- sr-users - kamailio.org

re-connect call in case of network issue
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 how can we implement re-connecting call instead of disconnect in case of network problem plz help thanks -- *Regards:* Gaurav Kumar

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VRF routing with kamailio 5.2
by Jonathan Hunter 18 Apr '19

18 Apr '19

Hi All, Hope everyone in community is well. I was wondering if anyone could let me know if VRF is supported with kamailio 5.2 ? Essentially I am looking to put kamailio between two SBCs and bridge the networks together. So currently Server A 1.1.1.1 connects to Server B 2.2.2.2 and they pass calls fine, and I want to add kamailio in between without changing any IPs, so using VRF I have one network interface intA with 1.1.1.1 and the other intB with 2.2.2.2. What I want to do is listen on both interfaces, which is fine but bridge or force traffic between the two. Is this possible with kamailio? As doing force socket and listening on both 1.1.1.1 and 2.2.2.2 kamailio just loops and routes to itself, and force socket doesnt help. Do you have any suggestions? Would running two instances of kamailio on same server help or even introducing to separate kamailio instances and force routing between the two for requests. I am currently just testing with SIP OPTIONS and it is already proving tricky. Any help or comments appreciated. I appreciate this might not be the best solution however we have a lead time with providers to make changes which are proving quite slow and impractical so just trying to work a solution. Thanks Jon

2 1

integration kamailio with Freeswitch for video conferencing
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 and freeswitch 1.8 on debian 9 i want to configure kamailio with freeswitch for video conformance plz provide any reference guide plz help Thanks -- *Regards:* Gaurav Kumar

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video chat with kamailio
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i m working on video chat app so how can i use kamailio for video chat OR i need to use other integration like freeSwitch with kamailio ? Plz help -- *Regards:* Gaurav Kumar

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Support for TLS v1.3 (RFC 8446)
by g.Helio 18 Apr '19

18 Apr '19

2 1

Via shows wrong protocol of Kamailio
by Andrew Chen 18 Apr '19

18 Apr '19

Hi all, Got another anomaly in 4.1.5. I have this block of code which does t_relay_tcp: if ($var(isTpgw) == 1) { xlog("L_INFO","[$ci][$mi][CSeq $cs] isTpgw = $var(isTpgw)"); if (!t_relay_to_tcp()) { xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp....Error...\n"); t_on_failure("DISPATCHER_FAIL"); } else { xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp\n"); } } But when I looked at my sip capture tool, it showed the Via marked by the Kam to use UDP: INVITE sip:######@<some domain>:5060 SIP/2.0 Record-Route: <sip:206.81.191.30;r2=on;lr=on> Record-Route: <sip:206.81.191.30;transport=tcp;r2=on;lr=on> Call-ID: DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM To: "sip:6204857@sjomaintpsg50.fuzemeeting.com" <sip:6204857@sjomaintpsg50.fuzemeeting.com> From: "LifesizeSIP" <sip:LifesizeSIP@gs-3924.GSLAB.COM:5060>;tag=DL325a9e10c7;epid=062EE468 CSeq: 1 INVITE Max-Forwards: 69 Via: SIP/2.0/UDP 206.81.191.30;branch=z9hG4bKb16b.bd6a0791c798e3c859f8608993d584ae.1;i=5611 Via: SIP/2.0/TCP 10.101.199.105:5060;rport=53272;branch=z9hG4bK-fc8090ecf8-DL 206.81.191.30 is the Kam. I also log the activity as seen in the code with those xlog: Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM][42][CSeq 1] isTpgw = 1 Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM][42][CSeq 1] LifeSize Softphone 8.1.12 (Windows) Relaying [INVITE] from (10.101.19 9.105) to (206.81.191.72) via tcp So why did the Kamailio mark the Via header on the outgoing INVITE UDP? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 1

"481 Dialog/Transaction Does Not Exist" after sending SUBSCRIBE with Expires: 0
by David Dean 18 Apr '19

18 Apr '19

When a client sends a SUBSCRIBE with Expires: 0, Kamailio responds correctly with a "202 OK". But then Kamailio sends a NOTIFY to the same client with "Subscription-State: terminated;reason=timeout". The client rejects this NOTIFY with a "481 Dialog/Transaction Does Not Exist". Why does Kamailio send this NOTIFY, and is there any way to fix or disable it because it's always rejected by the client? The client is IMSDroid, and Kamailio is 5.2.2 with a default configuration. Packet 1 = Client sending SUBSCRIBE with Expires: 0 SUBSCRIBE sip:X.X.X.X:443;transport=tcp SIP/2.0Via: SIP/2.0/TCP 192.168.0.21:45336;branch=z9hG4bK-670576127From: <sip:username@domain.com>;tag=1372208921To: <sip:kamailio.domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6Contact: <sip:username@192.168.0.21:45336;transport=tcp>Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 1321654295 SUBSCRIBEExpires: 0Content-Length: 299Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERAccept: application/pidf+xmlEvent: presence Packet 2 = Kamailio responding with 202 OK SIP/2.0 202 OKVia: SIP/2.0/TCP 192.168.0.21:45336;branch=z9hG4bK-670576127From: <sip:username@domain.com>;tag=1372208921To: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 1321654295 SUBSCRIBEExpires: 0Contact: <sip:X.X.X.X:443;transport=tcp>Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 0 Packet 3 = Kamailio sending a NOTIFY NOTIFY sip:username@192.168.0.21:45336;transport=tcp SIP/2.0Via: SIP/2.0/TCP X.X.X.X:443;branch=z9hG4bKefe7.5f8ec435000000000000000000000000.0To: <sip:username@domain.com>;tag=1372208921From: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6CSeq: 3 NOTIFYCall-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0Content-Length: 0Max-Forwards: 70Event: presenceContact: <sip:X.X.X.X:443;transport=tcp>Subscription-State: terminated;reason=timeout Packet 4 = Client responding to the NOTIFY with "481 Dialog/Transaction Does Not Exist" SIP/2.0 481 Dialog/Transaction Does Not ExistVia: SIP/2.0/TCP X.X.X.X:443;branch=z9hG4bKefe7.5f8ec435000000000000000000000000.0From: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6To: <sip:username@domain.com>;tag=1372208921Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 3 NOTIFYContent-Length: 0 There are two entries in the Kamailio log: INFO: {1 1321654295 SUBSCRIBE e8cffa8f-948d-eb99-69ca-c9f18bdf11a0} presence [notify.c:1614]: send_notify_request(): NOTIFY sip:username@domain.com via sip:Y.Y.Y.Y:45336;transport=tcp on behalf of sip:kamailio.domain.com for event presence : e8cffa8f-948d-eb99-69ca-c9f18bdf11a0 ERROR: {2 3 NOTIFY e8cffa8f-948d-eb99-69ca-c9f18bdf11a0} presence [subscribe.c:497]: delete_subs(): Failed to delete subscription from memory [slot: 174 ev: sip:kamailio.domain.com pu: presence ci: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0 ft: 1372208921 tt: c4eb72dabf44e7588b972bd3b454ea11-0cd6] These are the presence settings in the Kamailio config file: #!ifdef WITH_PRESENCE# ----- presence params -----modparam("presence", "db_url", DBURL)modparam("presence", "send_fast_notify", 0) # ----- presence_xml params -----modparam("presence_xml", "db_url", DBURL)modparam("presence_xml", "force_active", 1)modparam("presence_xml", "force_dummy_presence", 0)#!endif Appreciate your help.

2 1

Rewrite private IP in SDP
by Bonjour Madame 18 Apr '19

18 Apr '19

Hello all, I am dealing with a no audio issue, phones are not even ringing. There is a private IP address in the "VIA" section of the message header of the INVITE: Via: SIP/2.0/UDP 192.168.1.25:65291 ;rport;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34 and in the contact too: Contact: <sip:5002@192.168.1.25:65291;ob> In the "status 100 Attempting to connect your call" from the server, I can see the public IP of the phone under received: Via: SIP/2.0/UDP 192.168.1.25:65291 ;rport=53990;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34;received=PHONE_PUBLIC_IP And in the "status 200 ok" from the server, I have: Via: SIP/2.0/UDP 192.168.1.25:65291 ;received=PHONE_PUBLIC_IP;rport=53990;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34 Record-Route: <sip:SERVER_PUBLIC_IP;lr=on;ftag=9284528256a54f8b81e692133a15684c> I don't know what steps to take to resolve this issue.

3 3

Record route dynamic IP
by eyas barhouk 17 Apr '19

17 Apr '19

hello dears i'm trying to build kamailio & asterisk with the following : * Kamailio will listen to All the interfaces : listen=udp:0.0.0.0:5050 listen=udp:0.0.0.0:5060 * it will add the record header as in the following part : if (!isflagset(FLT_RR)) record_route(); setflag(FLT_RR); when the sip message get from kamailio to asterisk the record route header will be like "<sip:0.0.0.0:5060;....etc.>" So is there any way to make the Via and record route headers has a specific IP instead of (0.0.0.0) i mean Something like "record_route("$Ri")". thanks & regards

2 1

How can i change kamailio SIP port form 5060 to any other
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i want to change kamailio SIP port 5060 to other port for security purpose plz help me -- *Regards:* Gaurav Kumar

2 3

TCP connection lifetime
by Aymeric Moizard 17 Apr '19

17 Apr '19

Hi Again, Here is an issue with TCP connection being kept for more: Yesterday, I have discovered that a User-Agent (<Avaya IP Phone 1120E (SIP1120e.04.04.30.00)> tried to register a lot. It was sending REGISTER over new established TCP socket *every 2 seconds*. All the REGISTER was rejected with 401. (may be the device was misconfigured? or not receiving any of my answer? I can't tell) NOTE: You can see the expires header was very large: 86400, ie: 24 hours... I was checking the TCP/TLS connections on my server and discovered more than 1000 TCP established connection to that user/ip, and thus, I have tried to understand what happened. Checking the logs, I received 4855 REGISTER from this device from "Mar 25 03:47:09" to "Mar 25 07:56:13" which is a rate of approx one new TCP connection every 2.5 seconds... Today, I decided to check it again around 11am. jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1857", "tcp:con_timeout = 35927", "tcp:connect_failed = 25", "tcp:connect_success = 2", "tcp:current_opened_connections = 2291", "tcp:current_write_queue_size = 0", "tcp:established = 80778", "tcp:local_reject = 0", "tcp:passive_open = 80776", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 7305 } There was still A LOT of established connections. And the connections have been established more than 24 hours ago. At 11H16: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1161 At 11H22: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1018 At 11H35: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 655 At 13H $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 0 So the established connections are all gone now. Between 11h16 and 11H35, I was seeing the server regularly sending [FIN, ACK] over each TCP established connection, with retransmissions for all of them. (no incoming trafic) I do not have numbers/capture/stats, but I think that kamailio was already closing some connection yesterday. I don't know when kamailio started to try closing those connections. I'm now back with this status: At 13pm: jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1896", "tcp:con_timeout = 38042", "tcp:connect_failed = 26", "tcp:connect_success = 2", "tcp:current_opened_connections = 939", "tcp:current_write_queue_size = 0", "tcp:established = 81950", "tcp:local_reject = 0", "tcp:passive_open = 81948", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 12734 } With around 155 registration entries using TCP and TLS in my location database. As you can see, tcp:current_opened_connections = 939 is still pretty high compared to my currently registred users. I have "modparam("registrar", "max_expires", 86400)", because I'm keeping contact entries (even with TCP connection down) for push notifications. I have "tcp_connection_lifetime=3600" configured. Question 1 With "tcp_connection_lifetime=3600", I would expect kamailio to close the established connection after 3600 seconds without traffic. It is pretty obvious that no data has been exchanged over the 4855 established connection during a day. Despite the issue with the Avaya phones is solved automatically after a day, I guess similar stuff or happening, at a different rate, for other users as well. (because current_opened_connections is way higher than registred TCP/TLS users) Question 2 I can list TLS connection with "kamctl rpc tls.list" Can I get a similar list for TCP? (lsof returns a lot of duplicates...) Tks Aymeric -- Antisip - http://www.antisip.com

2 4

Route header
by Duarte Rocha 17 Apr '19

17 Apr '19

Greetings, Is there a difference for Kamailio between these 2 situations. 1 - Method with 2 Route Headers, each one with one address and one uri 2 - Method with 1 Route Header that includes 2 Route URI Best Regards, Duarte Rocha

1 0

allow calls only from registered users
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i wan to implement user based calling on kamailio , that only can call how registered with user plz help -- *Regards:* Gaurav Kumar

2 1

Send 200 OK response with body to INVITE requests
by David Dean 16 Apr '19

16 Apr '19

Hi all, I'm building an automated load testing system for an rtp media server with TURN that uses Kamailio for SIP. When a client sends an INVITE to Kamailio, I need Kamailio to respond with "200 OK" and the SDP generated by rtpengine. I've enabled auto-answer in the client, so as soon as it receives the response from Kamailio it will start sending RTP packets. Right now I'm using send_reply("200", "OK"); which is sending the right 200 response back to the client, but it doesn't include any body. How can I get Kamailio to send a "200 OK" and add the rtpengine_answer into the body? Appreciate any help, I've been struggling with this for three days. Cheers. rtpengine_offer("SIP-source-address ICE=force-relay RTP");send_reply("200", "OK");

4 7

Getting a new RTP Port on reINVITE during answered call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure (fake IPs) 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* 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:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.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 "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- 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) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- 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 we 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 */ 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") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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 route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

new RTP Port after INVITE during call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* 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:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.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 "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- 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) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- 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 we 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 */ 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") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN //route(PSTN); # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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 route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN xlog("L_INFO","Entering PSTN"); # route to PSTN dialed numbers starting with '+' or '0000' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) { xlog("L_INFO","$rU is not outbound call to PSTN"); return; } if (!starts_with("$si","10.19.139")) { xlog("L_INFO","$rU is not comming from any Media GW"); return; } # only local users allowed to call if(from_uri!=myself && !ds_is_from_list("1") ) { xlog("L_INFO", "from_uri $fu != myself, so this is Not Allowed!!"); sl_send_reply("403", "Not Allowed"); exit; } xlog("L_INFO","$si request"); xlog("L_INFO","$rU is an outbound call to PSTN"); $ru = "sip:" + $rU + "@62.28.173.10"; // + $sel(cfg_get.pstn.gw_ip); route(RELAY); /* if(!ds_select_dst("20", "4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); */ exit; #!endif xlog("L_INFO"," NO PSTN SET!!"); return; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Dispatch requests route[DISPATCH_TO_PSTN] { # round robin dispatching on gateways group '20' - PSTN if(!ds_select_dst("20", "10")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

single Contact per AOR dilema
by Sergiu Pojoga 16 Apr '19

16 Apr '19

Hi ppl, Say we want to allow 1 contact per AOR. modparam("registrar", "max_contacts", 1) Say some UA looses connection before de-registering, then comes back shortly after (before Contact expires) and tries to register with a different Contact, but gets rejected with error "test_max_contacts(): too many contacts for AOR". What's the approach to handle this situation? I'd like the registrar to accept and save that AoR based on the latest registration. Similar to how Asterisk's single-contact stack `chan_sip` works - whoever was the latest to register successfully gets to be the winner. Thanks, --Sergiu

