- sr-users - kamailio.org

$HN(d) null when DNS cache is enabled
by Julien Klingenmeyer 13 Feb '19

13 Feb '19

Hello, I have a small question regarding the internal DNS resolver. Here is my current Kamailio DNS configuration: dns_try_naptr=on use_dns_failover=yes use_dns_cache=yes I need the $HN variable inside my script and it appeared while testing that $HN(d) was null (and consequently $HN(f) was only containing the hostname part of the FQDN). I know /etc/hosts file is bypassed when the DNS cache is activated, so I thought Kamailio was using the domain parameter set in the /etc/resolv.conf file in such cases in order to get the domain. But it was not. I figured out Kamailio correctly retrieves the domain name as long as it is explicitly part of the /etc/hostname file. Is my understanding correct or am I wrong somewhere? What if /etc/hostname does not contain the domain name but hostname only? Thanks! Cheers Julien

1 0

pv_buffer_size config value actually ignored?
by Enrico Bandiera 13 Feb '19

13 Feb '19

Hi, I'm having some strange issues with a somewhat big $avp value, when it reaches a size of about 4K I get this error: 29(47) ERROR: <core> [core/pvapi.c:1452]: pv_printf(): no more space for spec value 29(47) ERROR: <core> [core/pvapi.c:1461]: pv_printf(): buffer overflow – increase the buffer size... In the docs I read that the default pv_buffer_size is 8K, I still tried to enforce it to something bigger and checked the logs to be sure it was set: 0(384) DEBUG: <core> [core/pvapi.c:2062]: pv_init_buffer(): PV print buffer initialized to [40][16384] But still when reaching the 4K size inside the $avp that error comes out, is there any limit I'm not aware of? Thanks, Enrico.

2 1

memory issue with jsonrpc
by Benjamin Roy 12 Feb '19

12 Feb '19

Hi, I'm hoping someone can guide me in the right direction. In kamailio 5.2.1 I'm getting the following memory errors after a while (hours or days) of sending messages with jsonrpc. Restarting kamailio clears them for a while. Any suggestions on what I should change, or how to better identify the issue? 24(6177) ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7f1cbe6fd000, 5712); Free fragment not found! 24(6177) ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f1cbe6fd000, 5712) called from tm: h_table.c: build_cell(339), module: tm; Free fragment not found! 24(6177) ERROR: tm [uac.c:422]: t_uac_prepare(): short of cell shmem 24(6177) ERROR: pua [send_publish.c:700]: send_publish(): in t_request tm module function 24(6177) ERROR: pua_rpc [pua_rpc.c:280]: pua_rpc_publish_mode(): pua send_publish failed I have these buffer values in the config file tcp_rd_buf_size=65536 pv_buffer_size=32768 and I'm running kamailio with these memory options "-m 1024 -M 16" The jsonrpc message I'm sending look like this { "jsonrpc": "2.0", "method": "pua.publish", "params": [ "sip:172.1.1.3:5060;transport=tcp", "300", "reg", "application/reginfo+xml", ".", ".", ".", ".", "<?xml version='1.0' ?> <reginfo state='partial' version='0' xmlns='urn:ietf:params:xml:ns:reginfo' xmlns:xsi='http://www.w3.org/2001/XMLSchema-instance'> <registration aor='sip:55555@uw.edu' id='a19685' state='active'> <contact duration-registered='906' event='registered' id='c19685--1732979172--1829864152-2' q='1.0' state='active'> <uri>sips:55555@172.1.2.4:62270;transport=tls;avaya-sc-enabled</uri> <elin></elin> </contact> </registration> </reginfo> " ], "id": 1 } Thanks, -Ben

2 3

Programmatically modify SIP Requests/Responses in Kamailio
by Filippos Vasilakis 12 Feb '19

12 Feb '19

Hi, First of all I would like to congratulate and thank everyone who has worked in Kamailio. It looks like an awesome tool. I am starting a side project and I am not very familiar with Kamailio although I know SIP and VoIP in general, so I am more looking for answers that tell me if it's possible or not what I am asking along with some hints on how/where to look. Is there a way to filter/forward/modify(through the headers) incoming SIP requests and Kamailio SIP responses programmatically in Kamailio ? I have seen that you can run embedded code like Ruby, but I wonder if this is for simple things or more complex stuff which probably is in my case since there will be plenty of business logic to figure out what action to take with a SIP request. I have also found the EVAPI that is supposed to be handle events through TCP but not much more info available. So to sum up, is there a way to filter/forward/modify incoming SIP requests and create responses programmatically in Kamailio ? If yes, what's the proper way of doing that ? If no, any idea where I should look for that? Thanks, Filippos

2 1

Kamailio modules that can be used in Lua
by Leonid Fainshtein 12 Feb '19

12 Feb '19

Hello, I am checking a possibility to replace the traditional cfg script with a Lua script. In the app_lua module documentation there is a relatively short list of modules that can be registered to Lua. The modules nathelper, rtpproxy, topoh and many others don't appear in the list. Does it mean that it is not possible to use that modules in Lua? Best regards, Leonid Fainshtein Xorcom Ltd

2 1

Repository for automatic deployment of Kamailio
by Gholamreza Sabery 11 Feb '19

11 Feb '19

Hi, A few years ago, I developed an Ansible repository for automatic deployment of Kamailio. Recently, I added some new features, and now you can deploy the playbooks on Docker containers. I think it is a good starting point for anyone who wants to deploy and use Kamailio for the first time. Link of the repository: https://github.com/ghrst/Kamailio-HA Regards Gholamreza Sabery Tabrizy

2 1

Kamailio TOPOS fail
by Marcos Pytel 11 Feb '19

11 Feb '19

Hi, When I activate the TOPOS function, the Kamailio lb fail. This is the traceback Program terminated with signal SIGABRT, Aborted. #0 __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:51 51 ../sysdeps/unix/sysv/linux/raise.c: No such file or directory. (gdb) backtrace #0 __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:51 #1 0x00007f66d5aab42a in __GI_abort () at abort.c:89 #2 0x00007f66d5ae7c00 in __libc_message (do_abort=do_abort@entry=2, fmt=fmt@entry=0x7f66d5bdb305 "*** %s ***: %s terminated\n") at ../sysdeps/posix/libc_fatal.c:175 #3 0x00007f66d5b701f7 in __GI___fortify_fail (msg=msg@entry=0x7f66d5bdb29c "buffer overflow detected") at fortify_fail.c:30 #4 0x00007f66d5b6e330 in __GI___chk_fail () at chk_fail.c:28 #5 0x00007f66b867e097 in memcpy (__len=<optimized out>, __src=<optimized out>, __dest=0x7f66b8892524 <_tps_redis_cbuf+4>) at /usr/include/x86_64-linux-gnu/bits/string3.h:53 #6 tps_redis_load_branch (msg=<optimized out>, md=<optimized out>, sd=0x7ffed3993d30, mode=<optimized out>) at topos_redis_storage.c:744 #7 0x00007f66b88b35c0 in tps_request_received (msg=msg@entry=0x7ffed3995f80, dialog=dialog@entry=1) at tps_msg.c:786 #8 0x00007f66b88b9255 in tps_msg_received (evp=<optimized out>) at topos_mod.c:332 #9 0x000055e1dc07da88 in sr_event_exec (type=<optimized out>, evp=<optimized out>) at core/events.c:211 #10 0x000055e1dc0415f5 in receive_msg ( buf=buf@entry=0x55e1dc511d40 <buf> "BYE sip:atpsh-5c4b0f6e-25b2-9@10.10.10.2 SIP/2.0\r\nVia: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK898122669\r\nFrom: <sip:2972426338@10.10.10.1>;tag=3296734298\r\nTo: <sip:414533@10.10.10.2>;tag=03D2CCF0-5"..., len=381, rcv_info=rcv_info@entry=0x7ffed3996980) at core/receive.c:157 #11 0x000055e1dbf5e6e3 in udp_rcv_loop () at core/udp_server.c:554 #12 0x000055e1dbef3a9f in main_loop () at main.c:1619 #13 0x000055e1dbeea8cb in main (argc=<optimized out>, argv=0x7ffed3996f38) at main.c:2638 This is a Kamailio error? I can fix it? Thanks. Marcos.

