sr-users November 2013

sr-users@lists.kamailio.org

100 participants
160 discussions

by arun Jayaprakash

Hello, can someone let me know how I can add a new field in the cdrs table? I would like to set a flag depending if a call is make to another extension number of if the call is made to a PSTN line. Thank you for the help. Arun

10 years, 12 months

Loosing rtp only in carrierroute failureroute

by Oliver Roth

Hi all We do have a strange problem with loosing rtp in case of carrierroute - failureroute. If we send traffic directly to the failure gateway, we do have rtp without any problem Situation: Ua --> Freeswitch --> kamailio 3.3 --> gw 1 Error 404 --> failureroute --> media gateway voiceprompt for error 404 ==> this does not work Ua --> Freeswitch --> kamailio 3.3 rewrite to dest for voiceprompt 404 -->media gateway voiceprompt for error 404 ==> this works!! Any idea? For me it seems being a problem with branching or so. Regards, Oli Configs below: Main routing block request_route { --- do all the checks route(CARRIERROUTE); route(RELAY); } route[RELAY] { xlog(, "L_INFO", "RELAY: Outbound sent via $avp(s:trunk_out)"); # 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"); xlog(,"L_INFO", "RELAX - Manage branch ...\n"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # carrierroute route[CARRIERROUTE] { xlog(,"L_INFO","Carrierroute module start \n"); $avp(s:tree) = "default"; # lookup from user and from domain cr_user_carrier("$fU", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)"); if($avp(s:cr_pref_carr) == 0){ # lookup from domain cr_user_carrier("", "$avp(s:trunk_in)", "$avp(s:cr_pref_carr)"); } if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", "call_id", "$avp(s:todirection)" )){ xlog("cr_route 403 no route found $avp(s:cr_pref_carr)"); sl_send_reply("403", "Not allowed"); exit; } $avp(s:trunk_out) = $avp(s:todirection); route(ALTERHEADER); } failure_route[MANAGE_FAILURE] { if (t_grep_status("486")){ xlog("L_INFO", "Status 486 - busy"); t_reply("486", "Busy"); } #revert_uri(); route(NATMANAGE); if (t_is_canceled()) { exit; } xlog("L_INFO", "failure_route $rd $T_reply_code $avp(s:trunk_in), $avp(s:tree), $rU, $rd, $T_reply_code, $avp(s:tree) \n"); if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){ xlog("cr_next_domain failed"); exit; } if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", "call_id", "$avp(s:todirection)" )){ xlog("cr_route failed"); exit; } $avp(s:trunk_out) = $avp(s:todirection); if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting setflag(FLT_ACCMISSED); # oro 28.10.13 } route(ALTERHEADER); t_on_failure("MANAGE_FAILURE"); if (!t_relay()) { xlog("failureroute t_relay failed"); exit; } }

10 years, 12 months

route all calls to a wav file

by Joli Martinez

Hello, I am getting a lot of hits on port 5060. I would like all registered users to be sent to the correct location (currently working), but all unwanted users allow them to connect and any number they dial it will alway play a wav file. Is this possible if so how would I implement this. Thanks

10 years, 12 months

Use of t_cancel_callid() to drop an early dialog

by Guillaume Bour

Hi All We wan't to prevent our users to make more than one call at time, so we choose to disconnect the previous call. When the previous call is established, we use dlg_bye(), and its ok. But when it is in early state, we use t_cancel_callid() to cancel its INVITE transaction. We face 2 issues: 1) we use local-request event route to account calls on timeout. Sometimes this route is called for the cancelled call (after default timeout of 1 hour) 2) t_cancel_callid() cancel previous call, but also _make current dialog disappear_: call is still ongoing and we can answer and talk to each other, but the dialog does not appear in 'kamctl stats dialog' and 'kamctl mi dlg_list' commands Is there a known limitation, or do we misuse t_cancel_callid() ? Regards, Guillaume

10 years, 12 months

Setting up Conference Calls in Kamailio

by Mark Zeman

Hello everyone, We're thinking of setting up conference call capability for our Kamailio server, but we can't find any tutorials or documentation. Could someone point us in the right direction or maybe even tell us how to do it? Thanks!

10 years, 12 months

Re: [SR-Users] Call hold and call conference implementation in kamailio

by Olle E. Johansson

26 nov 2013 kl. 14:22 skrev "Surendra" <surendra.pullaiah(a)plintron.com>: > Hi, > > Can we implement Call hold and Call conference features in kamailio. If how we can, else what will be the way to implement. > Can anyone help on this. These are media features better handled by end points and media servers. Kamailio by default doesn't handle media, it's a signalling server. Call HOld is between phones, so that's no issue. Call conference can be handled by using Asterisk, FreeSwitch or SylkServer and directing the calls that way in your Kamailio routing script. /O

10 years, 12 months

Calls via Kamailio - Audio delay

by Mark Zeman

Hello everybody, having set up our Kamailio server and test env with Jitsi and CSipSimple successfully, we've noticed a delay of ~5 seconds in the audio. We are using IPv6 and encrypting everything (control and data channel) but this should still not lead to such a big delay, as far as we understand. Does anyone know why there would be this delay? Attached is our config, if that is of any help in fixing this delay. Cheers, Mark Zeman

10 years, 12 months

Call hold and call conference implementation in kamailio

by Surendra

Hi, Can we implement Call hold and Call conference features in kamailio. If how we can, else what will be the way to implement. Can anyone help on this. Thanks in advance @surendra

10 years, 12 months

ACL based on User-Agent

by Igor Potjevlesch

Hello, I was using an old unofficial module called 'check_ua.so' (with an old SER version) to check if an User-Agent is allowed or not. I don't find something similar to do this job in Kamailio. Maybe someone has already something workable with the existing modules? Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

10 years, 12 months

Problem with loose_route for a BYE

by Igor Potjevlesch

Hello, I work on interconnecting an old SER version with Kamailio. When a BYE is sent by the old SER, the "if (loose_route)" instruction returns false and then a "404 Not here" is sent. So the BYE is not relay to the UA. If the BYE is sent by Kamailio, no problem. Here are the BYE headers received by Kamailio: Record-Route: sip:OLD_SER_IP_ADDRESS;ftag=e97b7b77c5a3a9cc;lr=on Via: SIP/2.0/UDP OLD_SER_IP_ADDRESS;branch=z9hG4bKc9e5.3cbad207.0 Via: SIP/2.0/UDP UA_IP_ADDRESS_CONNECTED_TO_OLD_SER:5060;rport=5060;branch=z9hG4bK1fbe0a49fe3 44e6af1efd45b74305a9a Route: sip:KAMAILIO_IP_ADDRESS;lr=on I looked at the RR module documentation and it's clearly said that "The loose_route function analyzes the Route: headers in the requests. [.]If a Route: header is found, the function returns 1 and behaves as described in section 16.12 of RFC 3261.". Someone has an idea? Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

10 years, 12 months

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sr-users November 2013