2 2

dlg_onroute(): failed to update dialog lifetime
by Olli Attila 15 Apr '19

15 Apr '19

Hello, I just activated snmpstats module on Kamailio 5.0.7 and added the following line to route(INITIAL) at the bottom of the route like this (For noc group to monitor Kamailio dialogs): # Per SIP request initial checks route[INITIAL] { ... dlg_manage(); } Now I started to gets this error few times in an hour: /usr/sbin/kamailio[22194]: CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f798ca51b98 tl->next=(nil) tl->prev=(nil) Apr 15 09:54:05 /usr/sbin/kamailio[22194]: ERROR: dialog [dlg_handlers.c:1439]: dlg_onroute(): failed to update dialog lifetime I see no call failures because of this but still interested what is happening here? --Olli Attila

1 0

one way Audio issue
by Gaurav Bmotra 12 Apr '19

12 Apr '19

i have installed kamailio 5.1.x on Ubuntu server 18.4 issue is i am able to connect the call but there is one way audio not both way as i have install rtpproxy also with following config /etc/default/rtpproxy EXTRA_OPTS="-F -s udp:127.0.0.1:7722 -l PUBLIC_IP -d DBUG:LOG_LOCAL0" /etc/kamailio/kamailio.cnf modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") after restart kamailio i got this logs dev /usr/sbin/kamailio[6926]: INFO: rtpproxy [rtpproxy.c:1559]: rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled during calling i got this logs Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): dr Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error whi Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db_request(): failed to insert in Apr 12 16:05:43 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13475]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction ans Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): drive Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db Thanks Gaurav *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] [sr-dev] Audio missing during call
by Mikko Lehto 12 Apr '19

12 Apr '19

Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > HI i have install Kamailio 5.1.2 on Ubuntu 18.4 > there is a audio issue , when we call call from one extension to another ,, > in bigning call goes good , after 9-10 sec it miss audio for one side (one > side can hear other side can't hear ) Hi, mailing list sr-dev is for Kamailio development related topics, please subscribe to sr-users for more general queries. Before posting, make sure you have described in detail your environment and also troubleshooting steps you have done so far: - what modifications you made to default configuration - what happens during SIP transactions in fail scenario - are you using RTP relay or not (rtpproxy/rtpengine) - what have you observed from packet captures Good luck -- Mikko

2 1

command out of sync with db_cluster and db_mysql
by Simone Lazzaris 12 Apr '19

12 Apr '19

Hi everybody; I'm trying to use db_cluster to connect my kamailio installation to a mariadb galera cluster, and it's driving me crazy. In my lab it works without issues, but when I try the setup in the production node, I get some strange errors: Apr 11 14:21:31 ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): driver error on query: Commands out of sync; you can't run this command now (2014) Apr 11 14:21:31 ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query This is my configuration: modparam("db_cluster", "connection","con1=>mysql://openser:PASSWD@X.X.X.30/ openser") modparam("db_cluster", "connection","con2=>mysql://openser:PASSWD@X.X.X.31/ openser") modparam("db_cluster", "connection","con3=>mysql://openser:PASSWD@X.X.X.32/ openser") modparam("db_cluster", "cluster", "cls1=>con1=9r8r;con2=9r8r;con3=9r8r") #!define DBURL "cluster://cls1" #!define SQLOPSURL "database=>cluster://cls1" modparam("dispatcher", "db_url", DBURL) modparam("acc", "db_url", DBURL) modparam("usrloc", "db_url", DBURL) modparam("auth_db", "db_url", DBURL) modparam("permissions", "db_url", DBURL) modparam("alias_db", "db_url", DBURL) modparam( "sqlops", "sqlcon", SQLOPSURL ) I'm using version 5.1.4 of kamailio, freshly compiled from source on a debian 9. Any ideas? *Simone Lazzaris* *Qcom S.p.A.* simone.lazzaris(a)qcom.it[1] | www.qcom.it[2] * LinkedIn[3]* | *Facebook[4]* [5] -------- [1] mailto:simone.lazzaris@qcom.it [2] https://www.qcom.it [3] https://www.linkedin.com/company/qcom-spa [4] http://www.facebook.com/qcomspa [5] https://www.qcom.it/includes/email-banner.gif

2 2

kamailio, asterisk and BLF seems its only using json?
by PICCORO McKAY Lenz 12 Apr '19

12 Apr '19

hi, i try to follow the how to in the wiki https://kb.asipto.com/kamailio:presence:k43-blf and configure my two only phones.. but for any states only one led its blinking, the busy state.. no other state work, (if i connect directly to asterisk all work perfeclty, i using asterisk 13 with kamailio 4.4.4) also read the archived https://lists.kamailio.org/pipermail/sr-users/2013-May/078234.html but no luck wht happened with that how to? it's different for asterisk if are integrated? do i need a pjsip asterisk event the oldes chan? see that in presense module parameters, the server address said that value of Contact header filed for 200 OK replies to SUBSCRIBE and PUBLISH and in NOTIFY messages umm if i have integrated and working asterisk (relatime mode) with kamailio, that address muswt be asterisk? taking in cosideration that i have a route rule said that send auth to asterisk and trust any incomming from it? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

1 0

having issue updating $td
by Andrew Chen 11 Apr '19

11 Apr '19

Hey guys, Question. I have this block of code at the beginning of route block: if(is_ip($td)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] IP detected in To domain. Resolving. \n"); exec_avp("host $td | awk '{print substr($$5, 1, length($$5)-1)}'", "$avp(s:newtd)"); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new newtd = $avp(s:newtd)"); $td = $avp(s:newtd); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new td = $td"); } but $td never updates. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] IP detected in To domain. Resolving. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new newtd = sjomaintpsg50.fuzemeeting.com Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new td = 206.81.191.30 Is there a reason why? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 3

Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Releasing Kamailio v5.1.8
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, I am considering to release v5.1.8, an minor version to be built from branch 5.1, on Thursday, this week (April 11, 2019). As usual, check that at least the patches for issues you reported were backported in order to be sure they were not missed during the typical backporting done by developers so far. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 1

Using AWS boto3 with KEMI Python routing scripts
by Michael Loughrey 10 Apr '19

10 Apr '19

Hi All, looking for some advice regarding the proper way to initialise an AWS API boto3 client object for send SNS messages within a KEMI Python 2.7 routing script. I'm sure process forking has a major impact on how this works - from kamailio.cfg fork=yes children=4 - but would greatly appreciate some guidance. I have tried various methods to allocate an client object self.sns_client = boto3.client('sns', region_name=MY_AWS_REGION) 1. within __init__(), only once from module initialisation 2. on demand within a function call by any client code e.g. called from ksr_route_request(), each creating it's own client object and so far all have resulted in intermittent crashes within botocore/client.py ultimately crashing thus : Core was generated by `/usr/local/sbin/kamailio -P /usr/local/kamailio/run/kamailio.pid -f /usr/local/'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 (gdb) bt #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #1 0x00007f8f9734f82e in X509_VERIFY_PARAM_free () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #2 0x00007f8f97642f5c in SSL_free () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #3 0x00007f8f86b09c64 in PySSL_dealloc.lto_priv.5 () at ./Modules/_ssl.c:1598 #4 0x00007f8f99cbb007 in insertdict_by_entry (mp=0x7f8f841f1e88, key='_sslobj', hash=<optimized out>, ep=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:519 #5 0x00007f8f99cbe2cf in insertdict (value=None, hash=1051385741686792393, key='_sslobj', mp=0x7f8f841f1e88) at ../Objects/dictobject.c:556 #6 dict_set_item_by_hash_or_entry (value=None, ep=0x0, hash=1051385741686792393, key='_sslobj', op={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/dictobject.c:795 #7 PyDict_SetItem (op=<optimized out>, key=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:848 #8 0x00007f8f99becec1 in _PyObject_GenericSetAttrWithDict (obj=<optimized out>, name='_sslobj', value=None, dict={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/object.c:1529 #9 0x00007f8f99bed437 in PyObject_SetAttr ( v=<SSLSocket(server_hostname=u'sns.us-east-1.amazonaws.com', _connected=True, _context=<SSLContext at remote 0x7f8f84232398>, server_side=False, _makefile_refs=0, _closed=False, _sslobj=None, do_handshake_on_connect=True, suppress_ragged_eofs=True) at remote 0x7f8f841f62a8>, name=<optimized out>, value=None) at ../Objects/object.c:1247 Here's my setup host : Debian9/stretch python --version Python 2.7.13 kamailio -v version: kamailio 5.2.2 (x86_64/linux) 67f967-dirty flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, MEM_JOIN_FREE ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 67f967 -dirty compiled on 23:44:31 Apr 8 2019 with gcc 6.3.0 Cheers Mike

2 3

Rtpengine - no RTP packets going out
by Istvan Mogyorosi 10 Apr '19

10 Apr '19

Dear all, This is my first post after reading a lot in this mailing-list. I'm trying to use Kamailio 5.1 with the dispatcher module and rtpengine acting as SIP + RTP proxy. I have 6 asterisk servers in a private subnet that should talk with the peer via a single IP like this: Asterisk 1..n|---> | GW.PRIVATE.IP -o- GW.PUBLIC.IP |----> PEER.SIP.TRUNK I'm on Centos 7, with firewalld configured, iptables module is loaded and the rule is well defined. Packet forwarding is also enabled. Chain rtpengine (1 references) target prot opt source destination RTPENGINE udp -- anywhere anywhere RTPENGINE id:40 My call flow seems to be fine, Kamailio/rtpengine private IP is the outboundproxy parameter of Asterisk instances. My problem is that RTP packets are not present on the public interface, the rtpengine final log showing the 2 sessions, but I'm not sure this is what I want or simply the firewall does not let it out ? (To be more precise PEER.SIP.TRUNK is the trunk for SIP traffic, I have multiple IP addresses for media to connect to, reinvites are allowed) Closing call due to timeout Final packet stats: --- Tag 'as6d12caea', created 1:00 ago for branch '', in dialogue with 'as541b1e61' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PRIVATE.IP:10000 <> 192.168.30.13:11152, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PRIVATE.IP:10001 <> 192.168.30.13:11153 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts --- Tag 'as541b1e61', created 1:00 ago for branch '', in dialogue with 'as6d12caea' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PUBLIC.IP:10000 <> PEER.SIP.TRUNK:28216, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PUBLIC.IP:10001 <> PEER.SIP.TRUNK:28217 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts Best regards, Istvan

2 2

Kamailio and CDR
by Yu Boot 10 Apr '19

10 Apr '19

Is there any other way to retrieve CDRs from Kami other than dialog/acc modules and writing CDRs to syslog only? Thanks.

6 13

Help request in regards to Kamailio
by Anas ZAHHAF 10 Apr '19

10 Apr '19

Dear, I am writing to ask for help, I found your Email after I've subscribed to Kamailio.org. I am a beginner in Linux environment ( I worked on Windows for 20 years), it would be very helpful if you can help me installing Kamailio on a computer running Debian and how to use the load balancing module so that I can split the charge between 2 asterisk servers, by sharing any useful tutorial or document or even by Teamviewer if this doesn't disturb you. Please help me, all the tutorials I find on internet aren't made for beginners. I will be awaiting your reply and excuse me again if I disturb you. Best regards, -- *Anas ZAHHAF* *105, rue des Moines75017 PARIS* *+33(0)7.66.31.31.85* azahhaf(a)zaion.ai <niluse(a)zaion.ai> | www.zaion.ai ᐧ

2 1

change failure status if reason is q.850 cause=31
by Techinnovation 10 Apr '19

10 Apr '19

Hello, How can I change the status where Reason: q.850;cause=31. ? This code change status for all where we receive status 480 to 404. Is it possible to this code also to add second condition - if reason is like "q.850;cause=31" failure_route[MANAGE_FAILURE] { ... if (t_check_status("480")) { t_reply("404","Unallocated number"); exit; } Need to modify this massage. SIP/2.0 480 Temporarily Not Available. Allow: UPDATE,REFER,INFO. Call-ID: 6b11b153405a5103330697b774b97f5a@192.168.2.2:5060. Contact: <sip:192.168.3.3:5060>. CSeq: 102 INVITE. From: "380668375820" <sip:380668375820@192.168.2.2>;tag=as445242c0. Reason: q.850;cause=31. Server: Cirpack/v4.76 (gw_sip). To: <sip:9052946032@192.168.3.3:5060>;tag=00-07929-6e4a3079-75447e2a5. Content-Length: 0. Thanks. BR, Aleksandr

2 1

Question regarding retcode
by Andrew Chen 09 Apr '19

09 Apr '19

Hey all, Question regarding the type of retcode. I ran a curl command and $retcode returned 200. I check against that variable if it's 200 or not. I tried to check 200 as a string and integer but never matches so I ran these lines of code: xlog("L_INFO", "[$ci][$mi][CSeq $cs] Validating meeting ($rU). Result: $var(result) Return code: $retcode \n"); if(is_int($retcode)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is an integer \n"); } if(typeof("$retcode", "str")) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is a string \n"); } None of these xlog outputed. How do I check for $retcode in if or switch statement? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Centos 7 opensuse repo issues for 5.2
by Jeffrey Gray 09 Apr '19

09 Apr '19

Good afternoon, We are unable to get kamailio 5.2 to install using the recommended opensuse repos for Centos 7. Is there any way to resolve the following errors: Thanks very much for your assistance! using the following repo setup in /etc/yum.repos.d/ [home_kamailio_v5.2.x-rpms] name=RPM Packages for Kamailio v5.2.x (CentOS_7) type=rpm-md baseurl=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rp… gpgcheck=1 gpgkey=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpm… enabled=1 Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: kamailio-5.2.2-3.el7.centos.x8 FAILED ================- ] 2.3 MB/s | 1.7 MB 00:00:01 ETA http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.2.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try. If we run the suggested clean metadata command, then the error changes to: sudo yum -y install kamailio Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirrors.advancedhosters.com * epel: mirror.cogentco.com * extras: mirrors.advancedhosters.com * updates: mirrors.advancedhosters.com Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-5.2.2-3.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try.

