both directions. Examination of the traffic coming into and out of this machine seems to indicate that the IP addresses aren't being mangled correctly. Specifically, it appears the internal IP address isn't being changed to reflect the IP address of the machine on which RTP Proxy is running, so that when the caller tries to send audio back, the IP it's given to reply to is 10.10.x.x, which obviously won't work. I have tried experimenting with specifically setting IP addresses in the rtpproxy_offer() and _answer() methods to no avail, as well as setting various flags in those methods. However, I must admit that I'm not entirely sure what's happening under the hood with these methods, or what rtpproxy is doing with that information when it gets it. Rather than continue to hack at this by trial and error, I'm hoping someone here can point me in the right direction. Any advice, example code or pep talks would be greatly appreciated. Thanks in advance, -- Joe Hart Voice Systems Integrator Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0673

8 37

Howto add peers and routing rules
by Joel Smith | VOZELIA 15 Sep '12

15 Sep '12

Hi, I'm getting started with kamailio and siremis. I've just finished getting the server ready and the software installed. Once in the siremis admin menu, i've added a domian and a subscriber and I have my softphone registered with kamailio. My doubt is, where do I add peers so I can set up the routing rules? Are there any tutorials? Thanks in advanced. Best regards. -- Joel Smith Cell: +34 639 03 13 53 E-Mail: joel(a)vozelia.com <joel(a)vozelia.com> <http://www.vozelia.com> <http://twitter.com/vozelia> <http://www.facebook.com/pages/Vozelia-Operador-de-telefonia-IP-para-empresa…>

2 1

Re: [SR-Users] using kamailio with clients in a nat environment
by David Thomson 15 Sep '12

15 Sep '12

Sorry, his name is Daniel Goepp. -----Original Message----- From: jdavidthomson(a)hotmail.com Date: Sat, 15 Sep 2012 03:53:51 To: Daniel-Constantin Mierla<miconda(a)gmail.com> Reply-To: jdavidthomson(a)hotmail.com Cc: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] using kamailio with clients in a nat environment I got it working. The advertise parameter wasn't available in the initial versions of rtpproxy I was using. A guy named daniel geopp has a patched version that enables advertisement of the public ec2 ip. Thanks for your help :) Ttyl, Dave -----Original Message----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Tue, 11 Sep 2012 06:24:33 To: <jdavidthomson(a)hotmail.com> Cc: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] using kamailio with clients in a nat environment Hello, the sdp does not show that rtpproxy was engaged. Check your config, you can use debugger module with cfgtrace on to see what actions are executed. Also, probably you have to advertise the public ip address of your ec2 instance -- see second parameter for rtpproxy module functions. Cheers, Daniel On 9/11/12 3:52 AM, David Thomson wrote: Hi, I'm using rtpproxy and per the documentation: rtpproxy -l public_ip -s udp:localhost:22222 -F Attached is the following: Dave registering Daniel registering Dave calling Daniel, where Dave has a public IP and Daniel is behind a nat. Please let me know what you think is up. ttyl, Dave [...] # U 207.219.69.217:40821 -> 10.248.96.110:5060 INVITE sip:daniel@54.245.31.65 SIP/2.0. Via: SIP/2.0/UDP 10.207.158.89:51362;rport;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 70. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Route: <sip:54.245.31.65;transport=udp;lr> <sip:54.245.31.65;transport=udp;lr> . Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. # U 10.248.96.110:5060 -> 207.219.69.217:40821 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.207.158.89:51362;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF;received=207.219.69.217. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Server: kamailio (3.3.0 (x86_64/linux)). Content-Length: 0. . # U 10.248.96.110:5060 -> 75.119.228.57:5060 INVITE sip:daniel@192.168.1.102:5060;transport=udp;registering_acc=54_245_31_65 SIP/2.0. Record-Route: <sip:54.245.31.65;lr=on;nat=yes> <sip:54.245.31.65;lr=on;nat=yes> . Via: SIP/2.0/UDP 54.245.31.65:5060;branch=z9hG4bKd6f6.b6db86e5.0. Via: SIP/2.0/UDP 10.207.158.89:51362;received=207.219.69.217;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 69. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Re: [SR-Users] using kamailio with clients in a nat environment
by David Thomson 15 Sep '12