4 6

reg_fetch_contacts and t_relay usage
by Володимир Іванець 11 Feb '19

11 Feb '19

Hello! I'm using *reg_fetch_contacts* function to get information about all contacts. This allows to define which contacts should receive the call. My question is if it's possible to choose only one contact before *t_relay* is called? Or *reg_fetch_contacts* call and verification should be done after *t_relay* in all branches and those which are not needed should be stopped? Thanks a lot!

2 2

rtpengine not listening on specified port range
by Narayan P 11 Feb '19

11 Feb '19

Hello, I am starting the rtpengine through this command,but rtpengine is not listening on the specified port range.It is listening on some other port outside of this.Here first IP is private IP and the second IP is public one.So issue is with audio.Can you please let me know what exactly I am missing. rtpengine --interface=172.20.xx.xxx\!34.209.1.xx -f -m 16384 -M 16884 -E --listen-ng=127.0.0.1:22222 & Any suggestion will be highly appreciated. Thanks, Narayan NP

2 2

trace sip msg topos
by Giuseppe Pandolfi 11 Feb '19

11 Feb '19

Hi All, is it possible print sip msg after stripping by topos module? I have need to print sip msg stripped by topos in syslog. Thanks

4 3

rtpengine not listening on specified port range
by Narayan P 11 Feb '19

11 Feb '19

Hello, I am starting the rtpengine through this command,but rtpengine is not listening on the specified port range.It is listening on some other port outside of this.Here first IP is private IP and the second IP is public one.So issue is with audio.Can you please let me know what exactly I am missing. rtpengine --interface=172.20.xx.xxx\!34.209.1.xx -f -m 16384 -M 16884 -E --listen-ng=127.0.0.1:22222 & Any suggestion will be highly appreciated. Thanks, Narayan

1 0

rat-proxy issue
by aroosa satti 10 Feb '19

10 Feb '19

Hi! I have issue regarding my rtpproxy.I have start it but it does't work and give error like Feb 9 02:46:50 ns1 ./kamailio[3682]: NOTICE: <core> [main.c:692]: handle_sigs(): Thank you for flying kamailio!!! Feb 9 02:46:53 ns1 ./kamailio[4318]: ERROR: <core> [main.c:1165]: parse_phostport(): bad protocol in unix:/var/run/rtpproxy.sock these are the issue I will be grateful for you if you solve this problem. kind regards Aroosa Satti

2 1

how to differentiate branches in parallel call forking
by Koray Vatansever 09 Feb '19

09 Feb '19

Hi, I'm trying to implement parallel call forking scenario. Since target clients have different media profiles, I need to set different rtpengine options for those forked calls. So according to the rtpengine module documentation, I set via-branch option on request and response, '1' and '2' respectively ('auto' option is not working. There is no code implementing this parameter). However all forked branches have the same via branch id which is the incoming INVITE's via branch id. As a result, rtpengine thinks that all the offers belong to the same session I guess. It seems I have only 'extra' option with extra_pv_id variable. In order to use it, I need an elegant way to differentiate forked branches in request and response processing. I would appreciate any suggestion.

1 1

rtpproxy issues
by aroosa satti 08 Feb '19

08 Feb '19

Hi! I have issue regarding my rtpproxy.I have start it but it does't work and give error like Feb 9 02:46:50 ns1 ./kamailio[3682]: NOTICE: <core> [main.c:692]: handle_sigs(): Thank you for flying kamailio!!! Feb 9 02:46:53 ns1 ./kamailio[4318]: ERROR: <core> [main.c:1165]: parse_phostport(): bad protocol in unix:/var/run/rtpproxy.sock these are the issue I will be grateful for you if you solve this problem. kind regards Aroosa Satti

1 0

Kamailio modules that can be used in Lua
by Leonid Fainshtein 08 Feb '19

08 Feb '19

Hello, I am checking a possibility to replace the traditional cfg script with a Lua script. In the app_lua module documentation there is a relatively short list of modules that can be registered to Lua. The modules nathelper, rtpproxy, topoh and many others don't appear in the list. Does it mean that it is not possible to use that modules in Lua? Best regards, Leonid Fainshtein

3 2

Waiting for calls
by Mark Hall 08 Feb '19

08 Feb '19

Hi, I am using Kamailio as a gateway for inbound calls to a group of Asterisk servers. One particular call scenario involves matching pairs of calls. Originally I was just sending the calls through to Asterisk for processing (including the 'matching up'), but there is a further requirement that the first received call must not be answered until its matching call arrives - and this is a little awkward in Asterisk, especially if there will be many calls. I thought of t_suspend/t_continue functionality in Kamailio. I have implemented this: CallA arrives. Send 180-ringing Use ds_select_dst() to choose an Asterisk to handle the paired calls Store destination and other data in an htable so that I can match the call later. Invoke t_suspend() for CallA. What I have implemented when the matching call arrives is this: CallB arrives (and some logic matches it to the suspended CallA). Invoke t_continue('id','label','CONTINUE_CALL'); so that CallA is now released and the route ends in t_relay() to send it on to Asterisk. Set $du on CallB to match the destination, then t_relay() this call to send it on to the same Asterisk. The solution almost works, but there is a problem. The t_relay() for CallA is fine, and the call is forwarded. When the script runs the t_relay() for CallB, I get this logged: ERROR: tm [t_fwd.c:1728]: t_forward_nonack(): no branches for forwarding ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (6/SL) AND after each call I notice an extra transaction stuck in Kamailio (when I use kamctl rpc tm.list). Can anyone point out where my Kamailio/SIP understanding has let me down? Many Thanks, Mark Hall

2 2

Re: [SR-Users] TLS challenge
by Gertjan Wolzak 07 Feb '19

07 Feb '19

Hello Kamailions, Julien, thank you for the help, I have added the letsencrypt ca certificate to the ca list, still no dice. So, still got lots of questions, but after my last booboo going to do some more research and testing. When I have no more hair left will get back to the list. Thanks for now. Rgds, Gertjan Wolzak

2 1

ERROR : EAGAIN and write queue full or failed for tcp Connection
by YASIN CANER 07 Feb '19

