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37519 discussions

Dialogue restore after initial processing
by Joao Alves 17 Jul '15

17 Jul '15

Hi, I'm trying to restore a SIP session dialogue context and found dlg_get(callid, ftag, ttag) function for that. However, I cannot access the To tag. I noticed on the 180 RINGING there was already an self-generated included on this response, but I also could not access its value. When receiving the SIP INVITE request I used the dlg_manage() and the context was initially stored. if(is_known_dlg()) { xlog("Request $rm from $ci is in-dialog\n"); xlog(">>From tag is $ft and To tag is $tt\n"); # Needed to later restore the dialog $shv(ftag)=$ft; $shv(ttag)=$tt; } 15(49100) ERROR: <script>: Request INVITE from 76589ZWQ5NzIyYjMzNDI1YjVjM2E5ZWU5NjhkYzEyNjE3MzQ is in-dialog 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1021 a=25 n=xlog 15(49100) DEBUG: pv [pv_core.c:376]: pv_get_xto_attr(): no Tag parameter 15(49100) ERROR: <script>: >>From tag is 8fd06474 and To tag is <null> Since there was a self-generated one (after the 180 RINGING being sent), I rechecked the $tt and $dlg(to_tag) values and both still returned null values. 15(49100) DEBUG: tm [t_lookup.c:1083]: t_check_msg(): DEBUG: t_check_msg: T already found! 15(49100) DEBUG: <core> [tcp_main.c:2196]: tcpconn_send_put(): send from reader (49100 (15)), reusing fd 15(49100) DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... 15(49100) DEBUG: <core> [tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7effdf7da860 n=396 fd=5 15(49100) DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf= SIP/2.0 180 ringing Via: SIP/2.0/TCP 192.168.1.5:58751;branch=z9hG4bK-524287-1---a64e0a42199ee419;rport=61134;received=85.138.200.8 To: <sip:1001@decasa.website>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f6a3 From: "1000"<sip:1000@decasa.website>;tag=8fd06474 Call-ID: 76589ZWQ5NzIyYjMzNDI1YjVjM2E5ZWU5NjhkYzEyNjE3MzQ CSeq: 1 INVITE Server: kamailio (4.3.0 (x86_64/linux)) Content-Length: 0 ..., shmem=0x7effdf7e1e00: SIP/2.0 180 ringingght(): DEBUG: reply sent out. buf=0x7effec023bf8: SIP/2.0 180 ringing 15(49100) DEBUG: tm [t_reply.c:657]: _reply_light(): DEBUG: _reply_light: finished 15(49100) DEBUG: sl [sl.c:272]: send_reply(): reply in stateful mode (tm) 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1031 a=25 n=xlog 15(49100) DEBUG: dialog [dlg_hash.c:640]: dlg_lookup(): ref dlg 0x7effdf7e1bf0 with 1 -> 2 15(49100) DEBUG: dialog [dlg_hash.c:642]: dlg_lookup(): dialog id=5769 found on entry 128 15(49100) DEBUG: dialog [dlg_hash.c:858]: dlg_unref(): unref dlg 0x7effdf7e1bf0 with 1 -> 1 15(49100) ERROR: <script>: Dialog To Tag: <null> <== xlog("Dialog To Tag: $dlg(to_tag)\n"); 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1033 a=25 n=xlog 15(49100) DEBUG: pv [pv_core.c:376]: pv_get_xto_attr(): no Tag parameter 15(49100) ERROR: <script>: >>From2 tag is 8fd06474 and To tag is <null> Many thanks, Joao Alves Solution Architect, Unified Communications This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

2 3

Configuring Kamailio as SIP Proxy
by Djamel Bahamid 16 Jul '15

16 Jul '15

Hi, I have terminals of video conference which are in public IP ( *DMZ*). I installed kamailio (4.2) which works correctly, it is also in IP pulique (*DMZ*). I wish configured the /kamailio.cfg/ file, so that kamailio works in *proxy SIP*. And declared my CODEC of visoconférence in URI, example: 'DR London(a)domaine.fr'. In what part of the file ( *kamailio.cfg* ) do I have to make these modifications ? This to avoid the flooding. How possibly write the rule of corresponding ACL. Thank you for your help. Best Regards,

