- sr-users - kamailio.org

Kamailio - FreeSWITCH - RTPproxy
by Jerry Kendall 20 Sep '21

20 Sep '21

Hello all, I am new to the list but have been reading plenty. Loads of good tips and help. I am playing with Freeswitch and would like to put Kamailio/RTPProxy in front for the user endpoints. Freeswitch will talk directly with the service provider. I am wondering *if* i can do this... 1) have the endpoints register with kamailio 2) have the media proxy via RTPProxy as it is needed for NAT 3) have the actual work (voicemail/ring groups/ivr/conference/etc...) done on Freeswitch 4) have all the SUBSCRIBE/PUBLISH/PRESENCE done on Kamailio Is this doable? Are there any guides to setting this sort of thing up? regards, Jerry

3 3

How to get $fU and $tU from MSRP module?
by BOEINGNG 20 Sep '21

20 Sep '21

Hello! I want to get $fU and $tU from event_route[msrp:frame-in] in order to filter out MSRP messages sent by a specific sender and received by a specific receiver, I know I can get message content from $msrp(buf), the message content is like this: MSRP 7a57f552155926c0 SEND To-Path: msrp://172.18.0.14:6060/9d8da4a34dc2102959e7;tcp msrp://172.18.0.12:5060/msrp-6142fc2b-84-1;tcp msrp://172.18.0.2:34862/2a67b54ae0750b7d82aa;tcp From-Path: msrp://172.18.0.14:6060/c71ae3675e114d08216b;tcp msrp://172.18.0.12:5060/msrp-6142fc2b-83-1;tcp msrp://172.18.0.2:34858/8baea7c52ba8755322fd;tcp Byte-Range: 1-197/197 Failure-Report: yes Success-Report: yes Message-ID: c792c94039b8a969 Content-Type: message/cpim From: sip:alice@ims.mnc001.mcc001.3gppnetwork.org To: sip:bob@ims.mnc001.mcc001.3gppnetwork.org DateTime: 2021-09-16T09:23:49.339805+00:00 MIME-Version: 1.0 Content-Type: text/plain hello -------7a57f552155926c0$ I want to get alice and bob from the content below： From: sip:alice@ims.mnc001.mcc001.3gppnetwork.org To: sip:bob@ims.mnc001.mcc001.3gppnetwork.org DateTime: 2021-09-16T09:23:49.339805+00:00 someone can suggest some method. In other route like request_route we can use $fU and $tU to access them. Regrads HANG LI