07 Feb '19

Hello, Kamailio gave this error that couldnt bind tcp connection other side. Thanks Yasin CANER Kamailio IPv6:TCP: -----> IPv6:TCP Kamailio -----> X ERROR XXXX IPv6:TCP Kamailio IPv6::C IPv6::8 IPv6::5 LOGS: 7(15781) DEBUG: <core> [core/msg_translator.c:2937]: create_via_hf(): id added: <;i=1>, rcv proto=2 7(15781) DEBUG: <core> [core/msg_translator.c:162]: check_via_address(): (FD61:C23C:B3B6:0:0:0:0:C, [FD61:C23C:B3B6:0:0:0:0:C], 0) 7(15781) DEBUG: <core> [core/tcp_main.c:1977]: tcp_send(): no open tcp connection found, opening new one 7(15781) DEBUG: <core> [core/ip_addr.c:242]: print_ip(): tcpconn_new: new tcp connection: fd61:c23c:b3b6:0:0:0:0:5 7(15781) DEBUG: <core> [core/tcp_main.c:1229]: tcpconn_new(): on port 5060, type 2 7(15781) DEBUG: <core> [core/tcp_main.c:1539]: tcpconn_add(): hashes: 507:2382:0, 2 7(15781) DEBUG: <core> [core/tcp_main.c:2873]: tcpconn_1st_send(): pending write on new connection 0x7fd15b5e7c98 (-1/1483 bytes written) (err: 11 - Resource temporarily unavailable) 7(15781) ERROR: <core> [core/tcp_main.c:2891]: tcpconn_1st_send(): [FD61:C23C:B3B6:0:0:0:0:5]:5060: EAGAIN and write queue full or failed for 0x7fd15b5e7c98 7(15781) DEBUG: <core> [core/tcp_main.c:3153]: tcpconn_chld_put(): destroying connection 0x7fd15b5e7c98 (2, -1) flags 0060 7(15781) ERROR: tm [../../core/forward.h:251]: msg_send_buffer(): tcp_send failed 7(15781) DEBUG: tm [t_fwd.c:1549]: t_send_branch(): send to FD61:C23C:B3B6:0:0:0:0:5:5060 (2) failed 7(15781) WARNING: tm [t_fwd.c:1567]: t_send_branch(): sending request on branch 0 failed CONF: dns_try_ipv6=yes dns_retr_time=1 dns_retr_no=1 dns_cache_flags=4 tcp_source_ipv6 = [fd61:c23c:b3b6::8] tcp_async=yes tcp_rd_buf_size=65536 tcp_wq_blk_size=3000 listen=tcp:[fd61:c23c:b3b6::8]:5060 tcp_connection_lifetime=3605 listen=tcp:[fd61:c23c:b3b6::8]:5060 ..... route[]{ remove_hf("Route"); #loose_route(); if (!is_method("INVITE")) return; $du="sip: mykamailio1.test:5060;transport=tcp"; if(!t_newtran()){ xlog("L_WARN"," Couldnt create new transaction \r\n"); sl_send_reply("503", "Service not available"); exit; } if(!t_relay()){ send_reply("503", "Service not available"); exit; } exit; }

4 7

Keep requests order
by Duarte Rocha 07 Feb '19

07 Feb '19

Greetings, I'm having some problems with keeping the order of the messages in Kamailio. In some cases i'm receiving an ACK for the first INVITE and immediately after that a RE-INVITE is sent by the client. What is happening is that the the ACK took double the time of the RE-INVITE to process and as a consequence, the RE-INVITE was relayed before the ACK. I have two questions : 1 - Is there a way to keep the order of the messages in a dialog, and if so, what are the downsides to it? 2 - The machine wasn't in a load state (only one concurrent call) and the ACK and the RE-INVITE should go roughly through the same processing methods and should take took the same time to be be relayed. What can be the reason for this? Best Regards, Duarte Rocha

4 7

Kamailio with RTPengine, allow source port of RTP to change during a call
by Laurent Schweizer 06 Feb '19

06 Feb '19

Hello, I'm using the RTPengine with Kamailio and I have a question for a specific case. I have some customer that are changing the source port of the RTP stream during the call ( no re-invite) I think it's more a NAT issue that a user agent issue... I see in the doc that I can handle this case with the "media handover" but in that case the rtpengine will allow a change of the source port but also of the source IP. Any possibility to allow only a change of the source port so source IP must still be the same ? Thanks BR Laurent

2 1

Kamailio failure routing to SRV Record
by Jesse Strahn 05 Feb '19

05 Feb '19

Hi all, I am trying to route requests from Kamailio to a private SRV record in AWS Route 53. If I setup the record as an A record and put the 3 destination sip server IPs in the record, Kamailio routes perfectly round-robinning between the 3 servers. However, If I switch my dispatcher.list over to the SRV record, the calls do not route and I get a 404 No Destination in my pcap. When I perform a `dig _sip._udp.pool.sip.servers` from the Kamailio server, it properly returns the hostnames of the 3 sip servers that the SRV record points to. I'm not sure if something in the Kamailio routing has to be changed to accommodate SRV record routing rather than A record? I have scoured the internet and can't really find much of anything. Would greatly appreciate any guidance. Thanks, Jesse

3 3

Error UAS
by Nicolas Breuer 05 Feb '19

05 Feb '19

Hello, How can we avoid these errors ? Feb 5 11:21:39 proxy1 /usr/sbin/kamailio[29563]: ERROR: tm [t_reply.c:1301]: t_should_relay_response(): status rewrite by UAS: stored: 408, received: 200 Thanks

