sr-users April 2018

sr-users@lists.kamailio.org

81 participants
104 discussions

Failure Route Issue
by mack＠dopensource.com 18 Apr '18

18 Apr '18

Hey All, I have a weird issue. I’m sending a SIP INVITE to a SIP Service that send’s me back a 433, which let’s me know that the number is invalid. I then send an INVITE to a media server (FreeSwitch) to play a message that the number can’t be called, which looks to be working. But, the final ACK from the SIP client is sent back to the SIP Service versus the media server. The following diagram below depicts this: https://snag.gy/qMWfvk.jpg. The code snippet in the failure_route for the the SIP Service has this: if (t_check_status("403|433")) { if ($avp(s:teleblock_media_enabled) == "1") { # make sure media server can route back to kamailio record_route(); revert_uri(); $ru = "sip:" + $T_reply_code + $rU + "@" + $sel(cfg_get.teleblock.media_ip) + ":" + $sel(cfg_get.teleblock.media_port); append_branch(); t_relay(); exit; }

2 5

Kamailio Websocket Questions
by Nagorny, Dimitry 17 Apr '18

17 Apr '18

Hi List, two questions regarding Kamailio Websockets: 1. How do I send INVITE request TO the listening Websocket? I usually used for UDP/TCP/TLS these format, but somehow this is not working for Websocket because it gets in an endless loop: if ($rU=~"^(\+|00|0)?[1-9][0-9]{4,20}$") { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ";user=phone;transport=udp"; force_send_socket(udp:MY_EXTERN_IP:MY_EXTERN_PORT); rtpengine_offer("internal trust-address RTP AVP"); route(RELAY); exit; } 2. Why do I get these errors every time my Websocket Client is registering, but it still works and DB is inserted with registered location: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 22) ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TCP 10.250.5.17:51283#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: 6uLXlD/aBrzu/j3PYP3DWO7rvLE=#015#012Server: kamailio (4.3.5 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012> I am using Kamailio 4.3.5 + CentOS 7.2. I appreciate every help I can get. Best Regards Dimitry Nagorny Trainee

3 4

Dialog module's keep-alive feature uses wrong CSeq?
by Oded Arbel 17 Apr '18

17 Apr '18

Hi all. I'm trying to use the dialog module's keep alive feature (ka_interval() and dlg_set_property()), and the documentation (under dlg_set_property) says this: If keep alive is enabled for a dialog, the module will send SIP OPTIONS requests with CSeq lower or equal than last request within dialog This is seems really weird to me, but the problem is worse than that - when setting dlg_set_property("ka-src"), the OPTIONS messages sent to the caller always have CSeq set to "0", regardless of what CSeq the UA sent on its invite. The problem is that while some UAs respond to these purposefully broken messages with some kind of a 5xx reply (for example, Linphone funnily sends "500 Internal Server Error"), several UAs don't respond at all. Worse than that - some of the UAs that send back 5xx replies to OPTIONS with a non-zero but invalid CSeq will completely ignore the OPTIONS keep-alive message that is sent to the caller with a CSeq 0 - for example, PJSIP responds with "503 Invalid CSeq" to a "ka-dst" OPTIONS message but will ignore a "ka-src" OPTIONS message with CSeq 0. My questions: 1. Why are keep-alive messages sent with a wrong CSeq? 2. If we're keeping the "wrong CSeq" approach, can we at least not send CSeq "0"? 3. Is there a way to change any of the above behaviors with a configuration (i.e. without patching the sources)? Thanks in advance -- Oded Arbel odeda(a)cloudonix.io

3 5

Kamailio with 2 Public and Private IPs- SIP Call flow issues
by arik halperin 16 Apr '18

16 Apr '18

Hello, I have the following setup: Kamailio: Public Interface 70.36.25.65 Private Interface 10.168.10.227 Freeswitch: Private Interface 10.168.10.200 I’m routing calls from kamailio to freeswitch, but when I try to hang up the call the freeswitch sends the BYE to the public address(70.36.25.65) and not to the private address 10.168.10.227. 200OK and ACK work correctly. It’s only the BYE that has an issue. Showing INVITES: Client Invite: T 82.80.164.63:53871 -> 70.36.25.65:5099 [AP] INVITE sip:+972549428460@70.36.25.65:5099 SIP/2.0. Via: SIP/2.0/TCP 10.0.0.42:51619;branch=z9hG4bK-524287-1---52dcc73ec891de14;rport. Max-Forwards: 70. Contact: <sip:972545234585@82.80.164.63:51620;rinstance=1f9d21b54116df89;transport=tcp>. To: <sip:+972549428460@70.36.25.65:5099>. From: "972545234585"<sip:972545234585@70.36.25.65:5099>;tag=902c5325. Kamailio to FS: T 10.168.10.227:46506 -> 10.168.10.200:5080 [AP] INVITE sip:+972549428460@70.36.25.65:5099 SIP/2.0. Record-Route: <sip:70.36.25.65:5099;transport=tcp;lr=on;ftag=902c5325;nat=yes>. Via: SIP/2.0/TCP 70.36.25.65:5099;branch=z9hG4bK1117.1b91cfe0632bd6f036b308f85898e862.1;i=1. Via: SIP/2.0/TCP 10.0.0.42:51619;received=82.80.164.63;branch=z9hG4bK-524287-1---9b6758521ff0f528;rport=53871. Max-Forwards: 69.