2 1

Reinite breaks the call
by Angelo Sipper 20 Sep '21

20 Sep '21

Hello, I am trying to understand a problem we have with reinvites. When there is a reinvite on the call our provider refuses to receive the reinvite and the call is dropped. The setup is the bellow: UA-> Kamailio1 -> Kamailio2 -> Provider On Kamailio2 we have Topos enabled. Looking on the trace I can see that on the second invite, Kamailio does send out the original contact and not the one that was originally send out to the provider. Is this correct? Is this breaking the call? Bellow the trace and the flow from Kamailio2 until the second reinvite. All reinvites are the same as per the last Thank you for your help! 2021/09/15 16:29:59.420162 50.50.50.181:5060 -> 200.200.200.193:5060 INVITE sip:1234567890@sip.mydomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.b5141bfea33b57c5223aff14ab3b5466.0 Max-Forwards: 67 From: 987654321 <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com> Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 INVITE Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 481 Contact: <sip:btpsh-b5-6141f513-11f69-4@50.50.50.181> v=0 o=FreeSWITCH 1631683257 1631683258 IN IP4 50.50.50.183 s=FreeSWITCH c=IN IP4 50.50.50.183 t=0 0 m=audio 26472 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:26473 a=ptime:20 m=video 26510 RTP/AVP 103 b=AS:1024 a=rtpmap:103 H264/90000 a=rtcp-fb:103 ccm fir a=rtcp-fb:103 ccm tmmbr a=rtcp-fb:103 nack a=rtcp-fb:103 nack pli a=sendrecv a=rtcp:26511 ====================================================================================== 2021/09/15 16:29:59.439009 200.200.200.193:5060 -> 50.50.50.181:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.b5141bfea33b57c5223aff14ab3b5466.0 From: 987654321 <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 INVITE Content-Length: 0 ====================================================================================== 2021/09/15 16:30:02.596613 200.200.200.193:5060 -> 50.50.50.181:5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.b5141bfea33b57c5223aff14ab3b5466.0 From: 987654321 <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 INVITE Contact: <sip:1234567890@200.200.200.193:5060> Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Content-Length: 313 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 27823 25478 IN IP4 200.200.200.193 s=SIP Media Capabilities c=IN IP4 200.200.200.196 t=0 0 m=audio 19958 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 m=video 0 RTP/AVP 103 b=AS:1024 a=rtpmap:103 H264/90000 a=inactive ====================================================================================== 2021/09/15 16:30:06.163783 200.200.200.193:5060 -> 50.50.50.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.b5141bfea33b57c5223aff14ab3b5466.0 From: 987654321 <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:1234567890@200.200.200.193:5060> Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS P-Asserted-Identity: <sip:6979111336@MOBILE.MOBILEPROVIDER.COM;user=phone> Require: timer Supported: timer Session-Expires: 300;refresher=uac Content-Length: 313 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 27823 25478 IN IP4 200.200.200.193 s=SIP Media Capabilities c=IN IP4 200.200.200.196 t=0 0 m=audio 19958 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 m=video 0 RTP/AVP 103 b=AS:1024 a=rtpmap:103 H264/90000 a=inactive ====================================================================================== 2021/09/15 16:30:06.405147 50.50.50.181:5060 -> 200.200.200.193:5060 ACK sip:1234567890@200.200.200.193:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.58e9d8820221b191b49fe0c91299e56f.0 Max-Forwards: 68 From: "987654321" <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 ACK Proxy-Authorization: Digest username="mycustomerid", realm=" sbc1.mydomain.com", nonce="YUH2g2FB9VcCxZ3l2WSvJCj7aUokLhAKksYQ1zfOfyQRWm90h2vJgH8I/B+A", algorithm=MD5, uri="sip:1234567890@sbc1.mydomain.com", response="dc4d3d7e9281c04ecc3e3 63ea162fb" Content-Length: 0 Contact: <sip:btpsh-b5-6141f513-11f69-4@50.50.50.181> ============================================= 2021/09/15 16:32:37.685497 50.50.50.181:5060 -> 200.200.200.193:5060 INVITE sip:1234567890@200.200.200.193:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK4db6.691137b194a14953af928c339e03d60e.0 Via: SIP/2.0/UDP 50.50.50.210;rport=5060;branch=z9hG4bK4db6.4619cd1b550ba7e45a3fbd0321a5c0b0.0 Via: SIP/2.0/UDP 62.1.2.62;received=62.1.2.62;rport=5060;branch=z9hG4bKXDZQF19XUj3QB Max-Forwards: 67 From: "987654321" <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302511 INVITE Contact: <sip:mycustomerid@62.1.2.62:5060 ;transport=udp;gw=1606bb6a-6563-43be-b4dd-4890d524d0e9> User-Agent: Supervoice VPBX Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Proxy-Authorization: Digest username="mycustomerid", realm=" sbc1.mydomain.com", nonce="YUH2g2FB9VcCxZ3l2WSvJCj7aUokLhAKksYQ1zfOfyQRWm90h2vJgH8I/B+A", algorithm=MD5, uri="sip:atpsh-b5-6141f513-11f69-4@50.50.50.181", response="50ed75ca9002 ca76fbfa306fb8a599" Session-Expires: 300;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 348 Remote-Party-ID: "987654321" <sip:987654321@sbc1.mydomain.com >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1631683257 1631683258 IN IP4 50.50.50.183 s=FreeSWITCH c=IN IP4 50.50.50.183 t=0 0 m=audio 26472 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:26473 a=ptime:20 m=video 0 RTP/AVP 103 a=rtpmap:103 H264/90000 |Time | 50.50.50.210 | 200.200.200.193 | | | | 50.50.50.181 | |0.000024 | INVITE SDP (g711A g7 | |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> Call-ID:dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq:41302510 | |(5060) ------------------> (5060) | | |0.001523 | 100 trying -- your c | |SIP Status 100 trying -- your call is important to us | |(5060) <------------------ (5060) | | |0.011760 | | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302510 | | |(5060) ------------------> (5060) | |0.030607 | | 100 Trying| |SIP Status 100 Trying | | |(5060) <------------------ (5060) | |2.690501 | | 183 Session Progress |SIP Status 183 Session Progress | | |(5060) <------------------ (5060) | |2.708868 | 183 Session Progress | |SIP Status 183 Session Progress | |(5060) <------------------ (5060) | | |3.188211 | | 183 Session Progress |SIP Status 183 Session Progress | | |(5060) <------------------ (5060) | |3.199099 | 183 Session Progress | |SIP Status 183 Session Progress | |(5060) <------------------ (5060) | | |6.755381 | | 200 OK SDP (g711A te |SIP Status 200 OK | | |(5060) <------------------ (5060) | |6.819017 | 200 OK SDP (g711A te | |SIP Status 200 OK | |(5060) <------------------ (5060) | | |6.973845 | ACK | | |SIP Request INVITE ACK 200 CSeq:41302510 | |(5060) ------------------> (5060) | | |6.996745 | | ACK | |SIP ACK From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302510 | | |(5060) ------------------> (5060) | |158.269723| INVITE SDP (g711A g7 | |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> Call-ID:dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq:41302511 | |(5060) ------------------> (5060) | | |158.275396| 100 trying -- your c | |SIP Status 100 trying -- your call is important to us | |(5060) <------------------ (5060) | | |158.277105| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |158.753781| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |159.753493| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |161.753502| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |165.753630| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |169.754980| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |173.753719| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |177.753676| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |181.753657| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |185.753503| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |188.254337| 503 Service not avai | |SIP Status 503 Service not available | |(5060) <------------------ (5060) | | |188.255085| ACK | | |SIP Request INVITE ACK 200 CSeq:41302511 | |(5060) ------------------> (5060) | | |188.326552| BYE | | |SIP Request BYE CSeq:41302512 | |(5060) ------------------> (5060) | | |188.342106| | BYE | |SIP Request BYE CSeq:41302512 | | |(5060) ------------------> (5060) | |188.373697| | 200 OK | |SIP Status 200 OK | | |(5060) <------------------ (5060) | |188.379670| 200 OK | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | Kind Regards, Angelo