2 1

TLS challenge
by Gertjan Wolzak 05 Feb '19

05 Feb '19

Hello Kamailions, Ive beent trying to configure a tls connection between kamailio servers, but looks like Im missing some knowledge. The situation: phone1 <> internet <> kamailio1 <internet> kamailio2<> internet <> phone2 The phones are yealink phones, they are connected using TLS v1.2, phone 1 registers at kamailio1 and phone2 registers at kamailio2. Both kamailio servers have an letsencrypt certificate. When the connection between the two kamailio servers is set as sip, a call from a phone1 to phone2 the singnaling is forwarded and rtp flows. When I set the connection between the two kamailio servers to tls, things go wrong.. or better said I go wrong. The kamailio log on the receiving kamailio(kamailio2) shows that things go wrong and where, but I just cant figure out how to solve it. The log shows that the tls connection will be handled by the TLS Default configuration, the CA certificate is missing, well unknown. 15(7209) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 116.203.53.212 15(7209) DEBUG: <core> [core/tcp_main.c:999]: tcpconn_new(): on port 44153, type 3 15(7209) DEBUG: <core> [core/tcp_main.c:1309]: tcpconn_add(): hashes: 1805:3331:3809, 3 15(7209) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xa77de0, 32, 2, 0x7ff9596d4b10), fd_no=23 15(7209) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa77de0, 32, -1, 0x0) fd_no=24 called 15(7209) DEBUG: <core> [core/tcp_main.c:4196]: handle_tcpconn_ev(): sending to child, events 1 15(7209) DEBUG: <core> [core/tcp_main.c:3878]: send2child(): selected tcp worker 1 8(7202) for activity on [tls:kamailio2:5061], 0x7ff9596d4b10 8(7202) DEBUG: <core> [core/tcp_read.c:1759]: handle_io(): received n=8 con=0x7ff9596d4b10, fd=9 8(7202) DEBUG: tls [tls_server.c:199]: tls_complete_init(): completing tls connection initialization 8(7202) DEBUG: tls [tls_server.c:228]: tls_complete_init(): Using initial TLS domain TLSs<default> (dom 0x7ff959498600 ctx 0x7ff9594b6430 sn []) 8(7202) DEBUG: tls [tls_domain.c:724]: sr_ssl_ctx_info_callback(): SSL handshake started 8(7202) DEBUG: <core> [core/tcp_main.c:2460]: tcpconn_do_send(): sending... 8(7202) DEBUG: <core> [core/tcp_main.c:2494]: tcpconn_do_send(): after real write: c= 0x7ff9596d4b10 n=2637 fd=9 8(7202) DEBUG: <core> [core/tcp_main.c:2495]: tcpconn_do_send(): buf= 8(7202) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xac78a0, 9, 2, 0x7ff9596d4b10), fd_no=1 8(7202) ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094418:SSL routines:ssl3_read_bytes:tlsv1 alert unknown ca 8(7202) ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7ff9596d4b10 r: 0x7ff9596d4b90 (-1) 8(7202) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xac78a0, 9, -1, 0x10) fd_no=2 called 8(7202) DEBUG: <core> [core/tcp_read.c:1683]: release_tcpconn(): releasing con 0x7ff9596d4b10, state -2, fd=9, id=3 (kamailio1]:44153 -> [kamailio1]:5061) 8(7202) DEBUG: <core> [core/tcp_read.c:1684]: release_tcpconn(): extra_data 0x7ff9596d2ab0 15(7209) DEBUG: <core> [core/tcp_main.c:3308]: handle_tcp_child(): reader response= 7ff9596d4b10, -2 from 1 15(7209) DEBUG: tls [tls_server.c:667]: tls_h_close(): Closing SSL connection 0x7ff9596d2ab0 I assume I have configured tls.cfg partially correct, as the phones register and can make tls calls. What I find strange is this line: 8(7202) DEBUG: <core> [core/tcp_read.c:1683]: release_tcpconn(): releasing con 0x7ff9596d4b10, state -2, fd=9, id=3 (kamailio1]:44153 -> [kamailio1]:5061) Would the "[kamailio1]:5061" be the client configuration? So, what do I need to configure within the tls.cfg file so when a tls connection is initiated from the other server it will work. I have tried adding the other server as server within the tls.cfg, which does not work as kamailio is not listening on the ip address configured. When I add the server as a client in the tls.cfg as follows, the connection is handled by the default TLS configuration. [server:default] method = TLSv1.2 verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/live/kamailio.kazlow.nl/privkey.pem certificate = /etc/letsencrypt/live/kamailio.kazlow.nl/fullchain.pem #ca_list = /etc/kamailio/tls/cacert.pem #crl = /etc/kamailio/tls/crl.pem [client:kamailio1:5061] method = TLSv1.2 verify_certificate = no require_certificate = yes certificate = /etc/kamailio/tls/fullchain.pem ca_list = /etc/kamailio/tls/cacert.pem # This is the default client domain, settings # in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] method = TLSv1.2 verify_certificate = no require_certificate = no ca_list = /etc/kamailio/tls/cacert.pem I have tried forcing the socket on the sending kamailio, but that does not work. Any feedback on what I am doing wrong or where I can find information to extend my knowledge would be appreciated. Rgds, Gertjan Wolzak

2 1

How to chain proxies with Route header and loose_route()
by Ivan Ribakov 05 Feb '19

05 Feb '19

Assuming there are several proxies that have to be hopped sequentially to deliver the request and the order is determined dynamically (!), how can one achieve that behaviour in Kamailio? My understanding was that Route header is used for that - it allows to keep R-URI intact but still route the egress message elsewhere. Documentation of the loose_route() from RR module seems to confirm that. But the catch is that loose_route() will not pick up any Route header that I might add in the config via append_hf() due to the delayed lump processing (as explained in https://sourceforge.net/p/openser/bugs/277/#5f42). Thanks in advance for any help or ideas. Regards, Ivan

2 1

TLS traffic visualization HOWTO
by Giovanni Maruzzelli 04 Feb '19

04 Feb '19

# How to TRACE and visualize TLS and non-TLS SIP traffic in real time (thanks to Homer's Lorenzo Mangani for pointing me toward Frida) apt-get install python-pip pip install frida pip install hexdump wget https://raw.githubusercontent.com/google/ssl_logger/master/ssl_logger.py #first ssh terminal # create fifo pipe, then will send the content from fifo pipe to an sngrep without gui, which will be reading pcap from stdin, and sending eep packets to the other sngrep (third terminal) mkfifo /tmp/pipe cat /tmp/pipe | sngrep -N -q -H udp:127.0.0.1:5077 -I - #second ssh terminal # writes as pcap to fifo pipe what freeswitch writes and reads from ssl lib python ssl_logger_giova.py -pcap /tmp/pipe freeswitch #third ssh terminal # sngrep that receives packets from both the Ethernet device, and the eep packets sent by the other sngrep (eg, the tls packets ssl_logger grabs from freeswitch's ssl lib) sngrep -L udp:127.0.0.1:5077 (you may want to edit ssl_logger.py and change 228 to be 101 - LINKTYPE_IPV4 to be LINKTYPE_RAW ) -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 2

Planning Fosdem 2019
by Daniel-Constantin Mierla 02 Feb '19

02 Feb '19

Hello, Fosdem 2019 is approaching, so I am writing to see if any of you plans to go to the event. If there is interest, we can try to organize again a dinner on Saturday evening, a tradition for our project at the past 10 editions or even more. Henning Westerholt will give a presentation about Kamailio in the RTC Devroom, I will be around as well. I know few more developers that plan to go to the event, so let's see who else from the community wants to join us. As usual, at Fosdem will be developers from other VoIP projects, like Asterisk, Janus, CGRates, Homer, Jitsi, ... At the past editions we typically had two "kamailio" events: 1) an "ad-hoc" developers meeting in the cantina (or other available room around) to discuss about short term plans for Kamailio -- time and place being decided as we meet there between us (expected in the afternoon of Saturday or during Sunday). 2) a dinner at a place nearby, with other VoIP folks joining us Reply if you plan to go to Fosdem and say if you want to join for a dinner. Just be aware that you have to pay for your food and drinks at the dinner, unless we are going to be surprised again by a generous sponsor that covers partially or completely to dinner. If you need more details about Fosdem, the website is: - https://fosdem.org 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 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com

3 5

kamailio + asterisk + webrtc
by arish haque 01 Feb '19

01 Feb '19

Hi Daniel, Within a network above tech stack is working fine for VOIP calls. But when i am trying from outside there is no sign of RTP. Cmd for rtp - /sbin/rtpengine -p /var/run/rtpengine.pid --interface=192.168.1.249\!106.51.78.78 --listen-ng 127.0.0.1:60000 -m 50000 -M 55000 --log-level 6 --log-facility local1 106.51.78.78 - public ip 192.168.1.249 - priv ip (kamailio + RTPEngine) Kamailio config when call is b/w WebRTC --> WebRTC rtpengine_manage("trust-address replace-origin replace-session-connection direction=internal direction=external"); Here is SDP, captured on the client sitting outside. v=0 o=root 1133801452 1133801452 IN IP4 106.51.78.78 s=Asterisk PBX 16.0.0 c=IN IP4 106.51.78.78 t=0 0 m=audio 50196 UDP/TLS/RTP/SAVPF 0 8 111 9 126 a=maxptime:60 a=ice-ufrag:5e13a1292fc0b7163d49328b7516f763 a=ice-pwd:4fa5e003195492fa165020c81eff9346 a=candidate:Hc0a801ac 1 UDP 2130706431 192.168.1.172 46810 typ host a=candidate:Hc0a801ac 2 UDP 2130706430 192.168.1.172 46811 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 AA:5A:51:BD:4C:53:65:E4:2B:EC:EB:BF:A9:07:DD:60:E3:46:D8:26:6D:04:C8:21:8B:B9:81:37:3A:EB:55:C0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=sendrecv a=rtcp:50196 a=rtcp-mux a=ptime:20 a=candidate:T7Tqd3oKM4NvATH3 1 UDP 2097152255 106.51.78.78 50196 typ host Looking forward for help from you guys. Thanks & Regards, Arish Haque

