- sr-users - kamailio.org

Announcement: Kamailio is now systemd-rtc-server
by Alex Balashov 03 Aug '15

03 Aug '15

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina,was quick to summarise the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Check if user is registered before reply is routed
by Dmytro Bogovych 01 Aug '15

01 Aug '15

Greetings. Sorry for previous false letter. I want to check if user is registered before reply is routed to its target user agent. Imho it should be like this: onreply_route[MANAGE_REPLY] { if ($rm=="INVITE") { if (registered("location", $ru)) { # Do job - pause for 15 seconds and give chance to reregister } } } But kamailio complains on registered function 0(6274) ERROR: <core> [cfg.y:3295]: yyparse(): misused command registered 0(6274) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 1121, column 35: Command cannot be used in the block Documentation confirms - registered() function can be used in REQUEST_ROUTE and FAILURE_ROUTE only. Is it possible to check if user is registered when reply is processed in scripts? Thank you!

4 3

Interpretation of Contact header
by Joachim Büchse 30 Jul '15

30 Jul '15

Good day, I’m experiencing some problems with our VoiP providers handling of REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done some wire tracing and to me it seems that the providers configuration is to blame, but then - there are many RFCs out there and many NAT and UAC bug workarounds. Anyway, I wanted to get the opinion of “the" experts about how the requests send to the UAS SHOULD be correctly interpreted. The REGISTER requests/responses look like this (outside of the firewall): Protocol TCP! client port 19091 <-> server port 5060 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Authorization: Digest username="90780408050", realm="pbx.peoplefone.ch", uri="sip:pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy", response="764f371a08d258157a249f8d1b852514" Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.6bda Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>;q=0;expires=180;received="sip:212.126.160.92:19091;transport=TCP" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 The ip:port the firewall is sending those requests from is ip 212.126.160.92 port 19091. So this does NOT match the port from the Contact header. For TCP this seems rather logical to me, as one cant be listening on a TCP port and use it for sending at the same time. The UAC closes this “register connection” with TCP FIN after the register, and so does the firewall. However unfortunately subsequent requests from the provider (ie UAS) come in on port 19091 (not port 6717 from the Contact header) and the firewall simply drops them. Observations: - the server does NOT include received=212.126.160.92 in the Via of the reponse. According to RFC3581 this is mandatory when rport is present in the request, so this is probably an error in the server. - the server does include received="sip:212.126.160.92:19091;transport=TCP” in the Contact of the response. I didnt see this in any RFC (which means nothing;-) but it could be an error. - after the client received the 200 OK it closes the TCP connection. - the server tries several times to re-contact the client (incoming TCP SYN). However not on port 6717 (defined in the Contact header) but on port 19091 (where the REGISTER came from). RFC3581 defines special behaviour when “rport” is defined in the request (i.e. response should go to the same port the request came from) - however it’s not so clear if this should apply to subsequent (INVITE/OPTIONS) requests from the server to the client. Those are strictly spoken not replies (or are they?). RFC5626 defines that a “proxy” should keep track of the flows over which it received a registration and send requests over the same flow. It is not clear if RFC5626 should be applied. The RFC5626 defines that a UAC includes an “ob” parameter in the Contact field if it whishes further requests over the same flow. Also the RFC mandates a client to add a "reg-id=x" in the Contact field. Both are not the case here, so in short I think RFC5626 should NOT be applied. In which case conecting to the originating port (instead of the Contact port) would be a server error. So in short and if I interpret the RFCs correctly, the client is reachable and should be contacted on Transport: TCP IP: 212.126.160.92 Port: 6717 If anyone who lives and breathes SIP could enlighten me if the UAS is right to call back on 19091 instead of 6717 I would really appreciate it;-)) Best regards, Joachim

3 4

Issue with ACK behind NAT
by Alberto Sagredo 30 Jul '15

30 Jul '15

I have Kamailio in a local network with RTPPROXY and NAT MANAGE I have seen an issue with ACK as show when Asterisk sends ACK to Kamailio, it sends to advertised Address instead to Kamailio IP address. What could i add to solve this? route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails dlg_manage(); } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } BR

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Re: [SR-Users] Memory Issue while accessing $rb (prakash)
by Surendra.P 30 Jul '15

30 Jul '15

Hi Prakash, Specify the more private memory with -M option, while running kamailio Like -M 64 :-) Regards @surendra