3 3

Upstream ignores reply, because To-tag doesn't match?
by Sergiu Pojoga 09 Apr '19

09 Apr '19

Hi ppl, Scenario: invite from upstream is t_relayed to a client gateway. After `fr_inv_timeout` occurs, in a failure route I simply t_reply with "503 - Service unavailable", then exit(). However, the upstream provider simply ignores this reply, call doesn't hang up. This doesn't happen when the client's gateway generates identical 503 or other negative replies. I suspect this is happening because the To-tag in the Kam generated 503 reply doesn't match with the To-tag that previously was forwarded from the client's gateway to the upstream provider in the `180 Ringing`. Any suggestions what can be done about it? 2019/04/06 10:42:46.127177 65.39.xxx.xxx:5060 -> 208.72.xxx.xxx:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 208.72.xxx.xxx:5060;received=208.72.xxx.xxx;rport=5060;branch=z9hG4bK1982422573 Record-Route: <sip:65.39.xxx.xxx;lr=on;did=915.8af1> From: "SERGIU" <sip:514XXXXXXX@208.72.xxx.xxx>;isup-oli=00;tag=1975755942 To: <sip:514XXXXXXX@65.39.xxx.xxx>;tag=as2ab54180 Call-ID: DID-28270826(a)208.72.xxx.xx CSeq: 477023 INVITE Supported: replaces, timer, path Contact: <sip:1514XXXXXXX@65.39.xxx.xxx:5060> Content-Length: 0 2019/04/06 10:42:51.072516 65.39.xxx.xxx:5060 -> 208.72.xxx.xxx:5060 SIP/2.0 503 Service Unavailable Call-ID: DID-28270826(a)208.72.xxx.xx Via: SIP/2.0/UDP 208.72.xxx.xxx:5060;rport=5060;branch=z9hG4bK1982422573;received=208.72.xxx.xxx From: "SERGIU" <sip:514XXXXXXX@208.72.xxx.xxx>;isup-oli=00;tag=1975755942 To: <sip:514XXXXXXX@65.39.xxx.xxx>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8e82 CSeq: 477023 INVITE Server: KAM Content-Length: 0 Thanks, --Sergiu

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modifying via header from public IP to private IP
by Narasimhaiah, Vinod Maralenahalli 09 Apr '19

09 Apr '19

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. The calls are landing on another sip gateway, they will allow only the private IP. Please let me know, is it possible to replace. I tried all the replace functions but it is not working. Thank you vinod.M.N -- ***This transmission contains information that is confidential and proprietary of Kirusa Inc.,its affiliates and/or its partners and customers. This information is intended to be shared only with the recipient of this e-mail who is bound by confidentiality. If you received this e-mail in error, please delete the e-mail and notify the sender by return e-mail.***

1 0

modifying via header from public IP to private IP
by vinod mn 08 Apr '19

08 Apr '19

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. The calls are landing on another sip gateway, they will allow only the private IP. Please let me know, is it possible to replace. I tried all the replace functions but it is not working. -- Thanks and regards Vinod.M.N

3 2

How to Register sip trunk in kamailio without username and password?
by Mostafa Moradi 08 Apr '19

08 Apr '19

Hi, I'm mostafa moradi from iran, I Have installed kamailio, and I have a sip trunk without username and password, I want to register the sip trunk in kamailio with UAC Module, but this module only accepts sip trunk with user and password, How Can I Register sip trunk in kamailio without username and password? Thank you in advance Best regards,

4 3

Re: [SR-Users] Timeout between 100 & 183
by Alex Balashov 05 Apr '19

05 Apr '19

On Fri, Apr 05, 2019 at 06:23:16PM -0400, Alex Balashov wrote: > route { > ... > > # Initial INVITE handling: > > if(is_method("INVITE")) { > # Set fr_inv_timer initially to 5 seconds, so that > # if we do not receive an answer or at least one non-100 1xx > # response, trigger a failure_route and try the next gateway > # or whatever. > > t_set_fr(50000); Sorry, this would have been a lot more pedagogically useful if it had read: t_set_fr(5000); -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

UAC authentication challenge response using KEMI/app_ruby
by Andrew White 05 Apr '19

05 Apr '19

Hi all, I’m continuing on my quest to use app_ruby for my logic - it’s working very well so far! The last major hurdle I’ve come across is UAC response to WWW challenge. On my outbound path, the correct URIs for From and To are set based on provider requirements, and the INVITE is sent using t_relay(): def ksr_route_to_trunk() KSR.info("Route to trunk") if KSR::TM.t_is_set("failure_route") < 0 then KSR::TM.t_on_failure("ksr_failure_manage_trunk") end # Relay KSR::PV.sets("$fu", "sip:#{$uacs[trunk]['r_username']}@#{$uacs[trunk]['r_domain']}") KSR::PV.sets("$fn", '') new_uri = "sip:61400123123@#{uacs[trunk]['r_domain']}" KSR.info("New URI is #{new_uri}") KSR.seturi(uri) KSR::UAC.uac_replace_to('', uri) KSR::TM.t_relay() exit end I’ve stripped it back, but this function is great and works perfectly. An INVITE is sent to the correct place, and the trunk receives the INVITE. It then responds with a 401/WWW-Authenticate. My failure route is called fine. Below is the stripped down code: def ksr_failure_manage_trunk() exit if KSR::TM.t_is_canceled() > 0 KSR.info("Hit trunk failure manage route") KSR::PV.sets("$avp(auser)", 'user') KSR::PV.sets("$avp(apass)", 'pass') KSR::PV.sets("$avp(arealm)", 'test.local') KSR.info("AVPs: #{KSR::PV.gete("$avp(auser)")} #{KSR::PV.gete("$avp(apass)")} #{KSR::PV.gete("$avp(arealm)")}") if KSR::UAC.uac_auth() then KSR.info("UAC authed, relaying") KSR::TM.t_relay() else KSR.info("UAC NOT authed, no relay") end exit end Also the AVP values are set (I believe) correctly: modparam("uac", "auth_username_avp", "$avp(auser)") modparam("uac", "auth_password_avp", "$avp(apass)") modparam("uac", "auth_realm_avp", "$avp(arealm)") This throws the following complaint, and doesn’t send a new response back to the trunk: Apr 5 02:55:22 ip-10-0-0-2 /usr/local/sbin/kamailio[26870]: ERROR: {1 102 INVITE 3a84da15272da66e527a79144c48516c@10.0.0.20:5060} tm [t_fwd.c:1728]: t_forward_nonack(): no branches for forwarding Apr 5 02:55:22 ip-10-0-0-2 /usr/local/sbin/kamailio[26870]: ERROR: {1 102 INVITE 3a84da15272da66e527a79144c48516c@10.0.0.20:5060} tm [tm.c:1619]: _w_t_relay_to(): t_forward_noack failed I don’t understand branches very well, however after some digging around, I appended KSR::COREX.append_branch() in before the t_relay() in the failure. This stops the error and now we send an ACK and a new INVITE, but still without the expected authorisation header. I feel like my lack of understanding about branches is the issue here - I seem to be sending an INVITE, but not adding the auth header that I suspect is being generated somewhere! Thanks for your help. ________________________________ Andrew White - Director uConnected Email: andrew(a)uconnected.com.au Web: www.uConnected.com.au

2 2

Statistics of received replies
by Duarte Rocha 04 Apr '19

04 Apr '19

Greetings, Does Kamailio have statistics like the sl.stats but for received replies instead of locally generated ones? I want to be able to look at how many "491" replies i'm receiving, for example. Best Regards, Duarte Rocha

2 1

Kamailio v5.0.8 Released
by Daniel-Constantin Mierla 04 Apr '19

04 Apr '19

Hello, Kamailio SIP Server v5.0.8 stable release is out. This is a maintenance release of the old stable branch 5.0, that includes fixes since the release of v5.0.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous v5.0.x versions are strongly recommended to be upgraded to v5.0.8 or even better directly to v5.1.x or v5.2.x because v5.0.x is getting out of the official maintenance time frame. Important note: this is the last planned release from branch5.0. For more details about version 5.0.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-0-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.2, at this moment with its latest release v5.2.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-2-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Kamailio-error reading: Connection reset by peer
by Mangesh Kharade 04 Apr '19

04 Apr '19

Hi Team, We are facing issue with kamailio application , we are not able to communicate to kamailio application getting below error. Can you please suggestion us how to resolved this issue issue, it production servers and we did not get any clue to resolve it. Error1:- Apr 2 06:53:54 urcmcs /usr/local/sbin/kamailio[2240]: ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) Apr 2 06:53:54 urcmcs /usr/local/sbin/kamailio[2240]: ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Apr 2 06:54:12 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) Apr 2 06:54:12 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Apr 2 06:54:17 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) Apr 2 06:54:17 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Error1:- Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3008 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3009 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3010 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3011 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3012 has closed Please let us know if you need more help from our end. Thanks and Regards, Mangesh Kharade Consultant - Microsoft Azure | Managed Services [cid:image003.png@01D15FFB.27B1AA80] Pune IT Park, 34 Aundh Road Bhau Patil Marg, Pune 411020 Tel: +91.20.30204000 | Extn. 9694 VOIP: +1.713.353.7464 Mobile: +91.9168181166 Mangesh.kharade(a)silicus.com<mailto:Mangesh.kharade@silicus.com>| www.silicus.com<http://www.silicus.com/> The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.

2 2

Releasing Kamailio v5.0.8
by Daniel-Constantin Mierla 04 Apr '19

04 Apr '19

Hello, I am considering to release v5.0.8 to mark the end of official maintenance for branch 5.0 on Thursday, April 4. If no major regression will be introduced, this will be the last release from branch 5.0. Afterwards, the stable branches maintained will be 5.1 and 5.2. In case you have in mind some specific commits that should get into 5.0 branch, then reply and list them to be sure they are not missed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

2 2

Kamailio & asterisk integration
by ayoub krari 04 Apr '19

04 Apr '19

Good morning, I want to Loadbalance calls equally to asterisk servers, i have already tried with opensips but didnt work for me, i want to try kamailio now, i have tried with the tutorial of " Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database" but i got some errors and packages missing during the installation and the configuration, my Linux distro is Debian 8.7.0 and i’m using Asterisk v 13, i want really to implement Kamailio , any one can help me with steps or anything , Thanks in advance !

2 1

dispatcher "failover feature disabled"
by David Villasmil 03 Apr '19

03 Apr '19

Hello guys, In my config i have: # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # PVs for hashing modparam("dispatcher", "hash_pvar", "$fU@$ci") # PVs to store results when calling ds_is_from_list modparam("dispatcher", "setid_pvname", "$var(setid)") modparam("dispatcher", "attrs_pvname", "$var(attrs)") # method to probe the gateways modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", " sip:dispatcher@proxy.voice-proxy.singlecommand.com") modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_ping_reply_codes","class=2") modparam("dispatcher", "ds_probing_mode", 1) # size of hash table storing data for call load dispatching, power of two modparam("dispatcher", "ds_hash_size", 10) # expiration time in seconds to remove the load on a destination if no BYE was received modparam("dispatcher", "ds_hash_expire", 3600) # expiration time in seconds to remove the load on a destination if no 200 OK for INVITE was received # and state updated with ds_load_update modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 30) But i'm getting on startup: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled What avp am I missing? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 3

Re: [SR-Users] Adding presentity_uri prefix for multi-tenanted presence/dialoginfo configuration
by Rhys Hanrahan 03 Apr '19