1 0

dispatcher alg - 8
by Andrew Chen 31 Jan '19

31 Jan '19

Hey all, I need some more clarification on dispatcher alg 8: “8” - use first destination (good for failover). What is meant by "first destination"? How is that defined if I have more than one carrier. More specifically in our case for some PSTN locations we have this setup: primary carrier - 1 sip trunk backup carrier - 4 sip trunks or primary carrier - 4 sip trunks backup carrier - 1 sip trunk the one that have 4 sip trunks is it going to round-robin through it? this is setup on our old 4.1.5 release. thanks -- Andy Chen 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

IMS early media
by Tsvetan Filev 31 Jan '19

31 Jan '19

Hi all. I want to use SIP early media to play music to the caller in kamailio IMS installation like this: http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html I looked a little bit but didn't find ready solution. The information is vague on this topic. Should this be done through a module or application server ? May I need to handle ringing in onreply_route and send OK with SDP to the caller in SCSCF ? Regards.

2 8

Kamailio/dispatcher routing with SRV record
by Jesse Strahn 31 Jan '19

31 Jan '19

Is it possible to route to the back-end SIP servers via DNS SRV record rather than specifying multiple IP addresses in the dispatcher list on the Kamailio server? Running in AWS and looking to be able to dynamically update my list of available sip servers if a server goes offline or if a server comes online. Have Cloudwatch/Lambda setup to manipulate the SRV record with the IP address additions or subtractions but when I try to put the SRV record as the destination in dispatcher, it fails.

2 2

Kamailio v5.1.7 Released
by Daniel-Constantin Mierla 31 Jan '19

31 Jan '19

Hello, Kamailio SIP Server v5.1.7 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.6. 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 v5.1.7 or to v5.2.x series. For more details about version 5.1.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/01/kamailio-v5-1-7-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 v5.2.1 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 Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com

1 0

Planing the release of Kamailio v5.1.7
by Daniel-Constantin Mierla 31 Jan '19

31 Jan '19

Hello, I am considering to release Kamailio v5.1.7 later this week, likely on Thursday, January 31, 2019. As usual, here come the notice to report any issue that you are aware of and not yet registered on the bug tracker in order to give them a chance to be analyzed and eventually fixed before the release. Also, take a look at the branch 5.1 in git and check if there are any missing commits from those you expected to be backported. 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 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com

1 1

Altering received parameter in Contact
by David Cunningham 30 Jan '19

30 Jan '19

Hello, I'm hoping someone can give advice on altering the received parameter in the Contact header on a reply. The situation is that a customer is REGISTERing using TLS, and our Kamailio replies with a 200 OK that includes a Contact header like this: Contact: <sip:123456789@xx.xx.102.10 :5061;transport=tls>;expires=3600;received="sip:xx.xx.102.10:36051;transport=TLS" The customer's device apparently had an issue with this, and the device manufacturer advises that they need the ";transport=TLS" on the received parameter removed. This sounds like a strange requirement, but it's what we've been told. Does anyone know how to alter this in a Kamailio configuration? Our system uses www_authorize to authenticate and is running Kamailio 4.2.1. Thanks in advance for any help! -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

5 9

Kamailio TCP/TLS Connection getting failed
by arish haque 30 Jan '19

30 Jan '19

Hi All, I am running kamailio 5.2.x on CentOS 7. Call is happening but i am getting the below error continuously in kamailio logs. WARNING: websocket [ws_frame.c:228]: encode_and_send_ws_frame(): TCP/TLS connection get failed Jan 30 17:57:01 localhost /usr/local/sbin/kamailio[22097]: ERROR: websocket [ws_frame.c:765]: ping_pong(): sending keepalive Any clue of why this is happening? Thanks, Arish Haque

1 0

TSILO on Kamailio 5.2.0
by Володимир Іванець 29 Jan '19

29 Jan '19

Hello! I am trying to repeat awesome configuration provided by Federico Cabiddu https://www.youtube.com/watch?v=4XIrR9bwUkM but only got it to work partially. So I'm wondering if someone else has same problem as me on latest Kamailio? I have two clients configured on 100-test account. Only one is registered. When *ts_store()* is executed on INVITE I get: *Jan 28 19:17:46 test /usr/sbin/kamailio[11204]: DEBUG: tsilo [ts_store.c:80]: ts_store(): storing transaction 29157:1612520187 for r-uri: 100-test* *Jan 28 19:17:46 test /usr/sbin/kamailio[11204]: DEBUG: tsilo [ts_hash.c:264]: insert_ts_urecord(): urecord entry 0x7f2bfcf7dd78* *Jan 28 19:17:46 test /usr/sbin/kamailio[11204]: DEBUG: tsilo [ts_handlers.c:56]: ts_set_tm_callbacks(): registered TMCB for transaction 29157:1612520187* *Jan 28 19:17:46 test /usr/sbin/kamailio[11204]: DEBUG: tsilo [ts_store.c:103]: ts_store(): transaction 29157:1612520187 (ruri: 100-test) inserted* *# kamcmd ts.dump*: *{* * Size: 2048* * R-URIs: {* * R-URI: 100-thirdlane* * Hash: 451001090* * Transactions: {* * Transaction: {* * Tindex: 29157* * Tlabel: 1612520187* * }* * }* * }* * Stats: {* * RURIs: 1* * Max-Slots: 1* * Transactions: 1* * }* *}* When second UAC registers, *ts_append("location", "$tU")* in route *PUSHJOIN* is executed. It throws following error: *Jan 28 19:18:10 test /usr/sbin/kamailio[11210]: ERROR: tsilo [tsilo.c:199]: ts_check_uri(): bad uri [100-test]* At same time running *kamcmd ts.lookup 100-test* command returns expected result: *{* * Transaction: {* * Tindex: 29157* * Tlabel: 1612520187* * }* *}* Thanks a lot!

2 2

Kamailio IMS Deployment
by Yağız AYDIN 29 Jan '19

29 Jan '19

Hello everyone, I'm new to Kamailio environment, and trying to implement Kamailio IMS services with FHoSS HSS on a single Ubuntu 18.04 . I have followed instructions on the Kamailio website, but faced some issues. I have managed to install FHoSS, but when I try to start HSS, i got following errors: [Thread-24] ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - StateMachine: Peer icscf.mnc001.mcc001.3gppnetwork.org:3869 not responding to connection attempt [Thread-24] ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - StateMachine: Peer scscf.mnc001.mcc001.3gppnetwork.org:3870 not responding to connection attempt I have started CSCFs by following commands: root@ulak-HP-ZBook-15:/opt/OpenIMSCore/FHoSS/deploy# kamailio -f /etc/kamailio/pcscf/kamailio-pcscf.cfg loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules Listening on udp: 10.3.0.188 [10.3.0.188]:5060 Aliases: *: pcscf.mnc001.mcc001.3gppnetwork.org:* ************************************************ root@ulak-HP-ZBook-15:/opt/OpenIMSCore/FHoSS/deploy# kamailio -f /etc/kamailio/scscf/kamailio-scscf.cfg loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules Listening on udp: 10.3.0.188 [10.3.0.188]:6060 Aliases: *: scscf.mnc001.mcc001.3gppnetwork.org:* *********************************************** root@ulak-HP-ZBook-15:/opt/OpenIMSCore/FHoSS/deploy# kamailio -f /etc/kamailio/icscf/kamailio-icscf.cfg loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules Listening on udp: 10.3.0.188 [10.3.0.188]:4060 tcp: 10.3.0.188 [10.3.0.188]:4060 Aliases: *: icscf.mnc001.mcc001.3gppnetwork.org:* How do I ensure that CSCFs work? And also how can I troubleshoot error that I've faced on FHoSS. I have attached my config files. Any helps will be appreciated. Best regards, Yağız --