15 Sep '12

I got it working. The advertise parameter wasn't available in the initial versions of rtpproxy I was using. A guy named daniel geopp has a patched version that enables advertisement of the public ec2 ip. Thanks for your help :) Ttyl, Dave -----Original Message----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Tue, 11 Sep 2012 06:24:33 To: <jdavidthomson(a)hotmail.com> Cc: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] using kamailio with clients in a nat environment Hello, the sdp does not show that rtpproxy was engaged. Check your config, you can use debugger module with cfgtrace on to see what actions are executed. Also, probably you have to advertise the public ip address of your ec2 instance -- see second parameter for rtpproxy module functions. Cheers, Daniel On 9/11/12 3:52 AM, David Thomson wrote: Hi, I'm using rtpproxy and per the documentation: rtpproxy -l public_ip -s udp:localhost:22222 -F Attached is the following: Dave registering Daniel registering Dave calling Daniel, where Dave has a public IP and Daniel is behind a nat. Please let me know what you think is up. ttyl, Dave [...] # U 207.219.69.217:40821 -> 10.248.96.110:5060 INVITE sip:daniel@54.245.31.65 SIP/2.0. Via: SIP/2.0/UDP 10.207.158.89:51362;rport;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 70. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Route: <sip:54.245.31.65;transport=udp;lr> <sip:54.245.31.65;transport=udp;lr> . Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. # U 10.248.96.110:5060 -> 207.219.69.217:40821 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.207.158.89:51362;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF;received=207.219.69.217. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Server: kamailio (3.3.0 (x86_64/linux)). Content-Length: 0. . # U 10.248.96.110:5060 -> 75.119.228.57:5060 INVITE sip:daniel@192.168.1.102:5060;transport=udp;registering_acc=54_245_31_65 SIP/2.0. Record-Route: <sip:54.245.31.65;lr=on;nat=yes> <sip:54.245.31.65;lr=on;nat=yes> . Via: SIP/2.0/UDP 54.245.31.65:5060;branch=z9hG4bKd6f6.b6db86e5.0. Via: SIP/2.0/UDP 10.207.158.89:51362;received=207.219.69.217;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 69. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