03 Apr '19

Hi All, I managed to solve my need to re-write NOTIFY packets with help from this thread: https://www.mail-archive.com/sr-users@lists.kamailio.org/msg08088.html This seems to work well. I am re-writing the packets as shown below… I guess my only question is, is this relatively safe/smart to do? I basically copied what our standalone Asterisk PBXes are doing in my captures, so I guess so? But I’m not confident enough to say it won’t cause some weird edge case issues. I can only say it works fine with the Snom and Yealink handsets I am testing with. I’m just concerned because it feels a bit like a hack. For SUBSCRIBE I re-write: * Request URI and To header from 101@ sent by the handset to tenant101@ (to match the SIP username of the handset - tenant101@) For NOTIFY I re-write: * From and Contact header to remove the tenant prefix, to go back to 101@ to match the BLF key configuration on the handsets. Any advice would be appreciated on if this is a bad idea, and there’s a better way. Thanks! Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 12 227 Elizabeth St, Sydney NSW 2000 From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Rhys Hanrahan <rhys(a)nexusone.com.au> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Tuesday, 26 March 2019 at 8:56 pm To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Adding presentity_uri prefix for multi-tenanted presence/dialoginfo configuration Hi Everyone, I’ve recently just started working with Kamailio – thanks everyone for this amazing software. Like many people, I’m in the process of trying to put Kamailio in-front of Asterisk to allow it to scale out, and my plan is for Kamailio to take over registrations, usrloc and presence/dialoginfo, as we’ve had issues where handsets are failing to get BLF updates/notifys, so I am hoping a couple of Kamailio boxes can scale better in this regard. DMQ is particularly exciting in how it will allow me to build a truly distributed platform. But I’m struggling to get presence to work the way I need it to. Our environment is multi-tenanted, but we do not (and can’t really) use multi-domains. Instead, we prefix the SIP usernames with the “tenant name” such as tenanta101 and tenantb101. My problem is that all the BLFs are configured for 101@PBX with no tenant name in the User part of the URI so that internal dialing and call pickup will work. Because in Asterisk we use “subscribecontext” this hasn’t been a problem in the past – Asterisk knows for subscriptions coming from that handset that it belongs to that tenant’s context in the dialplan and they are “isolated”, so having handsets subscribe as 101@ in their request URI was never a problem. Of course, with Kamailio I don’t have subscribecontext, and my main issue is that the “presentity_uri” being stored is 101@ while each of the SIP accounts of the handsets are registered as tenenata101@, and as such, no NOTIFYs are sent by Kamailio because it thinks that there are no watchers for the presentity_uri. If I change the SIP account username to 101 to match the BLF key, NOTIFYs are sent as expected. But then this breaks call pickup and internal dialing using the BLF keys. I would rather handle this in Kamailio than in Asterisk and having to re-configure the BLF keys for hundreds of handsets. I need to do something like: * When a SUBSCRIBE comes in, I need to prefix presentity_uri with the tenant name so the subscription changes from 101@ to tenanta101@ in the active_watchers table. * When Kamailio generates a NOTIFY, the notify would be built based on presentity_uri and would come out as tenanta101@, but the phone’s BLFs are configured for 101@ so I would need to *remove* the tenant prefix before sending out the notify. Is there a good, or recommended way to handle this scenario? Maybe something entirely different to changing the presentity_uri? I’ve written the following config to re-write the request URI and To field in any incoming subscribe requests, and successfully got the active_watchers table to store a presentity_uri containing my tenant prefix. But the problem I am having is that I can’t figure out how to modify the NOTIFY packets before they are sent – and looking at cfgtrace, it seems I might not be able to? My “standard” presence + dialoginfo configuration was taken from here as a starting point: https://kb.asipto.com/kamailio:presence:k43-blf Here is what I have so far in terms of trying to make my modifications – any guidance would be greatly appreciated. It feels like there’s probably a better way to do this than re-writing critical headers like To and Request URI? @@ -466,6 +467,8 @@ request_route { # authentication route(AUTH); + route(REWRITE_PRESENCE); + # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { # Asterisk is our voicemail route(TOASTERISK); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { # See REWRITE_PRESENCE - this should have been executed before we get here. handle_subscribe(); t_release(); } exit; #!endif +route[REWRITE_PRESENCE] { + if (is_method("SUBSCRIBE")) { + xlog("Re-writing subscribe to include tenant prefix\n"); + # The default presentity_uri needs to be prefixed with + # the tenant name + # So re-write the To header + route(TENANTINFO); + # Now grab the tenant name. + $var(subscribe_ru) = "sip:" + $var(tenant_name) + $rU + "@" + $rd; + xlog("Re-writing SUBSCRIBE To header as: $var(subscribe_ru)\n"); + insert_hf("To: $var(subscribe_ru)\r\n", "From"); + $ru = $var(subscribe_ru); + # Force change immediately: + # http://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers… + msg_apply_changes(); + } else if (is_method("NOTIFY")) { + xlog("Re-writing NOTIFY to remove tenant prefix\n"); # This code block does not get executed right now. + # We are storing the presentity_uri with a tenant prefix + # handsets do not expect this, only 101@, 102@ etc... In their BLF configs + # So this tenant prefix must be removed when building the reply. + route(TENANTINFO); + $var(notify_ru) = "sip:" + $(rU{s.substr,$(var(tenant_name){s.len}),0}) + "@" + $rd; + xlog("Re-writng NOTIFY to use: $var(notify_ru)\n"); + insert_hf("To: $var(notify_ru)\r\n", "From"); + $ru = $var(notify_ru); + # Force change immediately: + # http://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers… + msg_apply_changes(); + } +} Thanks for your time. Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 12 227 Elizabeth St, Sydney NSW 2000 [ttp://quintus.nexusone.com.au/~rhys/nexus1-email-sig.jpg]

1 0

uac_replace_from/to, msg_apply_changes and Re-INVITE
by Henning Westerholt 03 Apr '19

03 Apr '19

Hello, I am currently debugging a strange issue related to the usage of uac_replace_from/to() after msg_apply_changes(). It works all fine, but in a Re-INVITE case it inserts the wrong headers for the 100 - Trying case. The calle side gets then a reply with wrong From/To. This happens all the time with this particular cfg. I already checked for the usual issues: - msg_apply_changes() called several times - msg_apply_changes() called after transactions created/async processing etc.. Any ideas what else could be the problem? The msg_apply_changes() is needed for some SDP filtering logic in the cfg. Thanks and regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

3 8

TOPOS Call-ID masking
by Cinthia Leung 02 Apr '19

02 Apr '19

Greetings, Call-ID masking is not available, although the modparam is there. Just wondering if there are any plans to get that in soon. Thanks. Cindy

2 2

ENUM for CNAM dips
by Spencer Thomason 02 Apr '19

02 Apr '19

Hi all, I’m interested in extending the ENUM module to add support for draft-ietf-enum-cnam-08 CNAM queries using ENUM. I’m looking for high level guidance as to the best way to implement this and any feedback on naming of exported functions and parameters. Maybe something like enum_cnam_query("pvar", "destination", [,"suffix" [,"service"]]) ? Best regards, Spencer

2 1

Info: Kamailio World 2019 – Participation Grants
by Daniel-Constantin Mierla 02 Apr '19

02 Apr '19

Hello, we continue our tradition to give away 3 free passes to students and underrepresented people at Kamailio World Conference (www.kamailioworld.com) – thanks to the sponsors, we are able to offer them in 2019 as well. The passes cover the entire registration fee for all three days and the social networking events – costs with travel and accommodation have to be covered by the participants. Anyone enrolled in an academic activity (including master and PhD programs) as well as underrepresented people can apply for one of the passes via email to me or to: * conference [at] kamailioworld.com or by contacting us via web form at: * https://www.kamailioworld.com/k07/contact-us/ Applications done until April 15, 2019, will be selected by April 18, 2019. Afterwards, if there are still available passes, they will be allocated in the first come first served policy. The details for most of the presentations were published, the full agenda is expected to be ready in a matter of a few days. You can see more details at: * https://www.kamailioworld.com/k07/schedule/ Looking forward to meeting many of you in Berlin, at Kamailio World 2019! It’s going to be another great event about open source real time communications! * https://www.kamailioworld.com/k07/registration/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Port 5060 omitted in Record-Route
by Lợi Đặng 02 Apr '19

02 Apr '19

Hi all, i was using kamailio 4.2.1 located in 2 networks > listen = tcp:MY_ADDR:5060 advertise MY_ADDR:5060 > listen = tls:MY_ADDR:5061 advertise PUBLIC_NAT_ADDR:5061 when the call made from the inside network to out side, running `record_route()` resulted in 2 Record-Route headers added (enable_double_rr=1) Record-Route: PUBLIC_NAT_ADDR:5061;transport=tls;lr > Record-Route: MY_ADDR;transport=tcp;lr That was totally fine omitting the port in the first Record-Route when using tcp (or udp) on the first realm, but when i start switching to tls, it caused trouble Record-Route: PUBLIC_NAT_ADDR:5061;transport=tls;lr > Record-Route: MY_ADDR;transport=tls;lr The client is then told to send ACK/BYE to `MY_ADDR;transport=tls` located at `MY_ADDR:5061` as per rfc3263, then the call would failed. I had another try with `record_route_preset("PUBLIC_NAT_ADDR:5061;transport=tls", "MY_ADDR:5060;transport=tls");`, it really did add what i want with explicit 5060 port on RR, `ACK/BYE` travel on the correct path, but `loose_route()` only consumes the local `Route` header (it should consume 2). So my assumption is to stick with `record_route()` function to make `loose_route()` work properly. I tried using another port on the local realm, e.g: 5062 and the port is explicitly added to the Record-Route header `MY_ADDR:5062;transport=tls;lr` So is `5060` couldn't be explicitly added to the inbound Record-Route, or i just missed something? Any help will be appreciated. P/S: I also tried 4.4.7 and it still omit my 5060 port in the RR. rgds, Loi Dang Thanh Phone : +84. 774.735.448 Email : loi.dangthanh(a)gmail.com

3 5

Compressing SIP messages
by Igor Olhovskiy 01 Apr '19

01 Apr '19

Hi! Was looking on some way to compress SIP signalling between internal servers. But found out, there are no some pre-defined or recommended mechanism of doing this. Yes, I found gzcompress module, but it will help only in case of big body, but not whole message itself. Like in some cases you can have 4-5 RR headers, which is already a lot of info. What I googled is SigComp Compact Header form As I found, SigComp is really not widely-used method and it support along opensource SIP servers/PBXes is not documented. For Compact Header form - is there any way of using Kamailio to compress message this way? Maybe also there is a way of having some intermediate proxy-layer software, that will gzip/gunzip SIP signalling in realtime? All of this actually cause info in SIP/SDP grows rapidly, unfortunately MTU size is not.

8 11

presence_dialoginfo module not sending publish on dialogue change
by Richard Nunley 01 Apr '19

01 Apr '19

To whom can be of assistance, I’m am working towards enabling presence on my Kamailio proxy but have hit a bit of a brick wall. I’m seeing dialogues being created and persisted to the database, subscriptions are being handled and rows are being written to the active_watchers table but no PUBLISH is ever being received by the server to kickoff the sending of NOTIFY to the active watchers. Further, I’m never seeing a row in the watchers table though I don’t know that this isn’t expect behavior. I’ve attached my current config for review. Any help will be greatly appreciated! Thank in advance!

3 2

Use Kamailio as forward proxy between two networks
by Lauritz 01 Apr '19

01 Apr '19

Hello, I am trying to implement the following network setup using Kamailio and RTPProxy, but was not able to find any recent documentation or blog posts with hints: > local_net1 : local_net2 > 10.250.10.* : 10.250.11.* > : > (1) (2) : > +-------------+ +--------------+ +--------------+ > ! !.11 .12 ! Proxy !.21 ! Debian Host !.22 > ! Starface !---------------! + Kamailio !<<--------------! + Linphone !.23 > ! ! ! + RTPProxy ! ! ! > +-------------+ +--------------+ +--------------+ > eth0 : eth1 > : > : > As you can see, the proxy has two interfaces and its purpose is to forward traffic from the client's network to the Starface's network. Has anyone experience to share or got any other hints where to start? Thank you very much for your help :) Kind regards, Lauritz

2 1

Kamailio Managed Cloud
by Alex Balashov 01 Apr '19

01 Apr '19

For immediate release: ATLANTA, GA (1 April 2019)--Following its long tradition of forward-thinking innovation in open-source Real-Time Communications (RTC), the leadership of the Kamailio SIP server project (formerly known as OpenSER) has launched the Kamailio Managed Cloud (KMC). Alex Balashov of Evariste Systems and Fred Posner of The Palner Group were among several key North American stakeholders on hand to discuss it further. As Posner told us, "Kamailio Managed Cloud is a new, best-of-breed managed Kamailio hosting environment. We provide elasticity by leveraging human capital. Using our patent-pending Orchestration Turk system, our managed cloud technicians will purchase your SuperMicro bare-metal server, known as a KamBox, drive to the data centre, and rack-and-stack it on demand -- all in response to a SOAP API call! And when you want to remove your on-demand KamBox, just issue a 'delete' operation and we will decommission the hardware -- live!" "Dedicated KamBox colocation really puts the Management in Managed Cloud," emphasised Balashov. "But that doesn't mean we keep the customer out of the loop. We are using the most modern XMLHttpRequest and 'AJAX' techniques to provide a real-time dashboard with KamBox status to all users, though it will not work with Internet Explorer 6 -- we're addressing that." "Anyway, there are numerous other cloud platforms out there, many of which rely on virtualisation and automated orchestration and the like. It really takes the human touch out of the equation and gives the short shrift to quality, while also taking away jobs. And besides, how can a so-called 'virtual' machine be truly carrier-grade?" Kamailio Managed Cloud launched in two principal regions today -- US-EAST-1, located in Jacksonville, Florida, USA, and EU-CENTRAL-1, based in the booming tech hub of central Europe and birthplace of OpenSER: Berlin, Germany. Unnamed sources tell us that strategic expansion is planned for several other regions, including Flint, Michigan, USA. The mechanics of Kamailio Managed Cloud have invited questions about how other elastic cloud resources are managed, more especially elastic storage. According to KMC senior management, storage is provided via a mechanism developed by ASIPTO StorageWorks. ASIPTO chief Daniel-Constantin Mierla was on hand at the KMC launch to explain: "It's very easy. When a customer requests more space with a simple and lightweight SOAP call with just a little custom XML namespaces, then depending on the region, Fred or I receive an SMS on our smartphones from our advanced PHP backend and drive to the data centre to insert another hard drive into one of the KamBox's several drive bays. We are using the latest in advanced mobile communications, SMS, to ensure fastest response time. Of course our service level is comprehensive and includes Managed Partition Table Labelling and Managed LVM Insertion. Also there is optional Managed Filesystem Formatting for a complete portfolio of managed services. ASIPTO StorageWorks moves at the speed of global e-business. Just text us your desired filesystem layout and partitioning scheme. It is really that easy!" When asked about business focus, marketing turnaround veteran Henning Westerholt, Executive Vice President of Business Development for KMC responded: "Our target customer is a socially conscious Millennial-run corporation or start-up who see value in the unique blend of managed services and an artisanal, hand-crafted, organic approach that is compatible with social justice and takes a people-centric view." Asked to elaborate upon his vision of social justice, Westerholt added: "Virtualisation, orchestration and auto-scaling take the people out of the equation, and Kamailio is, in the end, about people, as communication in general is about people. What would it be without the people? Moreover, these mechanisms take away gainful opportunities from hard-working university graduates. Kubernetes, Docker Swarm, VMWare, and KVM contravene the social policy objective of full employment." At press time, Posner, who also heads up KMC Labs North America, was said to be lending his pioneering business vision and knack for seeing around technology trend corners to intensive work on next-generation, bleeding-edge functionality for KMC. While Posner was tight-lipped about the specifics, he did state that the 'dispatcher' module is slated to be deprecated in favour of a "human load-balancing approach to SIP requests that is both cloud-native and eco-conscious". Industry analysts agree that protocol-aware load-balancing strategies are complicated and that the application of human judgment to the problem represents a fertile field for dynamic innovation and investment. Reports from regional press confirm that KMC Labs are in the process of building a fullfilment centre for this innovative offering, said to be branded "Central Switchboard", at an undisclosed site in Bangladesh, where thousands of "Load Balance Operator" jobs have been advertised in and around the Dhaka market. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

db_redis and user
by Joan Salvatella 01 Apr '19

01 Apr '19

Hello, We are working on a project that's using two active-active Kamailio 5.2 instances that redirect traffic to several Asterisk 15.4 instances. Kamailio is acting as a registrar and should store the location information of the registered terminals. The backend database is db_redis. Phone terminals get registered and the information gets looked up from the subscriber schema in redis (That's cool) but we are having problems with the locations. This is the related code: modparam("db_redis", "keys", "location=entry:ruid&usrdom:username,domain&timer:partition,keepalive") if(ds_is_from_list("1", "2", "sip:$fd:5080")) { record_route(); route(INVITE_ASTERISK); exit; } route[INVITE_ASTERISK] { xlog("L_INFO", "[R-INVITE-RESOLVER:$ci] Entering ASTERISK INVITE resolver\n"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; } route(INVITE_POSTROUTE); } route[INVITE_POSTROUTE] { rtpproxy_manage(); t_on_reply("MAIN_REPLY"); if(!t_relay()) sl_reply_error(); } When looking up the location schema this errors appear: 9(39) WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'location' while performing query 9(39) WARNING: db_redis [redis_dbase.c:1101]: db_redis_perform_query(): scan key 0 is 'expires' 9(39) WARNING: db_redis [redis_dbase.c:1101]: db_redis_perform_query(): scan key 1 is 'expires' 9(39) ERROR: <core> [db_ut.c:92]: db_str2int(): Unexpected characters: [-03-29 18:16:36] 9(39) ERROR: <core> [db_val.c:76]: db_str2val(): error while converting integer value from string 9(39) ERROR: db_redis [redis_dbase.c:1062]: db_redis_convert_row(): Failed to convert redis column 'expires' to db value 9(39) ERROR: db_redis [redis_dbase.c:1231]: db_redis_perform_query(): Failed to convert redis reply for row 9(39) ERROR: db_redis [redis_dbase.c:1243]: db_redis_perform_query(): failed to perform the query 9(39) ERROR: db_redis [redis_dbase.c:1829]: db_redis_query(): failed to do the query 9(39) ERROR: usrloc [udomain.c:981]: udomain_contact_expired_cb(): db_query failed Any hints? Thanks and regards,