2 1

TM Module - Unfortunately error on sending to next hop occurred (477/TM)
by Ilie Soltanici 28 Jan '19

28 Jan '19

Hello, Can someone help me to understand this error given by TM Module? I have a Kamailio SIP Proxy running in front of a few Asterisk boxes. On average, he is processing 70-80 requests per second. The problem is that sometimes (90-100 times/day) - I see such errors in the logs: 500 I'm terribly sorry, server error occurred (1/SL) 477 Unfortunately error on sending to next hop occurred (477/TM) Trying to investigate this error - I found that this is happening randomly and for INVITES coming from the Asterisk Box. For ex, the INVITE below: 192.168.180.10 - Kamailio Server 192.168.180.36 - Asterisk Server INVITE sip:1001@192.168.180.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.180.36:5060;branch=z9hG4bK4624ab4e;rport Max-Forwards: 70 From: "Anonymous" <sip:anonymous@192.168.180.10>;tag=as11bed8a6 To: <sip:1001@192.168.180.10:5060> Contact: <sip:anonymous@192.168.180.36:5060> Call-ID: 0ea8535d3a5ecaa6432912a2566c807f(a)192.168.180.10 CSeq: 102 INVITE User-Agent: MYCOMPANY Date: Mon, 28 Jan 2019 16:01:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=SIP 2139556641 2139556641 IN IP4 192.168.180.36 s=MYCOMPANY PBX c=IN IP4 192.168.180.36 t=0 0 m=audio 14674 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Kamailio is replying back with 100 Trying and then with 500/477 errors: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.180.36:5060 ;branch=z9hG4bK4624ab4e;rport=5060;received=192.168.180.36 From: "Anonymous" <sip:anonymous@192.168.180.10>;tag=as11bed8a6 To: <sip:1001@192.168.180.10:5060> Call-ID: 0ea8535d3a5ecaa6432912a2566c807f(a)192.168.180.10 CSeq: 102 INVITE Server: MYCOMPANY SBC Content-Length: 0 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Via: SIP/2.0/UDP 192.168.180.36:5060 ;branch=z9hG4bK4624ab4e;rport=5060;received=192.168.180.36 From: "Anonymous" <sip:anonymous@192.168.180.10>;tag=as11bed8a6 To: <sip:1001@192.168.180.10:5060>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.2e93 Call-ID: 0ea8535d3a5ecaa6432912a2566c807f(a)192.168.180.10 CSeq: 102 INVITE Server: MYCOMPANY SBC Content-Length: 0 SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM) Via: SIP/2.0/UDP 192.168.180.36:5060 ;branch=z9hG4bK4624ab4e;rport=5060;received=192.168.180.36 From: "Anonymous" <sip:anonymous@192.168.180.10>;tag=as11bed8a6 To: <sip:1001@192.168.180.10:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2e93 Call-ID: 0ea8535d3a5ecaa6432912a2566c807f(a)192.168.180.10 CSeq: 102 INVITE Server: MYCOMPANY SBC Content-Length: 0 And I cannot actually get what is wrong with this INVITE - and why Kamailio cannot process it? Other calls for the same extension are working fine, this is happening randomly and with different extensions. Load on the server is very low: [root@kamailio root]# nproc 8 [root@kamailio root]# uptime 17:49:15 up 8 days, 21:17, 6 users, load average: 0.66, 0.56, 0.52 [root@kamailio root]# free -m total used free shared buff/cache available Mem: 7981 4283 142 818 3556 2581 Swap: 6141 137 6004 [root@kamailio root]# ss -4 -n -l | grep 5060 udp UNCONN 0 0 192.168.180.10 :5060 *:* udp UNCONN 0 0 127.0.0.1:5060 *:* tcp LISTEN 0 128 192.168.180.10 :5060 *:* tcp LISTEN 0 128 127.0.0.1:5060 *:* In the kamailio logs I found that kamailio was able to get the contact address: kamailio[18168]: INFO: {INVITE (1) Contacts loaded for 1001} kamailio[18168]: INFO: {INVITE (1) t_next_contacts - Only one contact found for 1001, calling} kamailio[18168]: INFO: {INVITE (1) Next Hop: <192.168.180.211:3126>} I don't know how to reproduce this - I tried to disconnect the phone from the power source - and made a call to that extension, and it is giving timeouts - like it is supposed to be. What could be the problem and how I can fix it? Thank You.

1 0

Playing with Diameter - Error opening connection to :3868
by Steve Bucklin 28 Jan '19