1 0

Planning to release v4.2.6
by Daniel-Constantin Mierla 30 Jul '15

30 Jul '15

Hello, I am planning to release v4.2.6 later this week, most likely on Thursday, July 30. If there is anything not reported that should be considered for this minor release of the branch 4.2, write to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Carrierroute and dispatcher
by Nelson Migliaro 29 Jul '15

29 Jul '15

Hello everybody, I am currently using Kamailio in order to separate traffic based on dialed number. Some traffic goes to several Asterisk in a load balance strategy and other traffic goes to other SIP devices. I use carrierroute module in order to route traffic based on destination. I use scan_prefix to classify traffic and rewrite_host to route the traffic. A destination number usually go to one of 4 asterisk server. I do this in order to load balance the traffic. In order to distribute traffic I use prob (0.25 for each destination). Using this configuration, I have to include on carrieroute table a row for each destination node. Since the amount of destination is huge, I have to repeat configuration in tables. It also complicates maintenance. I was thinking of using a combination of carrierroute and dispatcher module. Carrierroute for classify traffic, strip numbers, and rewrite destination host and dispatcher to distribute the traffic over asterisk. Using this configuration I have to indicate asterisk nodes once in dispatcher table once. Also in carrierroute table I have to indicate each prefix once. I was thinking in something like this. if(!cr_route("carrier", "domain", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); drop; } if(!ds_select_dst("1", "4")) { sl_send_reply("403", "Not allowed"); drop; } t_relay; Does it make sense, or am I complicating everything and there is a magic way to achieve this. Thank you.

2 1

Re: [SR-Users] Memory Issue while accessing $rb
by Daniel-Constantin Mierla 29 Jul '15

29 Jul '15

Hello, you have to increase the private memory -- see -M command line parameter -- as well as pv_buffer_size core parameter -- see core cookbook in the wiki. Cheers, Daniel On 29/07/15 13:37, prakash wrote: > > Hello, > > > > I am getting this Error: > > > > *ERROR: <core> [rvalue.c:1706]: rval_str_add2(): rv eval out of memory* > > *ERROR: <core> [lvalue.c:389]: lval_assign(): rval expression > evaluation failed (881,17-881,29)* > > > > When I tried access *$rb* when sending large amount of data(*nearly > 4MB*) as a XML body. > > How can I sort out this issue? Is there anything to be configured > properly? > > > > NOTE: > > Version that I’m using kamailio 4.1.5 > > > > Thanks and Regards, > > Prakash P > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Kamailio no audio
by goReeves Admin 29 Jul '15

29 Jul '15

Hi - I just installed Kamailio 4.3.0 on Ubuntu 14.04.2. I can connect using jitsi on windows and csipcimple on the android and do text messaging. However, when I try to do audio callis it seems that the mic is working but the audio out is not functioning. I used the tutorial to do the install so everything in the config files is pretty vanilla except for the database name, use and password. I also installed siremis and added my sip.xxxxx.com as a gateway. On the Ubuntu image I use VPN to a dedicated IP address on which all ports are open. I run web and cloud servers on this machine as well and they all work just fine. I am pretty new to this SIP stuff so it is probably my issue. Here is the network configuration: root@xxxxxxxx-VM:/var/www/siremis-4.2.0# ifconfig eth0 Link encap:Ethernet HWaddr xx:xx:xx:xx:xx:xx inet addr:192.168.x.x Bcast:192.168.x.255 Mask:255.255.255.0 inet6 addr: fe80::20c:29ff:fe84:447a/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1579455 errors:0 dropped:0 overruns:0 frame:0 TX packets:1444329 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:346579445 (346.5 MB) TX bytes:625010061 (625.0 MB) lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:65536 Metric:1 RX packets:98232 errors:0 dropped:0 overruns:0 frame:0 TX packets:98232 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:262176392 (262.1 MB) TX bytes:262176392 (262.1 MB) ppp0 Link encap:Point-to-Point Protocol inet addr:xxx.xxx.xxx.xxx P-t-P:xxx.xxx.xxx.xxx Mask:255.255.255.255 UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1400 Metric:1 RX packets:1105547 errors:0 dropped:0 overruns:0 frame:0 TX packets:1228003 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:3 RX bytes:184558026 (184.5 MB) TX bytes:235399612 (235.3 MB) -- Web Administrator <mail:webadmin@goreeves.net> goreeves <https://www.goreeves.net>

2 2

Memory Issue while accessing $rb
by prakash 29 Jul '15

29 Jul '15

Hello, I am getting this Error: ERROR: <core> [rvalue.c:1706]: rval_str_add2(): rv eval out of memory ERROR: <core> [lvalue.c:389]: lval_assign(): rval expression evaluation failed (881,17-881,29) When I tried access $rb when sending large amount of data(nearly 4MB) as a XML body. How can I sort out this issue? Is there anything to be configured properly? NOTE: Version that I'm using kamailio 4.1.5 Thanks and Regards, Prakash P