4 4

Re: [SR-Users] Error in ul_insert_ucontact() function of usrloc module
by hamed r 20 Sep '21

20 Sep '21

hello Ben I was using development functions and the C code of this function did not exist. I needed to manually add a contact to kamailio but this does not work and I used the HTable module to solve the problem. Thanks for the tips

1 0

Slack channel
by Sergio Carvalho 20 Sep '21

20 Sep '21

Hi everyone, How can I join the Kamailio slack channel? Is it open to the community? Best regards, Sergio Carvalho

3 2

Error in ul_insert_ucontact() function of usrloc module
by hamed r 18 Sep '21

18 Sep '21

The user is not added using the ul_insert_ucontact() function : ul_insert_ucontact("sip:900@192.168.51.138:1234", "< sip:900@192.168.51.138:1234>", "303", -1, "a2ecd821d4464d75851b73c35442b6ad", 2222, "0", "1", "MicroSIP/3.20.6", "udp: 192.168.51.106:5060"); And the following error is displayed in the syslog: Sep 18 14:36:48 srv139937 kamailio: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 838, column 148: Too many arguments#012 Sep 18 14:36:48 srv139937 kamailio: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 838, column 148: '('')' expected (function call) Sep 18 14:36:48 srv139937 kamailio: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 838, column 148: bad command: missing ';'? Sep 18 14:36:48 srv139937 kamailio: CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 838, column 150-152: bad command Sep 18 14:36:48 srv139937 kamailio: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 838, column 153: bad command Sep 18 14:36:48 srv139937 kamailio: CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 838, column 155-171: bad command Sep 18 14:36:48 srv139937 kamailio: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 838, column 172: bad command Sep 18 14:36:48 srv139937 kamailio: CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 838, column 174-198: bad command Sep 18 14:36:48 srv139937 kamailio: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 838, column 199: bad command Sep 18 14:36:48 srv139937 kamailio: ERROR: <core> [core/cfg.y:3532]: yyparse(): function used inside params of another function: is_method what shoud i do?

2 1

Re: [SR-Users] problem with LRKProxy module
by hamed r 18 Sep '21

18 Sep '21

Thanks Mojtaba, problem was solved. I performed LRKProxy first, then Kamailio.

