- sr-users - kamailio.org

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

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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

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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.

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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

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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

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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

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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

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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; }

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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

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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

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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

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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

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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

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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

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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

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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.*

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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.

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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.

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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

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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

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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

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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

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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!

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