avp_radius load time high
by Ricardo Martinez 14 Sep '12

14 Sep '12

> Hello List. > > I’m having the next issue. I have a kamailio v3.2 working with a Radius > server as a backend. All the INVITE’s coming to the kamailio server are > challenged against the Radius server and then avp_load is called from > Radius server too. I’m having excessive delay times when I have near 10 > call per second. So I find it very weird. I have made a benchmark for the > avp_load : > > bm_start_timer("timer-tranum"); > > if( !radius_load_caller_avps("$fU@$fd")) { > > sl_send_reply("403","Forbidden - Estamos experimentando problemas"); > > exit; > > } else { > > xlog("L_INFO", "[$ci]:[AUTH_REQUEST]: ok"); > > }; > > bm_log_timer("timer-tranum"); > > > > and this is the output : > > > > Sep 12 18:26:07 benchmark (timer timer-tranum [1]): 112441 [ > msgs/total/min/max/avg - LR: > 100/18446744073695545751/23297/18446744073709236247/184467440736955456.000000 > | GB: > 2100/18446744073708471097/10746/18446744073709542058/8784163844623081.000000] > > Sep 12 18:26:55 benchmark (timer timer-tranum [1]): 280543 [ > msgs/total/min/max/avg - LR: > 100/10238598/21317/18446744073708596212/102385.980000 | GB: > 2200/9158079/10746/18446744073709542058/4162.763182] > > Sep 12 18:28:20 benchmark (timer timer-tranum [1]): 312902 [ > msgs/total/min/max/avg - LR: > 100/5048656/15491/18446744073709546022/50486.560000 | GB: > 2300/14206735/10746/18446744073709546022/6176.841304] > > Sep 12 18:28:31 benchmark (timer timer-tranum [1]): 301818 [ > msgs/total/min/max/avg - LR: > 100/5622716/43785/18446744073708810522/56227.160000 | GB: > 2400/19829451/10746/18446744073709546022/8262.271250] > > Sep 12 18:28:44 benchmark (timer timer-tranum [1]): 18446744073708689124 > [ msgs/total/min/max/avg - LR: > 100/1329199/59661/18446744073709167366/13291.990000 | GB: > 2500/21158650/10746/18446744073709546022/8463.460000] > > Sep 12 18:29:26 benchmark (timer timer-tranum [1]): 44160 [ > msgs/total/min/max/avg - LR: > 100/18446744073701320457/19576/18446744073708922050/184467440737013216.000000 > | GB: 2600/12927491/10746/18446744073709546022/4972.111923] > > Sep 12 18:30:18 benchmark (timer timer-tranum [1]): 218573 [ > msgs/total/min/max/avg - LR: > 100/18446744073707306935/17459/18446744073708876520/184467440737073056.000000 > | GB: 2700/10682810/10746/18446744073709546022/3956.596296] > > Sep 12 18:31:06 benchmark (timer timer-tranum [1]): 132751 [ > msgs/total/min/max/avg - LR: > 100/9611955/17057/18446744073708726431/96119.550000 | GB: > 2800/20294765/10746/18446744073709546022/7248.130357] > > Sep 12 18:31:55 benchmark (timer timer-tranum [1]): 18859 [ > msgs/total/min/max/avg - LR: > 100/18446744073705270953/18859/18446744073708747515/184467440737052704.000000 > | GB: 2900/16014102/10746/18446744073709546022/5522.104138] > > Sep 12 18:32:48 benchmark (timer timer-tranum [1]): 61954 [ > msgs/total/min/max/avg - LR: > 100/7124234/18036/18446744073708596768/71242.340000 | GB: > 3000/23138336/10746/18446744073709546022/7712.778667] > > Sep 12 18:33:39 benchmark (timer timer-tranum [1]): 53489 [ > msgs/total/min/max/avg - LR: > 100/18446744073707180114/18177/18446744073708790040/184467440737071808.000000 > | GB: 3100/20766834/10746/18446744073709546022/6698.978710] > > Sep 12 18:34:35 benchmark (timer timer-tranum [1]): 60604 [ > msgs/total/min/max/avg - LR: > 100/18446744073701108277/33983/18446744073708839351/184467440737011072.000000 > | GB: 3200/12323495/10746/18446744073709546022/3851.092187] > > Sep 12 18:35:24 benchmark (timer timer-tranum [1]): 54705 [ > msgs/total/min/max/avg - LR: > 100/7185862/15880/18446744073708608536/71858.620000 | GB: > 3300/19509357/10746/18446744073709546022/5911.926364] > > Sep 12 18:36:21 benchmark (timer timer-tranum [1]): 60356 [ > msgs/total/min/max/avg - LR: > 100/18446744073705699677/19466/18446744073708844524/184467440737056992.000000 > | GB: 3400/15657418/10746/18446744073709546022/4605.122941] > > Sep 12 18:37:14 benchmark (timer timer-tranum [1]): 18446744073708679441 > [ msgs/total/min/max/avg - LR: > 100/3506212/16102/18446744073708791471/35062.120000 | GB: > 3500/19163630/10746/18446744073709546022/5475.322857] > > Sep 12 18:38:07 benchmark (timer timer-tranum [1]): 93933 [ > msgs/total/min/max/avg - LR: > 100/18446744073704680760/17569/18446744073708741284/184467440737046816.000000 > | GB: 3600/14292774/10746/18446744073709546022/3970.215000] > > Sep 12 18:38:58 benchmark (timer timer-tranum [1]): 142777 [ > msgs/total/min/max/avg - LR: > 100/681953/15933/18446744073708708694/6819.530000 | GB: > 3700/14974727/10746/18446744073709546022/4047.223514] > > Sep 12 18:39:44 benchmark (timer timer-tranum [1]): 165921 [ > msgs/total/min/max/avg - LR: > 100/5153736/15749/18446744073708657622/51537.360000 | GB: > 3800/20128463/10746/18446744073709546022/5296.963947] > > Sep 12 18:40:31 benchmark (timer timer-tranum [1]): 127615 [ > msgs/total/min/max/avg - LR: > 100/18446744073709161716/20690/18446744073708793559/184467440737091616.000000 > | GB: 3900/19738563/10746/18446744073709546022/5061.170000] > > Sep 12 18:41:21 benchmark (timer timer-tranum [1]): 47762 [ > msgs/total/min/max/avg - LR: > 100/9001175/18909/18446744073708632234/90011.750000 | GB: > 4000/28739738/10746/18446744073709546022/7184.934500] > > Sep 12 18:42:11 benchmark (timer timer-tranum [1]): 62944 [ > msgs/total/min/max/avg - LR: > 100/18446744073708163674/17934/18446744073708825788/184467440737081632.000000 > | GB: 4100/27351796/10746/18446744073709546022/6671.169756] > > > > > > If the output is in microseconds and I have from time to time (for the > last 100 messages) values like 184467440737081632.000000 microsecs > > IS THIS NORMAL? > > Then it backs to a more rational value : 90011.750000 microsecs. > > I also made measures in my Radius Server and is answering with average > time around 0.01 secs. > > So… what could it be the problem?.. Could be the RadiusClient-ng ? > > In my attempt to solve the issue I increased the childrens for the > kamailio server from 16 to 60, but the problem persists. > > Can someone help me here? > > > > Best Regards, > > Ricardo Martinez.- > > > > > > >