1 0

double rr and $fs
by Asgaroth 29 Jul '15

29 Jul '15

Hi All, I have kamailio setup and listening on logical interfaces (for failover purposes) and therefore need to force the sending socket on initial messages. We are testing sips+tls at the moment and what we are trying to achive is that the TLS connection occurs from the UAC to the edge proxy (also responsible for the domain) and then force UDP for internal communications to services. What I am comming up against is that when I force the sending socket on the internal interface to UDP, the record route header still shows up as a "sips" request. Then, at the presense server, when generating the NOTIFY it attempts to send it to the proxy using TLS. My understanding was that the record route would set the uri something like "sip:internal_ip" when forcing the sending socket to be "udp:internal_ip". Is my understanding correct, or am I doing something wrong here? An example subscribe message follows, the top most record-route header, from my understanding, should read sip:internal_ip, not sips:internal_ip. Is this the expected behaviour of double rr with these types of messages? How could I ensure that the top most record-route would always be a sip uri (never sips). SUBSCRIBE sips:subscriber@domain.com:5061 SIP/2.0. Record-Route: <sips:proxy_internal_ip;r2=on;lr;ftag=594055226>. Record-Route: <sips:proxy_external_ip:5061;transport=tls;r2=on;lr;ftag=594055226>. Max-Breadth: 60. Via: SIP/2.0/UDP proxy_internal_ip;branch=z9hG4bK6c9.d7dc4801e872ce9fb5730f9e09f1889e.0;i=1. Via: SIP/2.0/TLS 172.16.0.121:5060;rport=33273;received=78.143.152.30;branch=z9hG4bK708ea7ba. Max-Forwards: 69. Call-ID: 17212ff4-4421321c8feafd63bf800080f0808080@KX-HDV230X. From: <sips:subscriber@domain.com>;tag=594055226. To: <sips:subscriber@domain.com>. CSeq: 2 SUBSCRIBE. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER. Accept: application/dialog-info+xml,application/rlmi+xml,multipart/related. Contact: <sips:subscriber@172.16.0.121:5060;alias=78.143.152.30~33273~3>. Expires: 3600. Event: dialog. Any thoughts, tips, tricks would be greatly appreciated. Cheers

2 2

Kamailio - Missing INVITEs
by Ali Taher 28 Jul '15

28 Jul '15

Hello, I connected Kamailio to a postgresql database (through sqlops module) and fetching result from a table containing around 300K records , where time processing of the single query is taking around 12 ms. I noticed that on huge load (around 200 - 300 calls per second) Kamailio is not processing all INVITE packets. For instance , I initialized a counter in Kamailio that increments on every INVITE , and on the other hand I counted the number of INVITEs found in Wireshark. I got a big difference (from 30K INVITES in Kamailio , I got 15K in Kamailio), though no errors were raised in kamailio log. How can I proceed in order to detect the problem? :( Any help would be appreciated. Regards, Ali

2 1

RR module - Fail detecting strict routing
by Antonio Reale 28 Jul '15

28 Jul '15

Hi all, I have the following scenario: U1 (caller) ---> P1 (192.168.0.245, kamailio 4.3, loose-router) ----> P2 ----> (192.168.0.101, strict router) ----> .... ----> U2 (called) When U2 answers the call, at P1 arrives the 200 OK with: Record-Route: <sip:0000000011@192.168.0.101;transport=udp;pgw-call=call-28d2e> Record-Route: <sip:192.168.0.245;lr=on> The problem is that the ACK from U1 is forwarded from kamailio to P2, as if P2 is a loose router. P2 drops the ACK message. From kamailio logs I see: Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:88]: is_preloaded(): is_preloaded: No Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:783]: after_loose(): Topmost route URI: 'sip:192.168.0.245;lr=on' is me Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:869]: after_loose(): URI to be processed: 'sip:0000000011@192.168.0.101;transport=udp;pgw-call=call-289ce' Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:878]: after_loose(): Next URI is a loose router Why after_loose function considers the next URI a loose router? The Route HF in the ACK with the URI of the next hop does not contain ;lr . It seems that the function is_strict fails detecting the strict router. Thanks. Regards. Antonio