3 2

Kamailio v5.2 kamctl alias add problem
by Igor Potjevlesch 01 Apr '19

01 Apr '19

Hi kamailios, I meet a problem with the " kamctl alias add" command in the 5.2 version, I saw that the module used to add an alias had changed, and now the command result with an error : [kamailio]# kamctl alias add XXXX@XXXX sip:XXXXX@XXXX { "jsonrpc": "2.0", "error": { "code": 500, "message": "Not enough parameters or wrong format" }, "id": XXXXX } I read the function who return this message " static void ul_rpc_add(rpc_t* rpc, void* ctx)" in the ul_rpc.c in the usr_loc module. I tried to pass the command with two more parameters : # kamctl alias add 0175951068(a)trunk7.bornsip.fr sip:0175951068@trunk7.bornsip.fr NULL NULL { "jsonrpc": "2.0", "result": { }, "id": 29200 } And this time it works correctly, but I saw thaht the test prameters was in the field "path" , do you know why we have to indicate a path ? Are we obliged to indicate this two more parameters ? What is the utility of this field ? Thank you in advance, Igor

2 2

where to find the cfg languaje for program on it?
by PICCORO McKAY Lenz 01 Apr '19

01 Apr '19

recently i wrote a portion of my cfg file but i found that i based my code in other sources and results of try/tach experiments i cannot found the documents for the languaje of the core i only found the link: https://www.kamailio.org/wiki/cookbooks/devel/core but in that link there's no references about languaje per see only about how to write portions of the file and sections.. maybe i wrong or the document are wrote in different way? by example how to evaluate the output of a custom function/section? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

drop before exit and send ok reply first can be done?
by PICCORO McKAY Lenz 01 Apr '19

01 Apr '19

h have this for a custom rule agains spoecific messages: sl_send_reply("200", "OK"); drop(); exit; my question are if that rules are correct: i want to drop but send firts to client a ok response.. i mean client receive "all are ok dont worry" and then drop the rest packets.. that's correct? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 2

kamailio and rtpengine
by David Villasmil 01 Apr '19

01 Apr '19

Hello guys, i'm trying to configure rtpengine, but when starting kamailio (with an empty table) i'm getting ```build_rtpp_socks(): no more pkg memory for rtpp_socks``` Can anyone help? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 5

kamailio 5.2.0 with empty table of rtpengine - kamailio doesn't start
by * Paolo Visintin - evosip.cloud 31 Mar '19

31 Mar '19

Hello, I noticed that , with kamailio ver. 5.2.1 starting with empty rtpengine table causes strange messages in kamailio and it doesn't start modparam("rtpengine", "db_url", "text:///tmp/kamailio/dbtext") cat /tmp/kamailio/dbtext/rtpengine id(int,auto) setid(int) url(string) weight(int) disabled(int) stamp(int) 0(2856) WARNING: tm [tm.c:760]: mod_init(): reparse_on_dns_failover is enabled on a multihomed host -- check the readme of tm module! 0(2856) WARNING: rtpengine [rtpengine_db.c:100]: rtpp_load_db(): No rtpproxy instances in database 0(2856) ERROR: rtpengine [rtpengine.c:2768]: select_rtpp_set(): no rtpp_set_list->rset_first 0(2856) NOTICE: rtpengine [rtpengine.c:1664]: mod_init(): Default rtpp set 0 NOT found 1(2857) NOTICE: usrloc [udomain.c:434]: preload_udomain(): filtered by server_id[0] 2(2858) ERROR: rtpengine [rtpengine.c:1722]: build_rtpp_socks(): no more pkg memory for rtpp_socks 2(2858) ERROR: <core> [core/sr_module.c:847]: init_mod_child(): error while initializing module rtpengine (/usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so) (idx: 2 rank: 2 desc: [udp receiver child=1 sock=172.28.0.120:5060]) 2(2858) ERROR: <core> [core/pt.c:342]: fork_process(): init_child failed for process 2, pid 2858, "udp receiver child=1 sock=172.28.0.120:5060" 2(2858) CRITICAL: <core> [main.c:1633]: main_loop(): Cannot fork 4(2860) ERROR: rtpengine [rtpengine.c:1722]: build_rtpp_socks(): no more pkg memory for rtpp_socks 4(2860) ERROR: <core> [core/sr_module.c:847]: init_mod_child(): error while initializing module rtpengine (/usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so) (idx: 4 rank: 4 desc: [udp receiver child=3 sock=172.28.0.120:5060]) 4(2860) ERROR: <core> [core/pt.c:342]: fork_process(): init_child failed for process 4, pid 2860, "udp receiver child=3 sock=172.28.0.120:5060" 4(2860) CRITICAL: <core> [main.c:1633]: main_loop(): Cannot fork any hint on what could be wrong ? (5.2.0 works perfectly) many thanks! Paolo

6 7

app_ruby - KSR::PV.sets method not found
by Andrew White 31 Mar '19

31 Mar '19

Hi all, I’m trying to set a pseudovariable via app_ruby. I’m able to retrieve pseudovariables using PV.gete as such: However trying to set these using sets, I get “No method found”: KSR::PV.sets("$fu", "sip:#{$uacs[trunk]['r_username']}@#{$uacs[trunk]['r_domain']}") Mar 30 04:44:59 ip-10-0-0-2 /usr/local/sbin/kamailio[28421]: ERROR: {1 102 INVITE 0fae8097312bec2b5ec79ada30c227e7@127.0.0.1:5060} app_ruby [app_ruby_api.c:107]: app_ruby_print_last_exception(): exception: undefined method `[]' for nil:NilClass PV.seti appears to work correctly. Am I calling the wrong method? Thanks! ________________________________ Andrew White - Director uConnected Email: andrew(a)uconnected.com.au Web: www.uConnected.com.au

2 4

Low %
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30 Mar '19

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1 0

[--SPAM--] Low %
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30 Mar '19

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1 0

Lost connection to MySQL server during query
by JR Richardson 29 Mar '19

29 Mar '19

Hey Folks, Getting some annoying logs, pretty certain these are false positives. I'm running kamailio 4.4.7 using localhost mysql 5.5.60 on debian 7.11. Pretty simple setup, have acc and collecting cdrs. Relevant kamailio.cfg: loadmodule "acc.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----- acc params ----- modparam("acc", "db_url", DBURLRW) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "accountcode=$avp(s:acctcode)") modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql://rwuser:rwpass@localhost/kamailio") route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); } Logs: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013) ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [call kamailio_cdrs()] CDRs are working fine, no issues with the results, just the log messages. Errors occur every 5 minutes so this is related to the rtimer cycle, mode=1 is set for new connection so there really wouldn't be a lost connection, the old connection should have been relieved on last cycle. I've seen some other post talking about this issue but could not find any resolution or work around. Any help is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

3 3

log timestamp granularity
by Patrick Murphy 28 Mar '19

28 Mar '19

Hello - Is there a way I can print current timestamp in milli or microseconds in the log file? Something like: 2019-03-28 04:50:15.379 would be ideal The closest thing I could find was $TV(Sn) but it doesn't suit the purpose here since its not human readable. Thanks Pat

2 1

flags definition in kamailio.cfg + 5.2 documentation
by Joel Serrano 27 Mar '19

27 Mar '19

Hi, I saw in the Kazoo default.cfg this: ... ####### Flags ####### flags FLAG_INTERNALLY_SOURCED: 1, FLAG_ASSOCIATE_SERVER: 2, FLAG_SKIP_NAT_CORRECTION: 3, FLAG_ASSOCIATE_USER: 4, FLAG_TRUSTED_SOURCE: 5, FLAG_SESSION_PROGRESS: 6, FLAG_IS_REPLY: 7, FLAG_SIP_TRACE: 8; ... So then I went here to read about it: https://www.kamailio.org/wiki/cookbooks/5.2.x/core#flags I found out it's not documented... is this a drop-in alternative for a: #!define FLAG_NAME FLAG_BIT? Sorry, first time I see this if not I would update the docs myself... :( Thanks, Joel.

4 8

Uri's user part parameters
by Duarte Rocha 27 Mar '19

27 Mar '19

Greetings, Does Kamailio have a way of identifying the user-part parameters of an URI? I have situations where i have sip:user;parameter=yes@host. Best Regards, Duarte Rocha

2 1

Manipulating User Agent header in dispatcher OPTIONS
by Andrew White 27 Mar '19

27 Mar '19

Hi all, I’ve set the server_header variable in my primary config, and I’ve got a catch in the default reply route to modify headers where required. I’m masking this value for security reasons, and filling it with my own value for vanity reasons :). I’ve noticed that the OPTIONS packet that is send via the dispatcher module when probing mode is on contains a User-Agent field which has the full version and OS type. Unfortunately I don’t necessarily have trust of the end destination being dispatcher to, nor the packet switchers in-between. How can I catch this OPTIONS packet to manipulate the value? It doesn’t appear to go through the REQUEST route or the ONSEND route. Thanks! ________________________________ Andrew White - Director uConnected Email: andrew(a)uconnected.com.au Web: www.uConnected.com.au

2 2

accounting and db_redis
by Daniel Tryba 27 Mar '19

27 Mar '19

Under https://www.kamailio.org/docs/modules/5.2.x/modules/db_redis.html#db_redis.… there is an example for using db_redis for accounting. This doesn't appear to work (5.2.2) modparam("db_redis", "keys", "acc=entry:callid,time_hires&cid:callid") results in: db_redis [redis_dbase.c:1886]: db_redis_insert(): inserting to prefix (table) 'acc' db_redis [redis_dbase.c:382]: db_redis_build_entry_keys(): build entry keys db_redis [redis_dbase.c:295]: db_redis_find_query_key(): checking for existence of ent db_redis [redis_dbase.c:306]: db_redis_find_query_key(): found key in entry key db_redis [redis_dbase.c:133]: db_redis_val2str(): converting str value '1db9c0bc57291e db_redis [redis_dbase.c:338]: db_redis_find_query_key(): entry key so far is 'acc:entr db_redis [redis_dbase.c:295]: db_redis_find_query_key(): checking for existence of entry key 'time_hires' in query db_redis [redis_dbase.c:346]: db_redis_find_query_key(): key 'time_hires' for type 'entry' not found, unable to use this type db_redis [redis_dbase.c:405]: db_redis_build_entry_keys(): Failed to create direct entry key, no matching key definition db_redis [redis_dbase.c:1892]: db_redis_insert(): failed to build entry keys db_redis [redis_dbase.c:1976]: db_redis_insert(): failed to do the insert What does work is "time" instead of "time_hires". Since I have no idea what values are available (I guess the table names?) and time only has a 1 sec resolution, what is a correct version of time_hires?