3 7

collecting stats per subscriber
by Krishna Kurapati 14 Sep '12

14 Sep '12

Hi, Is there a module that can be used to collect number of MESSAGEs sent or received by a subscriber; number of registrations by a subscriber? These will be per subscriber as oppose to statistics collected for entire system. Thanks Krish Kura

2 2

How to do RTPProxy load balancing?
by Mino Haluz 14 Sep '12

14 Sep '12

Hi, mediaproxy has dispatcher and relay, so it load balances the traffic automatically. How it is done with rtpproxy ? Thanks, Mino

3 4

FAQ page on the wiki
by Daniel-Constantin Mierla 14 Sep '12

14 Sep '12

Hello, I just started a FAQ page on the wiki site. Hopefully people will collect there useful content, based on their experience, questions that repeat on the mailing lists or other sources. Added just a few content for the moment, to show the syntax -- the page is available at: * http://www.kamailio.org/wiki/tutorials/faq/main It is using the QNA plugin for dokuwiki: * https://www.dokuwiki.org/plugin:qna Anyone can make an own account to edit the wiki, the registration form is available at: * https://www.kamailio.org/wiki/start?do=register Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Re: [SR-Users] Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

thank you for helps cenk and pavel. I see that I had wrong information about kamailio.

2 1

Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

juha writes: > this kind of questions have been asked many times on this list. search > archive for responses. you are right maybe but it is hard to search the archieve. And if it is easier to say "serach" than please show me way to search the arcieve quickly. In this link I m trying to search archieve month by month: http://lists.sip-router.org/pipermail/sr-users/ I m new and thanks your kind help...

3 2

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 14 Sep '12

14 Sep '12

Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or CANCEL. It'll just figure out what to do on its own. None of this has to do with dialog state, though. Just rtpproxy control. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: > >> Peter: Thanks for the tip! Really interesting. But I do not >> understand, why also this list contains the calls that were ended by >> sipp... Should I search for some mistake in my kamaillio config ? > > Perhaps you don't close them with unforce_rtp_proxy: > > if(method=="BYE" || method=="CANCEL"){ > unforce_rtp_proxy(); > } > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > 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 _______________________________________________ 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

4 7

Re: [SR-Users] Max-Forwards: and loops. Testing purpose.
by Daniel-Constantin Mierla 14 Sep '12

14 Sep '12

Hello, do not write private emails from mailing list discussions, if not sensitive data was asked explicitly. If there is no rule to breaking the loop, after some time the message gets very big. Cheers, Daniel On 9/14/12 7:03 AM, Anton Kvashenkin wrote: > Thanks for respond. I just have used forward() based on RURI. > > got > > 0(2558) ERROR: <core> [msg_translator.c:1835]: ERROR: > build_req_buf_from_sip_req: out of memory > 0(2558) ERROR: <core> [forward.c:601]: ERROR: forward_request: > building failed > 0(2558) WARNING: <core> [receive.c:211]: WARNING: receive_msg: error > while trying script > > With maxfwd it works like a charm. http://paste.ubuntu.com/1204093/ > > 2012/9/12 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > > On 9/11/12 1:12 PM, Anton Kvashenkin wrote: > > Hi, List. > > I'm using sipsak to fire some predefined sip message with > Max-Forwards: header equal 0 to test mf_process_maxfwd_header > function. I'm doing this exclusively for learning purpose. As > far as we know, this exported function from maxfwd module > services only for adding Max-Forwards header, decrement or > detect Max-Forwards: 0 header to prevent loops. So, my > question guys, how can I create this loop in my lab, with two > proxies for example? What message should I craft? > > what kind of message are you looking to craft? > > Looping from one proxy to another can be achieved with > t_relay("_the other proxy address_") function from tm module. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - > http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - > http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - > http://asipto.com/u/katu > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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