1 0

BLF when phone boots up
by Jerry Kendall 17 Sep '21

17 Sep '21

Hi, I have looked at lots of postings but can't seem to find the answer I seek. I have 6 phones (Grandstream GXP2170, etc) connected to Kamailio 5.2.1 running default config file with changes per https://kb.asipto.com/kamailio:presence:k43-blf On x301, I have the 3 BLFs setup to monitor x302, x303 and x304. If I dial 303 from 301, the BLFs work (flash, red, then green) as expected. But, when the phone initially boots up, the BLFs are all dead - until the first PUBLISH event. So, the question is, how do I get the BLFs to become valid and with the correct status when the phone boots up??? regards, Jerry

1 0

listen directive during runtime
by Tihomir Culjaga 17 Sep '21

17 Sep '21

Hi, is it possible to add a listen directive during kamailio runtime ? The reason I'm asking this is in a situation when I add a new peering trunk (from a different network segment), I need to add a network interface to the kamailio server and then add a specific listen directive for that interface to reference the socket by name. listen=udp:10.1.1.11 name "mySN1" so i can further reference to a specific socket name when routing SIP messages. Please advise, Regards, Tihomir.

2 1

Re: [SR-Users] DMQ module issues
by Zé Santos 17 Sep '21

17 Sep '21

I don't think you understood. The issue is in v5.2.5, and I tried to update kamailio but it still remains. I honestly think this is a machine problem, since I can't even send the notifications to 0.0.0.0:5090. It is very weird. Best Regards, José Santos

2 3

xmlrpc always returns NO RESULT
by 岩間理佐 17 Sep '21

17 Sep '21

Hi, I have a question regarding the XMLRPC commands in the siremis GUI. 【Environment】・Centos7 ・kamailio 5.5.2 ・siremis 5.3 （installed by git clone https://github.com/asipto/siremis siremis-5.3.x）・php: $ rpm -qa | grep php php-process-5.4.16-48.el7.x86_64 php-common-5.4.16-48.el7.x86_64 php-mysqlnd-5.4.16-48.el7.x86_64 php-pdo-5.4.16-48.el7.x86_64 php-xml-5.4.16-48.el7.x86_64 php-cli-5.4.16-48.el7.x86_64 php-5.4.16-48.el7.x86_64 php-pear-1.9.4-22.el7.noarch ・apache 2.4.6 When I execute commands in command service → XMLPRC from siremis GUI, it always shows no result. It looks like the command is not being executed. I also executed these commands on server, and there are no problem. There is no error in apache log and kamailio is also working. This is what I have in /etc/kamailio/kamailio.cfg #!KAMAILIO #!define WITH_XMLRPC #!define WITH_DEBUG #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" # - flags #FLT_ - per transaction (message) flags #FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define WITH_MYSQL #!define WITH_NAT ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_JSONRPC loadmodule "xhttp.so" #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 disable_tcp=yes auto_aliases=no port=5060 listen=udp:172.23.0.246:5060 advertise 52.69.75.254:5060 sip_warning=no ####### Modules Section ######## # set module path #mpath="/usr/local/lib/kamailio/modules/" #loadmodule "db_mysql.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "uac.so" loadmodule "sqlops" loadmodule "rtimer" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") modparam("jsonrpcs", "dgram_mode", 0666) modparam("jsonrpcs", "dgram_user", "kamailio") modparam("jsonrpcs", "dgram_group", "kamailio") modparam("jsonrpcs", "transport", 0) # ----- xmlrpc params ----- #modparam("xmlrpc", "route", "XMLRPC") #modparam("xmlrpc", "url_match", "^/RPC") # ----- rr params ----- modparam("rr", "append_fromtag", 1) # ----- R Timer ----- modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") # ----- acc params ----- modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) modparam("tm", "auto_inv_100_reason", "Trying") # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_from", "sip:proxy@52.69.75.254") modparam("dispatcher", "ds_ping_interval", 60) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_timer_mode", 1) modparam("dispatcher", "xavp_dst_mode", 0) modparam("dispatcher", "ds_ping_latency_stats", 1) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_threshold", 10) modparam("dispatcher", "ds_inactive_threshold", 10) modparam("dispatcher", "ds_ping_reply_codes","class=2;code=403;code=404;code=484;class=3") # ----- uac params ----- modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_contact_addr", "52.69.75.254") modparam("uac", "reg_db_table", "uacreg") # ----- mysql ----- modparam("sqlops", "sqlcon","cb=>mysql://kamailio:kamailiorw@localhost/kamailio") ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) { return; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); } if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; sl_send_reply("404", "No registrar"); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "11")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500|486") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } I have tried my best to do some researches about it, but still stuck. Any help is appreciated. Thank you!