1 2

DBText does not exists!
by Andrew White 27 Mar '19

27 Mar '19

Hi all, I’m currently playing with the UAC module to hand off remote registrations with trunks to Kamailio. I want to keep Kamailio’s external connections low, so I’m planning to use db_text to load the UAC info, and populate the flat file via other methods. When attempting to load my UAC DB via db_text however, I get the following: Mar 26 04:59:30 ip-10-0-0-2 /usr/local/sbin/kamailio[31904]: ERROR: db_text [dbt_lib.c:143]: dbt_cache_get_db(): database [/etc/kamailio/uac.db] does not exists! Mar 26 04:59:30 ip-10-0-0-2 /usr/local/sbin/kamailio[31902]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/31902 Mar 26 04:59:30 ip-10-0-0-2 /usr/local/sbin/kamailio[31904]: ERROR: db_text [dbt_base.c:102]: dbt_init(): cannot get the link to database Mar 26 04:59:30 ip-10-0-0-2 /usr/local/sbin/kamailio[31904]: ERROR: uac [uac_reg.c:1318]: uac_reg_load_db(): failed to connect to the database When running using WITH_DEBUG flag, it does not appear to attempt on normal start. A kamcmd uac.reg_reload yields: Mar 26 05:07:52 ip-10-0-0-2 kamailio: 22(32177) DEBUG: ctl [io_listener.c:442]: handle_new_connect(): handle_stream read: new connection (1) on /var/run/kamailio//kamailio_ctl Mar 26 05:07:52 ip-10-0-0-2 kamailio: 22(32177) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): found export of <db_bind_api> in module db_text [/usr/local/lib64/kamailio/modules/db_text.so] Mar 26 05:07:52 ip-10-0-0-2 kamailio: 22(32177) DEBUG: <core> [db.c:210]: db_bind_mod(): using db bind api for db_text Mar 26 05:07:52 ip-10-0-0-2 kamailio: 22(32177) DEBUG: db_text [dbt_base.c:55]: dbt_init(): initializing for db url: [text:///etc/kamailio/uac.db] Mar 26 05:07:52 ip-10-0-0-2 kamailio: 22(32177) DEBUG: db_text [dbt_lib.c:125]: dbt_cache_get_db(): looking for db /etc/kamailio/uac.db! Mar 26 05:07:52 ip-10-0-0-2 kamailio: 22(32177) ERROR: db_text [dbt_lib.c:143]: dbt_cache_get_db(): database [/etc/kamailio/uac.db] does not exists! Mar 26 05:07:52 ip-10-0-0-2 kamailio: 22(32177) ERROR: db_text [dbt_base.c:102]: dbt_init(): cannot get the link to database Mar 26 05:07:52 ip-10-0-0-2 kamailio: 22(32177) ERROR: uac [uac_reg.c:1318]: uac_reg_load_db(): failed to connect to the database Mar 26 05:07:52 ip-10-0-0-2 kamailio: 22(32177) DEBUG: ctl [io_listener.c:495]: handle_stream_read(): handle_stream read: eof on /var/run/kamailio//kamailio_ctl The file of course does exist, and has fine permissions (Kamailio runs as root in this case): -rw-r--r-- 1 root root 169 Mar 26 04:49 /etc/kamailio/uac.db Contents of file: l_uuid(str) l_username(str) l_domain(str) r_username(str) r_domain(str) realm(str) auth_username(str) auth_password(str) auth_proxy(str) expires(int) flags(int) reg_delay(int) 123:test:test.com:test:test.com:test.com:user:pass:10.0.0.100:600:: (At this stage I just want to see the reg packet in Wireshark). 1) Is this the correct method to do UAC? Load credentials, then Kamailio will handle the registration and renewal? I’ve struggled a bit to understand the documentation, hence the testing 2) Why is my db_text failing to load the file? Thanks! ________________________________ Andrew White - Director uConnected Email: andrew(a)uconnected.com.au Web: www.uConnected.com.au

3 6

Extract URI from Headers
by Duarte Rocha 27 Mar '19

27 Mar '19

Greetings, In my SBC i usually have to do operations on P-Asserted-Identity and Contact header. For those operations i usually have to get the user part, domain, etc. For that i'm using the URI Transformations provided by Kamailio. However, in order to do this i have to trim the header value by hand in order to remove brackets, display infos, etc. Does Kamailio have a method for extracting the URI from those kinds of headers? Best Regards, Duarte Rocha

3 3

Disable Dispatcher GW from DB
by Patrick Wakano 26 Mar '19

26 Mar '19

Hello list, Hope you are all doing well! I am looking for a way to disable a Dispatcher GW from the DB. I know I can disable it using the "kamcmd dispatcher.set_state d ..."command, but in case I do a reload or restart after that, the disabled GW will be back as active. Of course I could also delete the GW from DB, but for CDR purposes I want to keep the entry but mark as permanently disabled, so Kamailio won't try to use it nor ping it anymore. Does anyone how can I do that in the DB? I am looking for the exact same behavior the "kamcmd dispatcher.set_state d" does but from some change in the DB. Thanks, Kind regards, Patrick Wakano

3 5

Presence module. How to update contact DB record after re-SUBSCRIBE ?
by Denys Pozniak 26 Mar '19

26 Mar '19

Hello! We are using Presence module with MySQL with parameters below. And looks like module does not update contact DB record in case of re-SUBSCRIBE. So we have a case when NAT router periodically changes the outgoing port, but NOTIFY from Kamailio goes to port from initial SUBSCRIBE. Is this normal module behavior or we missed something? modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "expires_offset", 0) modparam("presence", "max_expires", 1800) modparam("presence", "db_update_period", 30) modparam("presence", "clean_period", 180) modparam("presence", "send_fast_notify", 1) modparam("presence", "pres_htable_size", 32) modparam("presence", "subs_htable_size", 32) modparam("presence", "publ_cache", 0) modparam("presence", "notifier_processes", 0) # kamailio -v version: kamailio 4.4.2 (x86_64/linux) 892ad6 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 892ad6 compiled on 14:45:42 Sep 26 2017 with gcc 4.8.5 -- BR, Denys Pozniak

1 0

Adding presentity_uri prefix for multi-tenanted presence/dialoginfo configuration
by Rhys Hanrahan 26 Mar '19

26 Mar '19

Hi Everyone, I’ve recently just started working with Kamailio – thanks everyone for this amazing software. Like many people, I’m in the process of trying to put Kamailio in-front of Asterisk to allow it to scale out, and my plan is for Kamailio to take over registrations, usrloc and presence/dialoginfo, as we’ve had issues where handsets are failing to get BLF updates/notifys, so I am hoping a couple of Kamailio boxes can scale better in this regard. DMQ is particularly exciting in how it will allow me to build a truly distributed platform. But I’m struggling to get presence to work the way I need it to. Our environment is multi-tenanted, but we do not (and can’t really) use multi-domains. Instead, we prefix the SIP usernames with the “tenant name” such as tenanta101 and tenantb101. My problem is that all the BLFs are configured for 101@PBX with no tenant name in the User part of the URI so that internal dialing and call pickup will work. Because in Asterisk we use “subscribecontext” this hasn’t been a problem in the past – Asterisk knows for subscriptions coming from that handset that it belongs to that tenant’s context in the dialplan and they are “isolated”, so having handsets subscribe as 101@ in their request URI was never a problem. Of course, with Kamailio I don’t have subscribecontext, and my main issue is that the “presentity_uri” being stored is 101@ while each of the SIP accounts of the handsets are registered as tenenata101@, and as such, no NOTIFYs are sent by Kamailio because it thinks that there are no watchers for the presentity_uri. If I change the SIP account username to 101 to match the BLF key, NOTIFYs are sent as expected. But then this breaks call pickup and internal dialing using the BLF keys. I would rather handle this in Kamailio than in Asterisk and having to re-configure the BLF keys for hundreds of handsets. I need to do something like: * When a SUBSCRIBE comes in, I need to prefix presentity_uri with the tenant name so the subscription changes from 101@ to tenanta101@ in the active_watchers table. * When Kamailio generates a NOTIFY, the notify would be built based on presentity_uri and would come out as tenanta101@, but the phone’s BLFs are configured for 101@ so I would need to *remove* the tenant prefix before sending out the notify. Is there a good, or recommended way to handle this scenario? Maybe something entirely different to changing the presentity_uri? I’ve written the following config to re-write the request URI and To field in any incoming subscribe requests, and successfully got the active_watchers table to store a presentity_uri containing my tenant prefix. But the problem I am having is that I can’t figure out how to modify the NOTIFY packets before they are sent – and looking at cfgtrace, it seems I might not be able to? My “standard” presence + dialoginfo configuration was taken from here as a starting point: https://kb.asipto.com/kamailio:presence:k43-blf Here is what I have so far in terms of trying to make my modifications – any guidance would be greatly appreciated. It feels like there’s probably a better way to do this than re-writing critical headers like To and Request URI? @@ -466,6 +467,8 @@ request_route { # authentication route(AUTH); + route(REWRITE_PRESENCE); + # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { # Asterisk is our voicemail route(TOASTERISK); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { # See REWRITE_PRESENCE - this should have been executed before we get here. handle_subscribe(); t_release(); } exit; #!endif +route[REWRITE_PRESENCE] { + if (is_method("SUBSCRIBE")) { + xlog("Re-writing subscribe to include tenant prefix\n"); + # The default presentity_uri needs to be prefixed with + # the tenant name + # So re-write the To header + route(TENANTINFO); + # Now grab the tenant name. + $var(subscribe_ru) = "sip:" + $var(tenant_name) + $rU + "@" + $rd; + xlog("Re-writing SUBSCRIBE To header as: $var(subscribe_ru)\n"); + insert_hf("To: $var(subscribe_ru)\r\n", "From"); + $ru = $var(subscribe_ru); + # Force change immediately: + # http://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers… + msg_apply_changes(); + } else if (is_method("NOTIFY")) { + xlog("Re-writing NOTIFY to remove tenant prefix\n"); # This code block does not get executed right now. + # We are storing the presentity_uri with a tenant prefix + # handsets do not expect this, only 101@, 102@ etc... In their BLF configs + # So this tenant prefix must be removed when building the reply. + route(TENANTINFO); + $var(notify_ru) = "sip:" + $(rU{s.substr,$(var(tenant_name){s.len}),0}) + "@" + $rd; + xlog("Re-writng NOTIFY to use: $var(notify_ru)\n"); + insert_hf("To: $var(notify_ru)\r\n", "From"); + $ru = $var(notify_ru); + # Force change immediately: + # http://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers… + msg_apply_changes(); + } +} Thanks for your time. Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 12 227 Elizabeth St, Sydney NSW 2000 [ttp://quintus.nexusone.com.au/~rhys/nexus1-email-sig.jpg]

1 0

kamcmd error
by Zhong, Alec (NSB - CN/Qingdao) 26 Mar '19

26 Mar '19

Hello Could you please help check below error? I encountered below error when to use kamcmd to get the value of one script_counter. [cid:image001.png@01D4E3EB.98F9BB70] BR, Alec

1 0

FW: add script counter.
by Zhong, Alec (NSB - CN/Qingdao) 26 Mar '19

26 Mar '19

Hello, Could you please help check below error for counters module? BR, Alec From: Zhong, Alec (NSB - CN/Qingdao) Sent: 2019年3月26日 10:56 To: 'sr-dev(a)lists.kamailio.org' <sr-dev(a)lists.kamailio.org> Subject: add script counter. Hello, Could you please help check below error? I would like to add a script counter to count the number of received message by kamailio. Anything wrong to use counters module? Should counters.so be loaded? Thank you in advance. Kamailio.cfg change [cid:image001.png@01D4E3C2.32E8DFB0] Kamailo log [cid:image002.png@01D4E3C2.8237E9D0]

1 0

User part on contact header
by Duarte Rocha 25 Mar '19

25 Mar '19

Greetings, On an INVITE the user part in the Contact header mandatory? Can't seem to find info on this. Best regards, Duarte Rocha

3 2

Default port interference in TLS conneciton
by Dalibor Zeman 24 Mar '19

24 Mar '19

Hello, I am running Kamailio v. 5.1.2 with typical configuration for wss connections (default config + tls, websocket, nathelper modules), quite everything is running as should, but I am quite concerned about what I got on Homer server. My Kamailio is listening on port 4443 for TLS. Opening dialog runs as should, then UPDATE and BYE messages are received on 4443, but sent out on 5060. When I don't set up default port, 403 reply comes on a call. I'd like to know whether it is something defined by standard or it has a meaning and I should leave it as it is, or I should correct it somehow and then how. Also, if it has a meaning, then what it is. Thanks for your answer, Dalibor Zeman.

1 0

Modifying NOTIFY of presence_dialoginfo
by Kevin Olbrich 24 Mar '19

24 Mar '19

Hi! I need to customize the content of the NOTIFY sent by presence_dialoginfo. How can I change the message before it is sent? Kind regards Kevin

2 2

KEMI Ruby - exit from within function
by Andrew White 23 Mar '19

23 Mar '19

Hi all, I know ruby is pretty new to KEMI, so I understand this might not be supported. I’ve been able to build a functional basic configuration using KEMI based app_ruby. However the function KSR::x.exit is not exposed to the interpreter. Reading up on the function (https://kamailio.org/docs/tutorials/devel/kamailio-kemi-framework/kemimods/… <https://kamailio.org/docs/tutorials/devel/kamailio-kemi-framework/kemimods/…>) I can see either a native `exit` or a return from the request_route is recommended. The latter of these works fine, however the former throws the following error when called either within the request_route function or any others: Mar 22 03:40:46 guest /usr/sbin/kamailio[17474]: ERROR: {1 19731 BYE 8u6We4ReU01GuU-zoa7VuqC.qWwwg29S} app_ruby [app_ruby_api.c:105]: app_ruby_print_last_exception(): exception: exit Mar 22 03:40:46 guest /usr/sbin/kamailio[17474]: ERROR: {1 19731 BYE 8u6We4ReU01GuU-zoa7VuqC.qWwwg29S} app_ruby [app_ruby_api.c:1078]: app_ruby_run_ex(): ruby exception (6) on callback for: ksr_request_route (res type: 17) It appears the app_ruby wrapper is seeing an exit as an exception and throwing an error. I’m unsure if this is affecting my actual flow (still building the config), but the error in the logs is certainly annoying. Returning from the request_route function works fine, however this is obviously harder to do from within a sub function. Any direction on how to correctly exit in similar ways to native or other KEMI implementations would be appreciated. By the way, thanks Daniel for writing this module! This makes Kamailio significantly more accessible for myself and others like me! ________________________________ Andrew White - Director uConnected Email: andrew(a)uconnected.com.au Web: www.uConnected.com.au

2 3

Kamailio initial log - Mysql error
by Jorge Cornejo 22 Mar '19

22 Mar '19

Hi, I am new at Kamailio, I have installed a vicidial goautodial box and having some headaches. I can see at starup logs the following message: Mar 18 02:12:34 edialer1 /usr/sbin/kamailio[4028]: ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2 "No such file or directory") Mar 18 02:12:34 edialer1 /usr/sbin/kamailio[4028]: ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool Mar 18 02:12:34 edialer1 /usr/sbin/kamailio[4028]: ERROR: usrloc [dlist.c:654]: register_udomain(): failed to open database connection However if I execute the following command: [root@edialer1 kamailio]# kamctl db show subscriber database engine 'MYSQL' loaded Control engine 'RPCFIFO' loaded +----+------------+----------------------------+----------+---------------+- ---------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+------------+----------------------------+----------+---------------+- ---------------------------------+----------------------------------+------+ | 18 | 5474470533 | vaglxc01.goautodial.com | | NULL | 10685ad93a251bb460e7dd544a174288 | 1680d2aebe5f750c4eedb45ad6b5ba4f | NULL | | 20 | 5505791853 | edialer1.gevolution.com.br | | NULL | 04e377e3923716fb5038b0caa79b8ed3 | d65362c7ccd934f00efa21c45e479ec5 | NULL | | 21 | 2347600834 | edialer1.gevolution.com.br | | NULL | 065fc461faa06cec682213cf17352630 | 4d79bc7db16f74d13d1785dafb30391c | NULL | +----+------------+----------------------------+----------+---------------+- ---------------------------------+----------------------------------+------+ I get a DB response.so is Kamailio actually connected to the db or not? Also at logs I get Mar 18 02:13:35 edialer1 /usr/sbin/kamailio[4513]: DEBUG: db_mysql [km_res.c:66]: db_mysql_get_columns(): 1 columns returned from the query So I assume it read from the DB something.This is quite confusing as can't say properly if it is working or not. Why I am not sure? Because can't see any information on it sending information to asterisk listening on port 5070, logs won't show such action. As I can't find the asterisk information at the kamailio.cfg file, I assume it is stored at the DB, and if DB connection is broken, would explain why it doesn't work. Can you please help me to figure this out? Thanks in advance

2 1

How fair is the rtpengine loadbalancing?
by Daniel Tryba 22 Mar '19

22 Mar '19

I have 3 rtp backends defined with equal weights (33 each). But when I look at the number of calls being handled the spreaded load is always 5:4:3 for the machines in the orderd listed: modparam("rtpengine", "rtpengine_sock", "udp:10.235.32.60:7723=33 udp:10.235.32.59:7723=33 udp:10.235.32.58:7723=33") Looking at select_rtpp_node_new I see nothing unfair happening (with my total lack of knowledge of how the node set is populated). Anybody else noticing this?