3 3

Problem with resuming suspended transacation
by Arik Halperin 16 Apr '18

16 Apr '18

Hello, I’m trying to resume a suspended transaction, but keep getting this error: WARNING: tm [t_suspend.c:193]: t_continue(): transaction is not suspended [20608:1256194941] WARNING: tmx [tmx_mod.c:686]: w_t_continue(): resuming the processing of transaction [20608:1256194941] failed WARNING: tm [t_lookup.c:1483]: t_unref(): script writer didn't release transaction I’m implementing push handling, in INVITE I do the following: route[INVITE] { if (!lookup("location")) { send_reply("100", "Trying"); record_route(); route(SUSPEND); } else { # NAT detection route(NATMANAGE); if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); record_route(); t_relay(); ts_store(); $sht(vtp=>stored::$rU) = 1; } route(SENDPUSH); } # suspend the transaction route[SUSPEND] { if(!t_suspend()) //Transaction is suspended! { exit; } $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" +$T(id_label); } In Register I call the PUSHJOIN route: # append branches or resume the transaction route[PUSHJOIN] { $var(hjoin) = 0; lock("$tU"); $var(hjoin) = $sht(vtp=>join::$tU); $var(hstored) = $sht(vtp=>stored::$tU); $sht(vtp=>join::$tU) = $null; unlock("$tU"); if ($var(hjoin)==0) { if ($var(hstored)) ts_append("location", "$tu"); return; } $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); ($var(hjoin))\n"); t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } For some reason I keep getting: Apr 12 12:37:11 kamprod /usr/local/sbin/kamailio[6039]: ERROR: <script>: 1111 suspended transaction [20608:1256194941] FreeSWITCH => 972587102881 pr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: ERROR: <script>: 1111 tU=972587102881 20608:1256194941 0 Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: ERROR: <script>: 1111 resuming trasaction [20608:1256194941] 972587102881 (20608:1256194941) Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: WARNING: tm [t_suspend.c:193]: t_continue(): transaction is not suspended [20608:1256194941] Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: WARNING: tmx [tmx_mod.c:686]: w_t_continue(): resuming the processing of transaction [20608:1256194941] failed Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: WARNING: tm [t_lookup.c:1483]: t_unref(): script writer didn't release transaction

3 2

Audio stops after resuming call from hold
by gerry kernan 16 Apr '18

16 Apr '18

Hi Hoping someone can point me in the right direction. I have a Kamailio Ver: 4.2.3-1.1 running in front of a few asterisk servers Ver: 13.17.2 sip is routed to an asterisk server depending the domain name in the sip request, all working as expected . but if a call is put on hold after resuming the call the party that placed the call on hold can't hear any audio. The other party can hear . do I need to do anything special to handle re-invites for calls put on hold? Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

3 8

Help me!
by Do Quang Trung 15 Apr '18

15 Apr '18

I use kamailio 5.1.1 with openssl-1.0.2n. I built openssl with gost engine supported and the result when i list cipher list as follow: .................... GOST2001-GOST89-GOST89:GOST94-GOST89-GOST89:ECDH-RSA-AES256-GCM-SHA384:ECDH-ECDSA-AES256-GCM-SHA384:ECDH-RSA-AES256-SHA384:ECDH-ECDSA-AES256-SHA384:ECDH-RSA-AES256-SHA:ECDH-ECDSA-AES256-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA::GOST94-GOST89-GOST89:ECDH-RSA-AES256-GCM-SHA384:ECDH-ECDSA-AES256-GCM-SHA384:ECDH-RSA-AES256-SHA384:ECDH-ECDSA-AES256-SHA384:ECDH-RSA-AES256-SHA:ECDH-ECDSA-AES256-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA: ................ i must use exactly GOST2001-GOST89-GOST89 in kamailio to protect sip protocol. Please help me to resolve my problem. Best regards, Trung.