2 7

CNXCC lock
by Kelvin Chua 28 Jul '15

28 Jul '15

after a call exhausted credits, the next call will fail with a timeout. found out it gets stuck in cnxcc_mod.c line 1137 static call_t *__alloc_new_call_by_money(credit_data_t *credit_data, struct sip_msg *msg, double credit, double cost_per_second, int initial_pulse, int final_pulse) { call_t *call = NULL; lock_get(&credit_data->lock); ... the next lines will never be executed Kelvin Chua

2 2

SIPCapture and Message Queues
by Grant Bagdasarian 27 Jul '15

27 Jul '15

Hello, Would it be possible to send the messages captured by the sipcapture module to a message queue instead of inserted directly into a database? Be it DMQ or SQL Server Service Broker Queues, etc. Or maybe another approach, executing a stored procedure instead of an insert statement? I couldn't find a way to accomplish this, but maybe someone has a brilliant idea how to accomplish any of the above. Regards, Grant

2 2

Problem Storing USRLOC Additional Attributes
by Luca Mularoni 27 Jul '15

27 Jul '15

Hi All, I'm using kamailio 4.2.5 (compiled from source and aligned up to commit [00a36ae07c587992d6486277ce2e9ae6c009685f]). In order to enrich users data collected during registering phase, I would like to store them in USRLOC Additional Attributes through the use of 'xavp_contact' structure, but I'm having problems storing them. Below relevant chunks of config: ... # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("usrloc", "xavp_contact", "ulvals") ... route[REGISTRAR] { ... # Just an example $xavp(ulvals=>A) = "A"; $xavp(ulvals[0]=>B) = "B"; $xavp(ulvals[0]=>C) = "C"; $xavp(ulvals[0]=>D) = "D"; ... if (!save("location")) sl_reply_error(); exit; } route[LOCATION] { ... if(!lookup("location") { # Handle Failure ... } # Print USRLOC Additional Attributes # Just an Example $var(i) = 0; $var(N) = $branch(count); while($var(i)<=$var(N)) { xlog("LOC - [$var(i)] - A [$xavp(ulvals[$var(i)]=>A)]\n"); xlog("LOC - [$var(i)] - B [$xavp(ulvals[$var(i)]=>B)]\n"); xlog("LOC - [$var(i)] - C [$xavp(ulvals[$var(i)]=>C)]\n"); xlog("LOC - [$var(i)] - D [$xavp(ulvals[$var(i)]=>D)]\n"); $var(i) = $var(i) + 1; } ... } And this is the relevant chunk of syslog when a local subscriber is found in 'location' table: ... LOC - [0] - A [A] LOC - [0] - B [<null>] LOC - [0] - C [<null>] LOC - [0] - D [D] ... Summing up, it seems that when xavp structure is stored in memory, just first and last elementsare inserted. In real config I'm using more than 4 attributes and behaviour is the same: just first and last are stored. I've tried also with modparam("usrloc", "db_mode", 0) but the problem remains. Where I'm wrong in using this structure? Or it could be a bug? Thank You in advance for your help Luca Mularoni

2 11

Check if user is registered after receiving reply
by Dmytro Bogovych 27 Jul '15

27 Jul '15

Greetings. I want to check if user is registered before kamailio will try to route reply to its user agent. IMHO it should be: onreply_route[MANAGE_REPLY] {

2 1

Re: [SR-Users] FROM number based on destination number
by Keith 27 Jul '15

27 Jul '15

Hi, I have a requirement to change the FROM number to a different number depending on the destination number. For example if I dial 01789 from 0800 I want to change the 0800 from number to a 01789 number. Ideally I want to be able to do a db lookup on the destination number to then set the from number. Thanks Keith

1 0

Changing acc_cdrs time and duration columns from varchar to int/float
by mayamatakeshi 27 Jul '15

27 Jul '15

Hello, I am planning to enable acc cdr in my kamailio 4.1 server. I am testing in my lab and it is working fine. However, when I check the table definition I see this: mysql> describe acc_cdrs; +------------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +------------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | start_time | varchar(32) | NO | MUL | | | | end_time | varchar(32) | NO | | | | | duration | varchar(32) | NO | | | | +------------+------------------+------+-----+---------+----------------+ 4 rows in set (0.00 sec) The *_time and duration columns are defined as varchar. Because of this, i need to perform a cast when doing queries: select * from acc_cdrs where cast(start_time as unsigned) >= 1437120622; which would cause extra processing in the db server. So I experimented changing these columns to "int unsigned" and to "float unsigned" and it worked for both without problems (records properly added to acc_cdrs and no errors in kamailio.log): alter table acc_cdrs modify start_time int unsigned; alter table acc_cdrs modify end_time int unsigned; alter table acc_cdrs modify duration int unsigned; alter table acc_cdrs modify start_time float unsigned; alter table acc_cdrs modify end_time float unsigned; alter table acc_cdrs modify duration float unsigned; So I am thinking in doing this in my production server but I would like to know if there is any potential risk of doing this (as I am not sure why it was decided for these fields to be varchar instead of a numeric type). Regards, Takeshi