28 Jan '19

Hello all, I am brand new to this list and I am also on a steep learning curve with Kamailio and Diameter.... so please be gentle as I am just starting! I am trying to look at the Diameter signaling and have a glimpse into the information passed. I have set up a compiled 5.2 on CentOS 7. I suspect that I am missing something obvious, but not sure what. I am seeing this error: INFO: cdp [peerstatemachine.c:525]: I_Snd_Conn_Req(): I_Snd_Conn_Req(): Peer peer_connect(): peer_connect(): Error opening connection to :3868 >Name or service not known It looks like the FQDN in the diameter.xml under 'peer' is not being resolved, as it is seeing port 3868? Any pointers would be greatly taken :-) Here is my diameter: <?xml version="1.0" encoding="UTF-8"?> <DiameterPeer FQDN="mra1.testservers.xxx" Realm="testservers.xxx" Vendor_Id="10415" Product_Name="CDiameterPeer" AcceptUnknownPeers="1" DropUnknownOnDisconnect="1" Tc="30" Workers="16" QueueLength="32" TransactionTimeout="5" SessionsHashSize="128" DefaultAuthSessionTimeout="3600" MaxAuthSessionTimeout="3600" > <peer FGDN="mra2.testservers.xxx" Realm="testservers.xxx" proto="sctp" port="3868"/> <Acceptor port="3868" bind="192.168.1.20"/> <Auth id="16777251" vendor="10415"/>  <SupportedVendor vendor="10415" /> </DiameterPeer> And here is what I am getting on the startup: Jan 28 14:19:35 telco-lon1-02 kamailio: INFO: <core> [core/tcp_main.c:4722]: init_tcp(): using epoll_lt as the io watch method (auto detected) Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22778]: INFO: <core> [main.c:2649]: main(): processes (at least): 29 - shm size: 67108864 - pkg size: 8388608 Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22778]: INFO: cdp [cdp_mod.c:240]: cdp_child_init(): CDiameterPeer child starting ... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22783]: INFO: cdp [worker.c:332]: worker_process(): [1] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22778]: INFO: cdp [cdp_mod.c:242]: cdp_child_init(): ... CDiameterPeer child started Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22790]: INFO: cdp [worker.c:332]: worker_process(): [8] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22784]: INFO: cdp [worker.c:332]: worker_process(): [2] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22786]: INFO: cdp [worker.c:332]: worker_process(): [4] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22793]: INFO: cdp [worker.c:332]: worker_process(): [11] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22794]: INFO: cdp [worker.c:332]: worker_process(): [12] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22796]: INFO: cdp [worker.c:332]: worker_process(): [14] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22788]: INFO: cdp [worker.c:332]: worker_process(): [6] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22799]: INFO: cdp [receiver.c:455]: receiver_process(): receiver_process(): [] Receiver process doing init on new process... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22799]: INFO: cdp [receiver.c:188]: add_serviced_peer(): add_serviced_peer(): Adding serviced_peer_t to receiver for peer [] Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22799]: INFO: cdp [receiver.c:460]: receiver_process(): receiver_process(): [] Receiver process starting up... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22791]: INFO: cdp [worker.c:332]: worker_process(): [9] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22792]: INFO: cdp [worker.c:332]: worker_process(): [10] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22785]: INFO: cdp [worker.c:332]: worker_process(): [3] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22787]: INFO: cdp [worker.c:332]: worker_process(): [5] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22795]: INFO: cdp [worker.c:332]: worker_process(): [13] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22782]: INFO: cdp [worker.c:332]: worker_process(): [0] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22789]: INFO: cdp [worker.c:332]: worker_process(): [7] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22797]: INFO: cdp [worker.c:332]: worker_process(): [15] Worker process started... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22798]: INFO: cdp [receiver.c:455]: receiver_process(): receiver_process(): [] Receiver process doing init on new process... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22798]: INFO: cdp [receiver.c:460]: receiver_process(): receiver_process(): [] Receiver process starting up... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22800]: INFO: cdp [acceptor.c:81]: acceptor_process(): Acceptor process starting up... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22800]: WARNING: cdp [tcp_accept.c:121]: create_socket(): create_socket(): Trying to open/bind/listen on 192.168.1.20 port 3868 Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22800]: WARNING: cdp [tcp_accept.c:146]: create_socket(): create_socket(): Successful socket open/bind/listen on 192.168.1.20 port 3868 Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22800]: INFO: cdp [acceptor.c:95]: acceptor_process(): Acceptor opened sockets. Entering accept loop ... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22801]: INFO: cdp [timer.c:205]: timer_process(): Timer process starting up... Jan 28 14:19:35 telco-lon1-02 journal: [Diameter Routing Agent][22802]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Jan 28 14:19:36 telco-lon1-02 journal: [Diameter Routing Agent][22801]: INFO: cdp [peerstatemachine.c:525]: I_Snd_Conn_Req(): I_Snd_Conn_Req(): Peer Jan 28 14:19:36 telco-lon1-02 journal: [Diameter Routing Agent][22801]: WARNING: cdp [receiver.c:866]: peer_connect(): peer_connect(): Error opening connection to :3868 >Name or service not known Thanks, Steve

2 2

database reconnection db_unixodbc
by Karthik Srinivasan 27 Jan '19

27 Jan '19

Hello, I am testing how kamailio reacts to various database conditions. One such condition is if the database engine is simply shut down (that is, database server process no longer running, tcp listening socket closed, etc...) I am utilizing the db_unixodbc module to connect to an Informix database engine. I am currently running on Kamailio version 5.0. I have a test query that executes against the database engine every 10 seconds. Here is what i have noticed if i shut down the database engine at some point after i run Kamailio. the first test query that attempts to run against the db engine fails; it tries to reconnect and fails. The second test query (10 seconds after the 1st) results in a SIG_CHILD and shuts down the entire Kamailio process. Has anyone experienced this? Is there a solution to this? Ideally the second query should also fail and return gracefully; and ideally queries continue to fail until the database engine is back up. See logs below: Jan 17 20:07:25 [29297]: INFO: (s) SQL query: FIRST TEST QUERY Jan 17 20:07:25 [29297]: ERROR: db_unixodbc [dbase.c:135]: db_unixodbc_submit_query(): rv=-1. Query= FIRST TEST QUERY Jan 17 20:07:25 [29297]: ERROR: db_unixodbc [connection.c:220]: db_unixodbc_extract_error(): unixodbc:SQLExecDirect=08S01:1:-11020:[Informix][Informix ODBC Driver]Communication link failure. Jan 17 20:07:25 [29297]: ERROR: db_unixodbc [dbase.c:59]: reconnect(): Attempting DB reconnect Jan 17 20:07:25 [29297]: ERROR: db_unixodbc [dbase.c:74]: reconnect(): failed to connect Jan 17 20:07:25 [29297]: ERROR: db_unixodbc [connection.c:220]: db_unixodbc_extract_error(): unixodbc:SQLDriverConnect=08002:1:0:[unixODBC][Driver Manager]Connection name in use Jan 17 20:07:25 [29297]: ERROR: db_unixodbc [connection.c:220]: db_unixodbc_extract_error(): unixodbc:SQLDriverConnect=HY010:2:-11067:[Informix][Informix ODBC Driver]Function sequence error. Jan 17 20:07:25 [29297]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jan 17 20:07:25 [29297]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query FIRST TEST QUERY Jan 17 20:07:25 [29297]: INFO: (s) [123] SQL ret: fail (-1) Jan 17 20:07:25 [29297]: INFO: (s) [123] SQL res: no rows Jan 17 20:07:35 [29297]: INFO: (s) [123] SQL query: 10 seconds later the SECOND TEST QUERY (it's the same query as the first one) Jan 17 20:07:35 [29301]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 28 Jan 17 20:07:35 [29283]: ALERT: <core> [main.c:744]: handle_sigs(): child process 29297 exited by a signal 11 Jan 17 20:07:35 [29283]: ALERT: <core> [main.c:747]: handle_sigs(): core was not generated Jan 17 20:07:35 [29283]: INFO: <core> [main.c:759]: handle_sigs(): terminating due to SIGCHLD Jan 17 20:07:35 [29301]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29295]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29291]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29288]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29300]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29284]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29286]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29293]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29289]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29287]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29292]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29296]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29298]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29299]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29285]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29294]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29290]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Jan 17 20:07:35 [29283]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Thank you, Karthik

2 5

Why kamailio can not be a consumer of RabbitMQ?
by CuiGang 26 Jan '19

26 Jan '19

Hello every one. I noticed that module RABBITMQ just implements the publish functions. It means kamailio can only as a producer of RabbitMQ. My question is Why RABBITMQ module not implement consumer functions? As NSQ, let kamailio can be a consumer receive messages from RabbitMQ. Regards. CuiGang cuigang0120(a)aliyun.com

2 1

DMQ errors
by Aleksandar Sosic 25 Jan '19

25 Jan '19

Hi, when I have only one node and dmq enabled I've got this errors: ``` 17(45) ERROR: dmq [notification_peer.c:337]: add_server_and_notify(): error adding notification node 17(45) ERROR: dmq [dmq_funcs.c:516]: ping_servers(): cannot retrieve initial nodelist from sip:dmq-router-service.alex.svc.cluster.local:5062 ``` On the second node booting after this one I don't get any of these errors. Is there any flag/option to turn off this error on the `cannot retrieve initial nodelist` in case I have only one node for a period of time? Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

2 2

Re: [SR-Users] kamailio-lb fails after topos enabled.
by Daniel Tryba 25 Jan '19

25 Jan '19

On Fri, Jan 25, 2019 at 10:19:45AM -0300, Marcos Pytel wrote: > Could you help me to catch the coredump file? Please keep the mailinglist in the loop. > When I execute bt full the systems shows "no symbols". What package i need to perform it? I don't know what the sipwise config is, but it if is (anything like) debian the normal kamailio package is strippped of debugging symbols. You'll need to install the debug package (kamailio-dbg in debian).