3 4

interconnect two Kamailio server
by Kai Ohnacker 16 Jul '15

16 Jul '15

Hello, I have a setup with 2 Kamailio server and their own database. Is it possible to connect these 2 Kamailio server, that the client from the one can make a call to the other? If yes, how could this be realized? Many Thanks. Cheers, Karl --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com

2 1

Error forwarding to SCSCF
by Jan Devai 16 Jul '15

16 Jul '15

*Hi Experts,* *I would like to ask you if some of you have already come across this problem.* *We are integrating Kamailio all in one Solution (P/I/S-CSCF having all same IP only communicating between each other on different ports) taken from Carsten Bock example against an external HSS.* *The flow is as follows:* UE HSS P-CSCF I-CSCF S-CSCF --------------------------Sip Register--------------------------> ----------Sip Register---------> <---------------------------------UAR-------------------------------- ---------------------------------UAA--------------------------------> --------------SIP Register-------> *The last step "SIP Register" from I-CSCF to S-CSCF is not happenning for some reason.* *The Received AVP in UAA is "First Time Registration 2001 code".* *Now we do not know whether this UAA answer is OK, or some other URI is badly defined which could explain why I-CSCF has trouble contacting S-CSCF.If someone have some logs, traces which could be used as a reference for this scenario would be greatly appreciated.* *Find theses messages we get in the shown logs below:* *Thank you* *JanThis file /rtc/kamailio_icscf/kamailio.cfg is a function which sends message to SCSCF, see below:* route[REG_UAR_REPLY] { #xlog("$$avp(s:uaa_return_code) = $avp(s:uaa_return_code)\n"); #this is async so to know status we have to check the reply avp switch ($avp(s:uaa_return_code)){ case 1: #success if (I_scscf_select("0")){ t_on_failure("register_failure"); t_on_reply("register_reply"); #now relay to appropriate SCSCF if (!t_relay()) { t_reply("500", "Error forwarding to SCSCF"); } } else {#select failed I_scscf_drop(); t_reply("500", "Server error on SCSCF Select (UAR)"); } break; case -1: #failure xlog("L_ERR", "UAR failure - error response sent from module\n"); break; case -2: #error xlog("L_ERR", "UAR error - sending error response now\n"); t_reply("500", "UAR failed"); break; default: xlog("L_ERR", "Unknown return code from UAR, value is [$avp(s:uaa_return_code)]\n"); t_reply("500", "Unknown response code from UAR"); break; } } *Kamailio is getting an answer: "t_reply("500", "Error forwarding to SCSCF");"* *which means that th function I_scscf_select("0") is fine but the other function called "t_relay", is the one generating the error "Error forwarding to SCSCF".* *I-CSCF is not able to send the SIP message to S-CSCF, see this log below:* Jul 16 14:59:22 cw-aio /usr/sbin/kamailio[31604]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31599]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: scscf.epc.mnc004.mcc324.3gppnetwork.org Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches

1 0

Purpose of table location_attrs
by mayamatakeshi 16 Jul '15

16 Jul '15

Hello, I need to prepare two kamailio instances accessing the same mysql db. They will write to different location tables so i use function calls like these: Server1: save("k1_location"); lookup("k1_location"); Server2: save("k2_location"); lookup("k2_location"); This is working fine. However, checking db, I can see there is also a table called location_attrs and I am not sure what is its purpose (is this documented?). Anyway, I suspect I should also set a separate copy of it for each kamailio instance, but I am not sure how to do it. Regards, Takeshi

2 2

Error: make modules include_modules ="db_mysql" cfg
by kumar m 16 Jul '15

16 Jul '15

I am a newbie trying out kamailio *kamailio version: *4.3.0 *Operating system:* CentOS 7 Following the instructions provided at http://www.kamailio.org/wiki/install/4.3.x/git While I was trying to include the module "db_mysql", I get the following error [root@localhost kamailio]# make modules include_modules="db_mysql dialplan app_mono" cfg Makefile:56: modules.lst: No such file or directory target architecture <x86_64>, host architecture <x86_64> saving modules list... target architecture <x86_64>, host architecture <x86_64> which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) make --no-print-directory -wC ../.. cfg-defs target architecture <x86_64>, host architecture <x86_64> making config... which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) CC (gcc) [M db_mysql.so] km_dbase.o km_dbase.c:36:25: fatal error: mysql/mysql.h: No such file or directory #include <mysql/mysql.h> ^ compilation terminated. make[1]: *** [km_dbase.o] Error 1 make: *** [modules] Error 1 Please help me in fixing the error