3 2

Kamailio 4.2.8 has crashed
by igor.potjevlesch＠gmail.com 13 Apr '18

13 Apr '18

Hello, I had a crash on one Kamailio instance with the following backtrace: Core was generated by `/usr/local/sbin/kamailio -m 704 -M 128 -P /run/kamailio/kamailio.pid'. Program terminated with signal 6, Aborted. #0 0x00007f92366761f7 in raise () from /lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.168-8.el7.x86_64 glibc-2.17-196.el7_4.2.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-8.el7.x86_64 libacl-2.2.51-12.el7.x86_64 libattr-2.4.46-12.el7.x86_64 libcap-2.22-9.el7.x86_64 libcom_err-1.42.9-10.el7.x86_64 libdb-5.3.21-21.el7_4.x86_64 libgcc-4.8.5-16.el7_4.1.x86_64 libselinux-2.5-11.el7.x86_64 libstdc++-4.8.5-16.el7_4.1.x86_64 lm_sensors-libs-3.4.0-4.20160601gitf9185e5.el7.x86_64 lua-5.1.4-15.el7.x86_64 mariadb-libs-5.5.56-2.el7.x86_64 net-snmp-agent-libs-5.7.2-28.el7_4.1.x86_64 net-snmp-libs-5.7.2-28.el7_4.1.x86_64 nspr-4.13.1-1.0.el7_3.x86_64 nss-3.28.4-15.el7_4.x86_64 nss-softokn-freebl-3.28.3-8.el7_4.x86_64 nss-util-3.28.4-3.el7.x86_64 openssl-libs-1.0.2k-8.el7.x86_64 pcre-8.32-17.el7.x86_64 perl-libs-5.16.3-292.el7.x86_64 popt-1.13-16.el7.x86_64 rpm-libs-4.11.3-25.el7.x86_64 tcp_wrappers-libs-7.6-77.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-17.el7.x86_64 (gdb) backtrace #0 0x00007f92366761f7 in raise () from /lib64/libc.so.6 #1 0x00007f92366778e8 in abort () from /lib64/libc.so.6 #2 0x000000000061fb2c in qm_free (qm=0x7f91f4eaf000, p=0x7f92110d0300, file=0x7f9226dd7072 "dialog: dlg_hash.c", func=0x7f9226dd9aeb <__FUNCTION__.12761> "destroy_dlg", line=381) at mem/q_malloc.c:474 #3 0x00007f9226d6fb62 in destroy_dlg (dlg=0x7f920f775598) at dlg_hash.c:381 #4 0x00007f9226d75681 in dlg_unref (dlg=0x7f920f775598, cnt=2) at dlg_hash.c:874 #5 0x00007f9226da14d5 in dlg_ontimeout (tl=0x7f920f7755f8) at dlg_handlers.c:1488 #6 0x00007f9226db1e67 in dlg_timer_routine (ticks=8621904, attr=0x0) at dlg_timer.c:283 #7 0x00000000004af537 in compat_old_handler (ti=137950479, tl=0x7f91f7ad8cc8, data=0x7f91f7ad8cc8) at timer.c:1011 #8 0x00000000004aff14 in slow_timer_main () at timer.c:1145 #9 0x000000000052a2af in main_loop () at main.c:1684 #10 0x000000000052fe71 in main (argc=7, argv=0x7fff51f21418) at main.c:2581 (gdb) Is there enough information to understand the reason of the crash? Regards, Igor.

3 5

Help needed to compile and install kamailio 5.1.2
by vinay kumar 13 Apr '18

13 Apr '18

Hi All, can anyone share me the installtion steps for kamailio 5.1.2? can i compile and run kamailio 5.1.2 on ubuntu 14.04? i have already used 4.3.6 with polaris configuration to bringup kamailio and i wanna upgrade to kamailio 5.1.2 and the same way can i use 5.1.2 with polaris configuration? I have used polaris scripts to bringup kamailio 5.1.2 and its failing with status kamailio service with Result resources. if i use latest debian packages , are polaris configurations are compatible with latest packages?? Please help me out. Thanks in advance. Regards, vinay

1 0

Is RTPProxy the way to go?
by Robert Johnson 12 Apr '18

12 Apr '18

I'm encountering a situation where an endpoint will renegotiate it's RTP ports but RTPProxy does not honor the new ports and continues to send audio to the old ports. I realize that I'm running a (very) old version of RTPProxy and that I should upgrade before submitting any bugs or asking for help with the program. My question is, is RTPProxy still the way to go? And if so, is v2.0.0 the best version to go with? https://github.com/sippy/rtpproxy/releases At a basic level, the network looks like this +---------------+ +--------------+ | | | | | Priv. Network | | Pub. Network | | | | | +------+ +---------+--+ +---+-------+--+ +--+--------+ | | | | | | | | | PSTN <---> Softswitch <--------> Kamailio + <--------> EndPoints | | | | | | RTPProxy | | | +------+ +---------+--+ | | +--+--------+ | +---+-------+--+ | | | | | +---------------+ +--------------+

2 1

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sr-users April 2018