2 1

Auth for SUBSCRIBE and others
by Kevin Olbrich 22 Mar '19

22 Mar '19

Hi! I am implementing forwarding of SUBSCRIBE (BLF) to an Asterisk behind Kamailio. This works but Kamailio is not requesting for Auth. I then added SUBSCRIBE to: https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg#L746 And it now challenges the client correctly. Why does this line only show REGISTER? Shouldn't it request a challenge for all messages? And why does it work with INVITES ootb? Thank you! Kind regards Kevin

3 10

Job Offering: Software Engineer & Dev-Ops for VoIP & IMS
by Carsten Bock 22 Mar '19

22 Mar '19

Hi everyone, due to our ongoing growth in the area of mobile network infrastructure, we are currently looking into expanding our team. We are especially looking for: - Software Engineer VoIP & IMS ( https://www.ng-voice.com/software-engineer-voip-ims) - Dev-Ops (VoIP & IMS) (https://www.ng-voice.com/dev-ops-voip-ims) About us ng-voice was founded by a team of leading open-source and VoIP pioneers and passionate coders to develop software for mobile network operators. Our virtualized, cloud native solutions allow entrepreneurial-thinking mobile network operators to become software-centric and offer voice services on any 4G, 5G, WiFi or any other data network – from regular mobile network operators to private networks in remote areas. We are growing the team both in Hamburg and Berlin and are looking for new colleagues to join us. Your responsibilities • Help us deploying and integrating our software solutions • Have fun testing, validating and optimizing our solutions • Enjoy building things and solving complex problems • Proactively identify ways to unlock operational efficiency • Take care of your teammates and have fun growing a company Our requirements We are more interested in your experience and knowledge than in formal degrees: • Strong interest in telephony • Experience with signalling standards (e.g. SIP, DIAMETER) • Fluent in English • Efficient at driving deadlines and keeping yourself and others on topic Our technology stack • Docker; experience with Docker Swarm and Kubernetes is beneficial • Kamailio, RTPEngine, FreeSwitch • Redis, MySQL and RabbitMQ • SIP, RTP, DIAMETER What we offer • Flexible working hours and great office locations (Berlin and Hamburg) • Entrepreneurial culture and flat hierarchies • High degree of autonomy and no dress code • Competitive salary Please send your application to jobs(a)ng-voice.com. Let us know how your profile fits our requirements. And please let us know the earliest date you could join us plus your salary expectations. We look forward to getting to know you! Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

1 0

How can one change acc table name in Kamailio.cfg
by Z 22 Mar '19

22 Mar '19

For some reason these two modparams are always ignored. The table name used is always acc which is default name. How can I change the table name? loadmodule "acc.so" modparam("acc", "db_table_acc", "*acc_$time(year)_$time(mon)*") modparam("acc", "db_table_missed_calls", "*acc_$time(year)_$time(mon)*") Accounting works as expected otherwise. Appreciate any help. Thanks.

3 6

Re: [SR-Users] Problem at Rx Interface - kamailio P-CSCF 5.1.6
by Valentin Christoph 20 Mar '19

20 Mar '19

Hi all, Remember this question from 23.1. It's solved now, thanks for help:-) All the best Christoph Von meinem Samsung Galaxy Smartphone gesendet. -------- Ursprüngliche Nachricht -------- Von: Valentin Christoph <Christoph.Valentin(a)kapsch.net> Datum: 23.01.19 15:06 (GMT+01:00) An: sr-users(a)lists.sip-router.org Cc: Scherney Theodor <Theodor.Scherney(a)kapsch.net>, Onic Roman <Roman.Onic(a)kapsch.net> Betreff: [SR-Users] Problem at Rx Interface - kamailio P-CSCF 5.1.6 Hi all, We are using kamailio 5.1.6, configured as P-CSCF, facing following problem. Please indicate, if you need further information – e.g. I could provide a tshark trace. If during call setup (by simulated UAs) at the terminating P-CSCF the simulated PCRF rejects the Diameter AAR message, then the kamailio P-CSCF runs into a questionable behaviour. By calling the Rx_AAR() function in configuration, the processing of the 200 OK is suspended (by the ims_qos module) and then resumed with the receipt of the AAA message. Now the ims_dialog module is called with the dlg_terminate(“all”, “reason”) function, but the 200 OK has not yet been sent. The ims_dialog sends CANCEL downstream and 488 upstream. However, the 200 OK is sent additionally. We would rather have expected the ims_dialog module should send BYE in both directions. When we respond the CANCEL with 481 Call/Transaction does not exist (which we MUST according to RFC 3261), then we would expect the ims_dialog module at least now to react with BYE in both directions. Could you help? What’s our misunderstanding? How could we proceed to get this issue fixed? Kind regards Christoph P.S.: This is a retransmission, because the original message did not pass moderator approval due to the attached tshark trace The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

IMS: ims_isc, ims_registrar_scscf: some additions for 3rd party registration
by Valentin Christoph 20 Mar '19

20 Mar '19

Dear all, Dear Daniel, We had implemented a few additions for third party registration, based on kamailio 5.1.0 IMS modules. * either service info OR original REGISTER request or original 200 OK can be added as body to the 3rd party Registration Request, if indicated by the HSS in the user data * change: S-CSCF does not check for TEL URI, before it sends P-Associated URI to AS (SIP URIs are also sent) We would like to provide a pull request within the next days or week to have the changes as basis for a change in release 5.3.x. Drawback: probably our solution would need to be slightly modified, because we removed the check for TEL URI, if the S-CSCF sends the P-Associated-URI header to the AS in the 3rd party register request (maybe introduce a module parameter?). Any comments? Would it be reviewed and maybe taken for 5.3? All the best Christoph ____ Christoph VALENTIN Software Developer | R&D Core / SIP Core P +43 50 811 3785 | M +43 664 628 3785 christoph.valentin(a)kapsch.net<mailto:christoph.valentin@kapsch.net> Kapsch CarrierCom AG | Lehrbachgasse 11 | 1120 Vienna | Austria Company Register at: Commercial Court Vienna FN 223804 z | Registered Office Vienna | www.kapsch.net<http://www.kapsch.net/> [Kapsch Logo]<https://www.kapsch.net/> The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

2 1

event_route tm:branch-failure and onreply_route shared variables
by Володимир Іванець 20 Mar '19

20 Mar '19

Hello! I have Kamailio setup with registrar configured to allow multiple registrations. *branch_route[MANAGE_BRANCH]* contains *t_on_reply("REPLY")* and *t_on_branch_failure("FAILURE")*. *event_route[tm:branch-failure:FAILURE]* is used to do some logic, store information in htable and in the end it calls *t_relay* function. This creates new branch. Replies for this branch go to* onreply_route[REPLY]*. The problem is that I can not find appropriate variable that would be available in both *event_route tm:branch-failure* and *onreply_route*. I need it to get saved data from htable. $T_reply_ruid is available in first one but not in second. $T_branch_idx obviously changes so it can not be used. Some registrations do not have *Received* value so source IP address and ports can not be used. *Address* value can be compared with *Contact* header from responce tacket but it is so long that it looks like it's not optimal way to do. What interesting is that $T_reply_ruid is accessible in original replies for branches that have not failed. Does anyone know what variable can be used in this case or how to make sure that $T_reply_ruid will be set? Thanks a lot!

3 6

Pseudo-Variable {$au} is NULL
by Daniel Meyer-Wendt 20 Mar '19

20 Mar '19

Hi All, I am updating our Kamailio from 4.0.6 to the latest 5.2.2, we use kamailio as LB in front of two asterisk server. My problem is that during registration forward to the asterisk servers {$au} is NULL This is the code we have, which is the one you find in most documentations $uac_req(furi)="sip:" + $au + "@" + $var(rip); I logged some Pseudo-Variable to see their contents: {$au} NULL {$aU} NULL {$Au} username(a)domain.com {$fU} username {$oU} NULL {$ru} rU: {sip:domain.com;transport=UDP and ended up doing this which works $uac_req(furi)="sip:" + $fU + "@" + $var(rip); Any idea what could be wrong? Thanks, Daniel

2 2

Kamailio books
by Kevin Olbrich 20 Mar '19

20 Mar '19

Hi! Are there any Kamailio books? I am still struggling with Kamailio on scenarios that need more then REGISTER-offloading (like Kamailio cluster with BLF, custom NOTIFY, DMQ, etc.). Online docs are very limited or use Kamailio as a single tenant PBX. What I need is a high available and high performance proxy in front of asterisk, capable of handling thousands of tenants (each with BLF working). I am perfectly fine with building a solution for several months but I am still unable to understand everything from the docs. Where do the PROs and service companies get their knowledge from? Kind regards Kevin

4 4

Kamailio crashing and infinite-looping when trying to free memory
by Enrico Bandiera 19 Mar '19

