sr-users February 2016

sr-users@lists.kamailio.org

80 participants
133 discussions

siptrace module plain sip mode
by Grant Bagdasarian 17 Feb '16

17 Feb '16

Hello, The documentation of the siptrace module specifies duplication of sip messages to a capture server using the HEP protocol or plain SIP mode. Which params to I need to set (and with which values) in the siptrace and sipcapture to enable plain SIP mode? I tried several configurations but none of the worked. Seems like only HEP works. My test kamailio (sip capture node with sipcapture module) doesn't receive the duplicated SIP messages. Any ideas? Regards, Grant

2 2

Installatioin issues
by Ed Todd 17 Feb '16

17 Feb '16

I am installing KAMAILIO under DEBIAN. I have got it to work fine (TLS) until I upgraded my Comcast to business service to get static IP’s. Now it no longer works. I am using the Cisco DPC3941 modem/router and am tring to connect an IPhone on WIFI to the PC SIP server on the LAN. Here are the log entries: Feb 16 19:14:59 SIP-Server kamailio[2010]: NOTICE: <core> [main.c:738]: handle_sigs(): Thank you for flying kamailio!!! Feb 16 19:14:59 SIP-Server kamailio[2019]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 16 20:01:25 SIP-Server kamailio: INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... Feb 16 20:01:26 SIP-Server kamailio: INFO: <core> [tcp_main.c:4745]: init_tcp(): using epoll_lt as the io watch method (auto detected) Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: ctl [init_socks.c:126]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: ctl [ctl.c:283]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module ctl (/usr/lib/x86_64-linux-gnu/kamailio/modules/ctl.so) I am informed that SIP-ALG is disabled. Any ideas?

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Remove_hf_is not working
by aman jeff 17 Feb '16

17 Feb '16

Hi , I am working kamailio 4.4.0 ,i want to remove my Contact header completely.So I used remove_hf..but its now working ..Let me know the solution Am very thankful to you Regards Jefff

2 1

db_cluster.so module and kamctlrc DBHOST= field
by Hosted Services 16 Feb '16

16 Feb '16

Hi there, New user to Kamailio here. We currently have it up and running in a virtualized environment with 1 Kamailio sever, 1 Asterisk server and 1 MySQL server. I'm currently writing install scripts to make deploying new nodes/servers easy and to keep settings the same across the board. I've chosed to load the db_cluster.so module in kamailio.cfg, as we will have 2x MySQL servers in master-master replication which will contain the 'kamailio' and 'asterisk' tables. I've just hit a stumbling block - in `kamctlrc`, there is a field called `DBHOST=`. How can I reference my cluster here? In kamailio.cfg, I simply define DBURL as "cluster//<clustername>". What is the syntax for 'DBHOST=' in 'kamctlrc'? Can I reference the cluster? Can I have 2 separate DBHOST= lines? Looking for some guidance on this one. Thanks, Derek B.

2 1

Database Insert
by Daniel W. Graham 16 Feb '16

16 Feb '16

I see there is a new SQLops function sql_query_async that has been added in 4.2 (I believe?). Is there a good way to perform database insert using version 4.0? Using mysql Thanks! -dan

2 4

Need to validate a IP address with subnet mask
by Sunil More 16 Feb '16

16 Feb '16

Hi All, I want to validate a IP address in kamailio for requests coming from different IPs. I have browsed and found permissions module, however is there a way where i can directly compare a source IP address to an address stored in an avp and get a true or false value. something like this $si==192.168.2.0/24 Thanking You, Sunil More

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Textops and Multipart Body - adding ISUP
by Sergey Basov 16 Feb '16

16 Feb '16

Hi All. I need to use SIP-I with my upstream to set A number as unknown. I was read about adding incapsulation ISUP using if(has_body("application/sdp")) { set_body_multipart(); msg_apply_changes() $var(acm) = "7e Od 04 55 75 69 20 4d 61 6b 65 43 61 6c 6c"; append_body_part("$var(acm)","application/isup;version=itu-t92+","signal; handling=optional"); msg_apply_changes() xlog("L_INFO", "ISUP Changes Applied Succesfully"); } It is converts body to multipart and inserts ISUP But it inserts is as text. Is there any possibility to insert basic ISUP message to update it after using sipt_destination($rU, 31, 4); and sipt_set_calling($fU, 4, 0, 3); Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

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kamailio 4 with asterisk
by Sébastien Brice 15 Feb '16

15 Feb '16

this one (written by Daniel) http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb maybe a bit outdated but still consistent ? the thing i am really stuck with (and concerning real-time) is that none of my extensions (from asterisk CLI) are online: ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] that drives asterisk crazy ! and logger reports: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) every time i place a call. The tutorial written by daniel mention a channel configuration pretty minimal: INSERT INTO sipusers (name, defaultuser, host, sippasswd, fromuser, fromdomain, mailbox) VALUES ('102', '102', 'dynamic', '102', '102', 'yoursip.com', '102'); and since there's no context associated to the 102 extension i cant figure out where that channel enter the dialplan ? [public] [LocalSet] [default] ???? and a dialplan exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,101,Voicemail(${EXTEN},b) exten => _1XX,102,Hangup i am sorry to bother you with issues more asterisk oriented than kamailio. By the way i took a good start with kamailio as it seems to work flawlessly on my system. thx you. On Mon, Feb 15, 2016 at 12:26:06PM +0100, Sébastien Brice wrote: > Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration. Which tutorial? > the only thing i missed is asterisk behaviors'r regarding sip registration ? That was a part of a tutorial I once saw. In essence asterisk uses the kamailio database, UA registers on kamailio and is stored there, asterisk sees the same data (realtime). Sébastien BRICE VoIP, Support et Intégration

2 1

no pkg memory left
by Gholamreza Sabery 15 Feb '16

15 Feb '16

Hello. I am seeing the following error repeating again and again in the log file: /sbin/kamailio[14510]: ERROR: registrar [reply.c:199]: build_contact(): no pkg memory left I am using 64MB of SHM memory and 8MB of PKG memory. What is the reason for this error? How to solve it?

2 2

kamailio 4 with asterisk
by Sébastien Brice 15 Feb '16

15 Feb '16

Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration. the only thing i missed is asterisk behaviors'r regarding sip registration ? I tryed to place a call between 102 and 103 extensions and experimenting an issue Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail ! -- Executing [103@public:1] Dial("SIP/102-00000001", "SIP/103") in new stack [Feb 14 21:00:15] WARNING[19444][C-00000001]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] apart that my sip.conf and extensions.conf are very minimal: [LocalSets] exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,102,Voicemail(${EXTEN},b) exten => _1XX,103,Hangup [general] context=LocalSets ; Default context for incoming calls. Defaults to 'default' rtcachefriends=yes ; Cache realtime friends by adding them to the internal list i did INSERT TO the users in mysql tables (sipusers, sipregs and voicemail) and registering extensions from UA works ok (i am using jitsi) Whats wrong with my setup ? thank you

2 1

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sr-users February 2016