2 3

New SIP INVITE from UAS (new dialogue)
by Joao Alves 27 Jul '15

27 Jul '15

Hi, What would be the best way to generate and send an INVITE message generated by the Kamailio server. This would be for a new dialog session, UAS initiated (e.g. to make a click 2 call like service). I've tried the $uac_req(method)="INVITE"; uac_req_send(); but didn't work. I gather something would be needed first to open the dialogue before doing it but couldn't find it on the dialogue module. Thanks, Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

4 10

Issues sending SIP UPDATE inside an existing dialogue.
by Joao Alves 24 Jul '15

24 Jul '15

Hi, I need to send an SIP UPDATE inside an existing dialogue (in order to update the SDP offer). I've been using the following uac functions from the command file. $uac_req(method)="UPDATE"; $uac_req(ruri)=$shv(userAcontact); $uac_req(turi)=$tu; $uac_req(furi)=$fu; $uac_req(callid)=$ci; $uac_req(hdrs)="Content-Type: application/sdp\r\n"; $uac_req(body)=$var(sdp); # uac_restore_from(); uac_req_send(); As a result, the SIP UPDATE is being sent but since the original From Tag is not being used (i.e. and a new tag is generated), the SIP client (sipML5) does not recognize it as part of the existing dialogue. Attach is a pcap of the test made. Subsequently, I've also tried to recover the from using uac_restore_from() function but got the following an error message. DEBUG: uac [replace.c:511]: restore_uri(): getting 'vsf' Route param DEBUG: uac [replace.c:515]: restore_uri(): route param 'vsf' not found Thanks, Joao Alves Solution Architect, Unified Communications +351 214094660 (desk) +351 912783702 (mobile) AMDOCS | EMBRACE CHALLENGE EXPERIENCE SUCCESS This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

1 0

Installation issues
by ngn zone 24 Jul '15

24 Jul '15

>Hello all, I am new to Kamailio. I Tried installing Kamailio 4.0 and I had the following error. I will appreciate any help >ERROR: could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL'

2 1

dialog keepalive timeout - which parameter rules it?
by Andrey Utkin 24 Jul '15

24 Jul '15

What happens if dialog parties don't respond to keepalive requests sent by dialog module? It looks it is not coded in modules/dialog, I guess it can be set in t_uac_prepare() from modules/tm/uac.c. Please explain what timeout or lifetime parameters rule - at which delay the unanswered keepalive request sent by Kamailio within dialog triggers dialog termination. Thanks a lot. -- Andrey Utkin

2 2

dmq_usrloc replication and max_contacts
by Asgaroth 24 Jul '15

24 Jul '15

Hi All, I have a situation where I have 2 devices registering with the same AOR, I have the registrar module's max_contacts parameter set to 1 and I use the 0x04 flag on the save function. When I use dmq_usrloc for replication, I can see 2 contacts registered for the AOR on the "client" nodes whereas the node where the actual registration took place only has 1 contact for the AOR. Is this expected behaviour of the dmq_usrloc module in terms of registration or have a come accross another bug? Thanks

2 5

"kamctl ul show"/xmlrpc ul.lookup AOR not found for valid registration
by Asgaroth 24 Jul '15

24 Jul '15

Hi All, I have a situation where a device has successfully registered and its location saved/stored in memory. I can see all the stored information when I issue a "kamctl ul show" However, if I perform a kamctl ul show subscriber@domain, where the 'subscriber@domain' is copied/pasted from the output of the "kamctl ul show" command, it results in an AOR not found error. However, with other AOR's on the same registrar, the "kamctl ul show subscriber@domain" works. I checked the usernames for the failed commands, and I dont see anything obvious, they are a combination of lower/upper-case leters and digits. String length is around 24 characters for the username (the username table for the subscriber is 64 characters). I'm not sure what elese to check to see why the AOR lookup for these particular clients are failing when they are indeed registered. All other functions around the lookup seem to work, for example, a location lookup for these AOR's are working. I wonder if I have come accross an issue with the kamctl function and/or the equivilient xmlrpc ul.lookup method. Any thoughts on this would be greatly appreciated. Thanks

1 3

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