19 Mar '19

Hi guys, recently we had some issues with kamailio in production, a proxy at the edge of the infrastructure was crashing seemingly randomly, it happened a couple of times in a timespan of 4-5 days until a point where we found it in an endless loop, unable to process SIP packets anymore and printing in loop messages like these: 50(60) CRITICAL: <core> [core/mem/q_malloc.c:512]: qm_free(): BUG: freeing already freed pointer (0x7ff40a359450), called from core: core/usr_avp.c: destroy_avp_list_unsafe(626), first free core: core/usr_avp.c: destroy_avp_list_unsafe(626) - ignoring 50(60) CRITICAL: <core> [core/mem/q_malloc.c:512]: qm_free(): BUG: freeing already freed pointer (0x7ff409c4fa58), called from core: core/usr_avp.c: destroy_avp_list_unsafe(626), first free core: core/usr_avp.c: destroy_avp_list_unsafe(626) - ignoring We managed to replicate the crash and the loop in a dev environment with a lot of difficulties. It seems not related to the peak load, but to the growing number of processed calls The kamailio version is: version: kamailio 5.2.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 5.3.1 Every time this happened it was the timer pid that was crashing or looping, we managed to get 2 core dumps of the crash and 1 of the loop (got it after killing the timer with SIGQUIT). The timer process crashes after attempting to free memory that it's not correctly managed. In particular we saw that it happens inside destroy_avp_list_unsafe() called by the tm timer, when uri_avps_from points to an invalid memory address. We don't use dialog, uac, http_async_client, but we do use topos. In the crash dumps it seems that there's something wrong inside free_cell_helper(), when trying to clean the dead_cell something goes wrong while freeing uri_avps_from, here's the crash log: 61(76) CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 66 0(15) ALERT: <core> [main.c:740]: handle_sigs(): child process 65 exited by a signal 11 0(15) ALERT: <core> [main.c:743]: handle_sigs(): core was generated 0(15) INFO: <core> [main.c:766]: handle_sigs(): terminating due to SIGCHLD <childs getting the SIGTERM signal> 0(15) CRITICAL: <core> [core/mem/q_malloc.c:512]: qm_free(): BUG: freeing already freed pointer (0x7f4d60a2e168), called from htable: ht_api.c: ht_cell_free(217), first free core: core/usr_avp.c: destroy_avp_list_unsafe(626) - ignoring 0(15) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Before the crash we have no warnings on errors about shared memory allocation backtrace: (gdb) bt full #0 0x00000000006371d7 in destroy_avp_list_unsafe (list=0x7f34f7a411b0) at core/usr_avp.c:625 avp = 0x2 foo = 0x2 __func__ = "destroy_avp_list_unsafe" #1 0x00007f351c6909ab in free_cell_helper (dead_cell=0x7f34f7a41018, silent=0, fname=0x7f351c796a02 "timer.c", fline=689) at h_table.c:260 b = 0x0 i = 1 rpl = 0x0 tt = 0x0 foo = 0x11c68daf2 cbs = 0x0 cbs_tmp = 0x7f34f80c3790 __func__ = "free_cell_helper" #2 0x00007f351c7291e6 in wait_handler (ti=1984410416, wait_tl=0x7f34f7a410a0, data=0x7f34f7a41018) at timer.c:689 p_cell = 0x7f34f7a41018 ret = 1 unlinked = 0 rcount = 1 __func__ = "wait_handler" #3 0x00000000004d0f56 in timer_list_expire (t=1984410416, h=0x7f34f6be7850, slow_l=0x7f34f6bea368, slow_mark=2702) at core/timer.c:874 tl = 0x7f34f7a410a0 ret = 0 #4 0x00000000004d13bd in timer_handler () at core/timer.c:939 saved_ticks = 1984410416 run_slow_timer = 0 i = 654 __func__ = "timer_handler" #5 0x00000000004d1845 in timer_main () at core/timer.c:978 No locals. #6 0x0000000000425cc8 in main_loop () at main.c:1727 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=127.0.0.1:5060 \000\000\000\000q\000\000\000@6y\000\000\000\000\000\000\b\206|\210X\322:\032\301\255\000\000\000\000\000\000\000\000 \000\000\000\000\000\000\000\002\000\000\000\000\063\275{\000\000\000\000\000>", '\000' <repeats 15 times>, "\320\220\377\226\374\177\000\000\351\022a\000\000\000\000" nrprocs = 8 woneinit = 1 __func__ = "main_loop" #7 0x000000000042ca76 in main (argc=10, argv=0x7ffc96ff9398) at main.c:2696 cfg_stream = 0x25ca010 c = -1 r = 0 tmp = 0x7ffc96ffa4c9 "" tmp_len = -1761635728 port = 32764 proto = -1761635632 options = 0x76c598 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 1492539644 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 2 n_lst = 0x7f351f6f6718 p = 0xffffffff <error: Cannot access memory at address 0xffffffff> st = {st_dev = 2049, st_ino = 1282695, st_nlink = 2, st_mode = 16895, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 4096, st_blksize = 4096, st_blocks = 8, st_atim = {tv_sec = 1552931478, tv_nsec = 972761643}, st_mtim = {tv_sec = 1552931478, tv_nsec = 972761643}, st_ctim = {tv_sec = 1552931478, tv_nsec = 972761643}, __glibc_reserved = {0, 0, 0}} __func__ = "main" In the loop dump we have a different situation, there's a loop in the uri_avps_from structure as you can see from the next pointers: (gdb) p *dead_cell->uri_avps_from $3 = {id = 66, flags = 275, next = 0x7ff40a359450, d = {p = 0x7ff40a9ad540, l = 140686126667072, data = "@\325\232\n\364\177\000"}} (gdb) p *dead_cell->uri_avps_from.next $4 = {id = 66, flags = 275, next = 0x7ff409c4fa58, d = {p = 0x7ff40a359480, l = 140686120031360, data = "\200\224\065\n\364\177\000"}} (gdb) p *dead_cell->uri_avps_from.next.next $5 = {id = 66, flags = 275, next = 0x7ff40a359450, d = {p = 0x7ff409c4fa88, l = 140686112651912, data = "\210\372\304\t\364\177\000"}} backtrace: (gdb) bt full #0 0x00007ff43152e67e in _IO_default_xsputn () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007ff43150150b in vfprintf () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x00007ff431502ef1 in ?? () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #3 0x00007ff43150032d in vfprintf () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #4 0x00007ff4315087f7 in fprintf () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000686e43 in qm_free (qmp=0x7ff409336000, p=0x7ff409c4fa58, file=0x7e0a75 "core: core/usr_avp.c", func=0x7e2d70 <__func__.8439> "destroy_avp_list_unsafe", line=626, mname=0x7e0a70 "core") at core/mem/q_malloc.c:512 __llevel = -3 qm = 0x7ff409336000 f = 0x7ff409c4fa20 size = 72 next = 0x0 prev = 0x0 __func__ = "qm_free" #6 0x0000000000637209 in destroy_avp_list_unsafe (list=0x7ff409e1b6c8) at core/usr_avp.c:626 avp = 0x7ff40a359450 foo = 0x7ff409c4fa58 __func__ = "destroy_avp_list_unsafe" #7 0x00007ff42ee5d9ab in free_cell_helper (dead_cell=0x7ff409e1b530, silent=0, fname=0x7ff42ef63a02 "timer.c", fline=689) at h_table.c:260 b = 0x0 i = 1 rpl = 0x0 tt = 0x0 foo = 0x12ee5aaf2 cbs = 0x0 cbs_tmp = 0x7ff409c5d420 __func__ = "free_cell_helper" #8 0x00007ff42eef61e6 in wait_handler (ti=89094309, wait_tl=0x7ff409e1b5b8, data=0x7ff409e1b530) at timer.c:689 p_cell = 0x7ff409e1b530 ret = 1 unlinked = 0 rcount = 1 __func__ = "wait_handler" #9 0x00000000004d0f56 in timer_list_expire (t=89094309, h=0x7ff4093b4850, slow_l=0x7ff4093b8058, slow_mark=5981) at core/timer.c:874 tl = 0x7ff409e1b5b8 ret = 0 #10 0x00000000004d13bd in timer_handler () at core/timer.c:939 saved_ticks = 89094309 run_slow_timer = 0 i = 861 __func__ = "timer_handler" #11 0x00000000004d1845 in timer_main () at core/timer.c:978 No locals. #12 0x0000000000425cc8 in main_loop () at main.c:1727 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=127.0.0.1:5060 \000\000\000\000q\000\000\000@6y\000\000\000\000\000\000\032 t\220\260\242\024\032\301\255\000\000\000\000\000\000\000\000 \000\000\000\000\000\000\000\002\000\000\000\000\063\275{\000\000\000\000\000>", '\000' <repeats 15 times>, "\220\306\336\071\377\177\000\000\351\022a\000\000\000\000" nrprocs = 8 woneinit = 1 __func__ = "main_loop" #13 0x000000000042ca76 in main (argc=10, argv=0x7fff39dec958) at main.c:2696 cfg_stream = 0x2292010 c = -1 r = 0 tmp = 0x7fff39dee4c9 "" tmp_len = 970901552 port = 32767 proto = 970901648 options = 0x76c598 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 2710999419 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 2 n_lst = 0x7ff431ec3718 p = 0xffffffff <error: Cannot access memory at address 0xffffffff> st = {st_dev = 2049, st_ino = 1282695, st_nlink = 2, st_mode = 16895, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 4096, st_blksize = 4096, st_blocks = 8, st_atim = {tv_sec = 1552931523, tv_nsec = 918435869}, st_mtim = {tv_sec = 1552994274, tv_nsec = 814438687}, st_ctim = {tv_sec = 1552994274, tv_nsec = 814438687}, __glibc_reserved = {0, 0, 0}} __func__ = "main" We are not sure on how to keep debugging this issue which is causing some serious troubles in our environment, any help is appreciated Here you can find the dumps (beware, 12M download but 1.5G once uncompressed): https://drive.google.com/file/d/1U0LZedyona8jJZHq6HaoqXpfnU83p6Sp/view?usp=… Due thanks to Giacomo Vacca who helped us to dig into this issue. Thanks

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Kamailio 5.1.7 crash
by Jon Bonilla (Manwe) 18 Mar '19

18 Mar '19

Hi all A customer of mine has some crashes every week. We installed the dbg and finally generated a core. Bt full attached. Kamailio installed from kamailio deb 5.1 repos kamailio -V version: kamailio 5.1.7 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 Any hints? thanks, Jon -- https://pekepbx.com https://www.issabel.com/multitenant

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Kamailio as Peer Trunk in Asterisk
by Isravel Raja 18 Mar '19

18 Mar '19

Hi, I wanted to achieve the following scenario with Kamailio and Asterisk/FreeSwitch Kamailio as a trunk for Receiving incoming calls and forward to Asterisk and Make outgoing calls. My current working setup like this VoIP Provider -> Kamailio -> Asterisk Clusters Where Kamailio(NAT for public) and Asterisk in the same local network, So I had to do only one thing just create the SIP trunk in Asterisk for authenticating all the Kamailio Incoming calls and making outgoing calls like this [kamailio] host=172.18.1.1 type=friend disallow=all allow=ulaw context=trunkinbound This is IP based authentication, This is working fine since both servers are in the same local network or having the public IP's What I want is, I want to register the Kamailio Trunk in an asterisk with Username Password authentication, My current setup for making that, VoIP Provider -> Kamailio(Public) -> Asterisk(Randomly Changing Public IP, Broadband Connection) I want to register Kamailio SIP in asterisk trunk so that I can forward the incoming calls to asterisk via SIP extension [kamailio] host=172.18.1.1 user=user secret=password type=friend disallow=all allow=ulaw context=trunkinbound Let me know if anyone doesn't understand my concern This e-mail message and any documents attached to it are confidential and may contain information that is protected from disclosure by various federal and state laws, including the HIPAA privacy rule (45 C.F.R., Part 164). This information is intended to be used solely by the entity or individual to whom this message is addressed. If you are not the intended recipient, be advised that any use, dissemination, forwarding, printing, or copying of this message without the sender's written permission is strictly prohibited and may be unlawful. Accordingly, if you have received this message in error, please notify the sender immediately by return e-mail or call (insert phone number), and then delete this message.

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RFC: rating - ranking system for community
by Daniel-Constantin Mierla 18 Mar '19

18 Mar '19

Hello, starting to continue the discussions here on mailing lists about some of the approached topics during the last IRC meeting -- there will be couple of them. The one now is about finding and deploying a ranking/rating system that could eventually help community members to make decisions easier in regard to what components to use in their voip systems. The need was exemplified by user verticelo with the dificulty to decide what KEMI scripting language to use. While maintaining all the app_xyz are expected to be easy, at least I know that for those I created, the arguments were also from the perspective of the community. Like what others are using, so one can expect good assistance, hints and share of knowledge via community forums. This can be also extended to related tools, like what people use for shared IP high availability of kamailio, preferred database servers, ... This email is to ask if those that didn't participate to IRC devel meeting find such system useful and, if there is a positive feedback, is anyone aware of some OSS that we can deploy on kamailio.org for such purpose? It should be something that allows posting a topic (title and short description) along with a list of answers/options (each can be again like a title and short description) and provide a way to rate the options (like, thumbs up, starts, ...), eventually allowing also comments for each option. I guess it sounds a bit like stackoverflow ... Being users related, the email is sent only to sr-users, let's keep the discussion on this mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training - Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com

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Re: [SR-Users] RFC: roadmap to next major release - v5.3.0
by Ilie Soltanici 18 Mar '19

18 Mar '19

Daniel, Sounds good, would be nice to have some improvements on usrloc module, like requested in this topic: https://lists.kamailio.org/pipermail/sr-users/2019-January/104329.html Regards,

2 1

Which RTP addon for RFC4040 Clearmode passthrough support ?
by Olivier 15 Mar '19

15 Mar '19

Hello, I'm looking for a solution to integrate legacy devices to a SIP network. More precisely, I need to forward to and receive Clearmode RTP traffic (see [1]). 1. Do you know any Kamailio-compliant RTP engine (rtpproxy, rtp engine, ..) that support this on a Debian host ? 2. Suggestion ? [1] https://tools.ietf.org/html/rfc4040 Best regards

2 2

rtpengine force-relay
by Vitalii Aleksandrov 15 Mar '19

15 Mar '19

Hi there, Well it's mostly rtpengine question but didn't know where should I send it and probably the answer will be more or less useful for kamailio users. By default if an offer has ICE candidates rtpengine leaves them untouched and just inserts itself as an additional low priority candidate and also rewrites c= line for the callee without ICE support. Using ICE=force-relay rtpengine inserts itself as a "relay" candidate what is great but doesn't touch c= line and callee without ICE have no chance to send RTP to rtpengine. Is it possible somehow to make a third behavior like force-relay + rtpengine IP in c= param? It would allow any callee to send media to rtpengine regardless of ICE support.

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cdp avps issues in kamailio 5.2
by Pavel S 15 Mar '19

15 Mar '19

Hello Mojtava, > Did you use FHOSS or IMS_DIAMETER_SERVER module? No, I am using our own hss server. >Anyway what mandatory parameters do you mean exactly? According to RFC6733 section 5.3.1 - https://tools.ietf.org/html/rfc6733#section-5.3.1 The Capabilities-Exchange-Request (CER), indicated by the Command Code set to 257 and the Command Flags' 'R' bit set, is sent to exchange local capabilities. Upon detection of a transport failure, this message MUST NOT be sent to an alternate peer. When Diameter is run over SCTP [RFC4960] or DTLS/SCTP [RFC6083], which allow for connections to span multiple interfaces and multiple IP addresses, the Capabilities-Exchange-Request message MUST contain one Host-IP-Address AVP for each potential IP address that MAY be locally used when transmitting Diameter messages. Message Format <CER> ::= < Diameter Header: 257, REQ > { Origin-Host } { Origin-Realm } 1* { Host-IP-Address } { Vendor-Id } { Product-Name } [ Origin-State-Id ] * [ Supported-Vendor-Id ] * [ Auth-Application-Id ] * [ Inband-Security-Id ] * [ Acct-Application-Id ] * [ Vendor-Specific-Application-Id ] [ Firmware-Revision ] * [ AVP ] Kamailio 5.1 with loaded cdp, cdr_avp, ims_icsf modules and default configuration described here https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/icscf sends Host-IP-Address but Kamailio 5.2 with the same configuration doesn't. Any idea how to fix it? Regards, Pavel Siderov >Hello, >The CER/CEA messages are used just for connections between two peers >diameter interface. Did you use FHOSS or IMS_DIAMETER_SERVER module? In >both cases, you should not have any issue. Anyway what mandatory parameters >do you mean exactly? >WIth Best Regards. Mojtava ><https://www.kamailio.org/docs/modules/5.1.x/modules/ims_diameter_server.html> >On Tue, Mar 12, 2019 at 8:17 PM Pafel <pafels at gmail.com <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> wrote: > >>* Hello, *>>>>* I am testing ims setup with kamailio 5.2 using the default configuration *>>* provided here: *>>* https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/ <https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/> . *>>* Тhere is an issue I am facing with cdp avp in request sent to hss. There *>>* are some missing mandatory parameters in capabilities exchange request like *>*> 257 host-ip-address for example. Any idea how to add it via icscf.xml or *>*> kamailio.cfg ? The same parameter is passed to hss with kamailio 5.1 using *>*> default ims configuration. I read cdp and cdp_avp documentation but *>*> couldn't understand if it is possible to add it. *>>>>* Regards, *>>* Pavel Siderov *>>* _______________________________________________ *>>* Kamailio (SER) - Users Mailing List *>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> *>*> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> *>>

4 3

Siremis on Kamailio 5.2
by Joel Serrano 15 Mar '19

15 Mar '19

Hi, Just curious if siremis works would work on Kamailio 5.2? The webside might not be updated: Installation Siremis installation is straightforward, being done mostly through a web wizzard. In just a few steps and less than 10 minutes you have your Siremis web site set up and ready to grant you easy administration of your Kamailio SIP server. You need to install first a compatible version of Kamailio SIP Server <https://www.kamailio.org/>. Siremis v5.1.x (devel) *Last release: v5.1.0 (devel version)* *Compatible with: Kamailio 5.1.x, 5.0.x* Thanks! Joel.

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