2 1

Unclear documentation for "priority" field in Dispatcher module
by Sven Neuhaus 16 Jul '15

16 Jul '15

Hello, the documentation doesn't clearly state if a larger value for the priority field means that it gets used first before entries with smaller priority values are used - or vice versa. It currently states: “8” - select destination sorted by priority attribute value (serial forking ordered by priority). and priority: sets the priority in destination list (based on it is done the initial ordering inside the set) This is just ambiguous. It should clearly say wether the list is processed in ascending or descending order for the priority value when using algorithm 8 (serial forking aka failover). Regards, -Sven Neuhaus

2 1

Issue handling SRTP and RTP with rtpproxy and rtpengine
by Alberto Sagredo 15 Jul '15

15 Jul '15

Hi Im using an Asterisk cluster behind Kamailio and some services are running on Asterisk. Traffic always must go to Asterisk so RTPPROXY and RTPNGINE do the work and send traffic to Asterisk and traffic from Asterisk go to RTP Proxy I have been able to get rtpproxy working and later tried RTPengie to be able to manage SRTP I have been able to make SRTP To RTP to Asterisk Im able to call between RTP using rtpproxy as recording is a must and rtpengine does not record yet. But im not able to call between SRTP extensions, i understand also SRTP to RTP would work as im doing with Asterisk (Only the speak SRTP as rtpengine trasncode) I have this code on NATMANAGE but i do not know what could be wrong to handle SRTP Any idea? Thanks in advance if(src_ip==ASTERISK1_LAN || src_ip==ASTERISK2_LAN ){ if (sdp_get_line_startswith("$avp(mline)", "m=")) { if ($avp(mline) =~ "AVP") { add_rr_param(";rtp=AVP-kamtoast"); set_rtp_proxy_set("1"); rtpproxy_manage("rwei"); } if ($avp(mline) =~ "SAVP") { add_rr_param(";rtp=SAVP-kamtoast"); rtpengine_manage("trust-address direction=external replace-origin replace-session-connection ICE=remove RTP AVP"); } } } else{ if (sdp_get_line_startswith("$avp(mline)", "m=")) { if ($avp(mline) =~ "AVP") { add_rr_param(";rtp=AVP-asttokam"); set_rtp_proxy_set("1"); rtpproxy_manage("rwie"); } if ($avp(mline) =~ "SAVP") { add_rr_param(";rtp=SAVP-asttokam"); rtpengine_manage("trust-address direction=external replace-origin replace-session-connection ICE=remove "); } } } If you need any more info let me know.

4 12

Missing BYE at the far end
by Kliment Toshkov, Netfinity JSC 15 Jul '15

15 Jul '15

Dear colleagues, We have an issue with a couple of peers - when the called party hangs up, they (the peer) does not receive the appropriate BYE message and call appears as still connected at peers end. Then it drops after 30 seconds and sends us BYE. Current setup is the following: PEER IP > KAMAILIO 4.2.2 > (Asterisk PBX 1, 2, 3, …, N) We provision only termination services, which is: 1) peer calls us 2) Kamailio chooses destination using dispatcher module 3) forwards the call to it. Attached is captured trace. Clearly BYE is retransmitted a few times, then request for UPDATE is received but it is not possible to answer it because the call is already hung up (as far as I understand). Kamailio routing file is available upon request. Thank you in advance! Климент Тошков Изпълнителен директор 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/> t: +359 2 4918888 f: +359 2 4815555 <http://www.netfinity.bg/> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

1 0

ISUP decode and store to be able to append_body_part() later
by andre second 15 Jul '15

15 Jul '15

Hi, I am trying to extract and save ISUP from SIP-I packet for a later use. The problem is that data is stored in binary format and needs decoding before saving to a variable. Can Kamailio perform any decoding so that later I can use append_body_part() to append it back to a multipart message? Thanks,Andrei

1 0

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