1 0

kamailio-lb fails after topos enabled.
by Marcos Pytel 25 Jan '19

25 Jan '19

Hi team! I'm using Kamailio version: kamailio 5.1.6 (x86_64/linux). When I enabled topos, after a few minutes i get this erros in the log file and the LB Service goes down.: Jan 24 13:20:21 sipwise lb[30130]: ERROR: ndb_redis [redis_client.c:903]: redisc_exec_argv(): Redis error: Server closed the connection Jan 24 13:21:36 sipwise lb[30132]: ERROR: ndb_redis [redis_client.c:903]: redisc_exec_argv(): Redis error: Server closed the connection Jan 24 13:29:06 sipwise lb[30135]: ERROR: ndb_redis [redis_client.c:903]: redisc_exec_argv(): Redis error: Server closed the connection Jan 24 13:31:36 sipwise lb[30133]: ERROR: ndb_redis [redis_client.c:903]: redisc_exec_argv(): Redis error: Server closed the connection Jan 24 13:34:41 sipwise lb[30130]: ERROR: ndb_redis [redis_client.c:903]: redisc_exec_argv(): Redis error: Server closed the connection Jan 24 13:43:06 sipwise lb[30134]: ERROR: ndb_redis [redis_client.c:903]: redisc_exec_argv(): Redis error: Server closed the connection Jan 24 13:45:21 sipwise lb[30136]: ERROR: ndb_redis [redis_client.c:903]: redisc_exec_argv(): Redis error: Server closed the connection Jan 24 13:47:06 sipwise lb[30132]: ERROR: ndb_redis [redis_client.c:903]: redisc_exec_argv(): Redis error: Server closed the connection Jan 24 13:47:21 sipwise lb[30136]: ERROR: topos [tps_storage.c:340]: tps_storage_link_msg(): bad sip message or missing Contact hdr Jan 24 13:47:35 sipwise lb[30163]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 23 Jan 24 13:47:35 sipwise lb[30066]: ALERT: <core> [main.c:738]: handle_sigs(): child process 30120 exited by a signal 6 Jan 24 13:47:35 sipwise lb[30066]: ALERT: <core> [main.c:741]: handle_sigs(): core was generated I attached the coredump file. I tried to get bt full but i think that i need something else to catch it. Thank you! Marcos.

2 2

ERROR while using KEMI - Python
by Prabhat Kumar 25 Jan '19

25 Jan '19

Hi, I am getting this error message while starting kamailio. Enviroment: OS- Ubuntu 16.04.5 LTS (Xenial Xerus) python 2.7 ERROR: app_python [app_python_mod.c:164]: mod_init(): failed to load python script ERROR: <core> [core/sr_module.c:990]: init_mod(): Error while initializing module app_python (/usr/local/lib64/kamailio/modules/app_python.so)

2 2

Error while installing kamailio
by Prabhat Kumar 25 Jan '19

25 Jan '19

I am getting this error while installing kamailio LD (gcc) [U kamcmd] kamcmd /usr/bin/ld: cannot find -lncurses collect2: error: ld returned 1 exit status ../../src//Makefile.rules:191: recipe for target 'kamcmd' failed make[3]: *** [kamcmd] Error 1 ERROR: make --no-print-directory -C ../../../utils/kamcmd install-if-newer failed ../../Makefile.modules:145: recipe for target 'install-utils' failed make[2]: *** [install-utils] Error 1 Makefile:511: recipe for target 'install-modules' failed make[1]: *** [install-modules] Error 1 make[1]: Leaving directory '/usr/local/src/kamailio-5.1/kamailio/src' Makefile:36: recipe for target 'install' failed make: *** [install] Error 2

2 1

Kamailio Ignore Re-INVITE
by Alvaro Araos 25 Jan '19

25 Jan '19

Hi, When receive a Re-INVITE with new Media IP and PORT, kamailio keeps sending RTP to the old IP PORT. Seems kamailio not update info to the rtpproxy. Saludos, ALVARO ARAOS MÉNDEZ Director de Tecnología en FONOIP +56 2 2947 9000 alvaro(a)fonoip.com | www.fonoip.com [1] Moisés Chacón 46, El Monte, RM Links: ------ [1] http://www.fonoip.com

2 1

presence_mwi body check problem
by Володимир Іванець 24 Jan '19

24 Jan '19

Hello! I'm trying to process PUBLISH on Kamailio (v. 5.2.1) but constantly getting "500 Server Internal Error" responce. For packet body: *Messages-Waiting: yes* *Message-Account: sip:100-test@172.16.30.151 <sip%3A100-test(a)172.16.30.151>* *Voice-Message: 1/2* ... debug output is: *kamailio: 2(27127) DEBUG: presence [event_list.c:348]: search_event(): start event= [message-summary/6]* *kamailio: 2(27127) DEBUG: presence [publish.c:351]: ki_handle_publish_uri(): SIP-If-Match header not found* *kamailio: 2(27127) DEBUG: presence [presentity.c:88]: generate_ETag(): etag= a.1548164100.27127.7.0 / 22* *kamailio: 2(27127) DEBUG: presence [publish.c:360]: ki_handle_publish_uri(): new etag = a.1548164100.27127.7.0* *kamailio: 2(27127) DEBUG: presence [publish.c:385]: ki_handle_publish_uri(): Expires header found, value= 16* *kamailio: 2(27127) ERROR: presence_mwi [add_events.c:110]: mwi_publ_handl(): check of body <Messages-Waiting: yes* *kamailio: Message-Account: sip:100-test@172.16.30.151 <sip%3A100-test(a)172.16.30.151>* *kamailio: Voice-Message: 1/2* *kamailio: > failed at character number 22* *kamailio: 2(27127) ERROR: presence [publish.c:480]: ki_handle_publish_uri(): in event specific publish handling* I'm not sure, but it appears to me that check in line 95 from https://github.com/kamailio/kamailio/blob/5.2/src/modules/presence_mwi/add_… is not passing. I was trying to add "\n" and "\r" to the end of each string but I always had failure on character number 22. Does anyone know how exactly PUBLISH packet body should look like? Thanks a lot!

2 4

Problem at Rx Interface - kamailio P-CSCF 5.1.6
by Valentin Christoph 23 Jan '19

23 Jan '19

Hi all, We are using kamailio 5.1.6, configured as P-CSCF, facing following problem. Please refer to the attached tshark trace, too. If during call setup (by simulated UAs) at the terminating P-CSCF the simulated PCRF rejects the Diameter AAR message (frame 46/47 at the attached trace), then the kamailio P-CSCF runs into a questionable behaviour. By calling the Rx_AAR() function in configuration, the processing of the 200 OK (frame 45) is suspended (by the ims_qos module) and then resumed with the receipt of the AAA message (frame 47). 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 ____ 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.

1 1

MWI notification through pua.publish
by willy 23 Jan '19

23 Jan '19

Hello. Have anyone tried to use pua.publish to send MWI notification? Can it work at all? I'm sending request through jsonrpc server, but it is not dispatched by handle_publish. Thank you, Kirill

6 9

Maximum Number of Probed Gateways
by Duarte Rocha 23 Jan '19

23 Jan '19

Greetings, Is there a limitation to the number of Gateways probed by Dispatcher? I have several gateways configured to be probed, but the OPTIONS is only sent to the first 16? Is there a configuration regarding this? Thanks in advance, Duarte Rocha

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