2 1

problem with LRKProxy module
by hamed r 16 Sep '21

16 Sep '21

Hello I have problem with lrkproxy module. First, I installed the packages from the link below : https://github.com/mojtabaesfandiari/pylrkproxy And the service was properly listened to the local ip (192.168.51.106:22333 ). In the next step, the parameters were added to the kamalio.cfg file : loadmodule "lrkproxy.so" modparam("lrkproxy", "lrkproxy_sock", "udp:192.168.51.106:22333") I also changed the /etc/pylrkproxy/pylrkproxy.ini to set local ip. But when I make a call, the ip inside the sdp not change. And the following error is displayed in the syslog: Sep 4 07:18:05 debian /usr/local/sbin/kamailio[1724]: INFO: {1 2 INVITE NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1507]: lrkproxy_force(): Here is SIP_REQUEST & METHOD_INVITE Sep 4 07:18:05 debian /usr/local/sbin/kamailio[1724]: ERROR: {1 2 INVITE NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1084]: select_lrkp_node(): script error -no valid set selected Sep 4 07:18:05 debian /usr/local/sbin/kamailio[1724]: ERROR: {1 2 INVITE NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1513]: lrkproxy_force(): can't ping any lrk proxy right now. Sep 4 07:18:07 debian /usr/local/sbin/kamailio[1723]: INFO: {2 2 INVITE NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1625]: lrkproxy_force(): Here is SIP_REPLY of METHOD_INVITE Sep 4 07:18:07 debian /usr/local/sbin/kamailio[1723]: ERROR: {2 2 INVITE NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1630]: lrkproxy_force(): No found entry in hash table

2 1

DMQ module issues
by Zé Santos 16 Sep '21

16 Sep '21

Hi guys, I am currently using kamailio v5.2.5 and running it inside a container. I already attempted to upgrade the kamailio to the latest version however the problem persists. This is the core dump that I got from using gdb: Core was generated by `/usr/sbin/kamailio -f > /etc/kazoo/kamailio/kamailio.cfg -m 512 -M 64 -x tlsf -w'. > Program terminated with signal 11, Segmentation fault. > #0 0x000000000071c63c in select_cfg_var (res=0x7ffd57d49b10, > s=0x7f2649395040, msg=0x7ffd57d4c340) at core/cfg/cfg_select.c:211 > 211 i = *(int *)p; It seems that the problem is with this line: modparam("dmq", "notification_address", "sip:DMQ_NOTIFY_ADDRESS") > Also, I only have this issue while running on an AWS instance, I have the same setup on my local computer, however kamailio is able to start correctly. While kamailio is starting, I have this line #!substdef "!DMQ_NOTIFY_ADDRESS!10.10.10.10:5090!g" It seems that after DMQ is able to obtain the IP address, when it comes to actually sending the message, it is interpreting the ip address of the DMQ_NOTIFY_ADDRESS as an integer and then it gives out a segmentation fault and the module stops. Another interesting thing is that if the DMQ_NOTIFY_ADDRESS is the MY_IP_ADDRESS variable instead of the 10.10.10.10, it seems to work properly. #!substdef "!DMQ_NOTIFY_ADDRESS!MY_IP_ADDRESS:5090!g" It also doesn't seem related to the ip being in a variable, since I already tried using a DNS record, and once again, it is able to obtain the IPs from the DNS record, and the same issue applies. Has anyone ever seen anything like this? I can't find any evidence of this actually happening. Thank you kindly for your time. Best Regards, José Santos

3 3

Best practice for Kamailio "cluster" front for many TLS connections
by Angelo Sipper 15 Sep '21

15 Sep '21

Hello, We are thinking to use kamailio to fully support our current voip service. I would like to know what is the best practice to use kamailio for several thousands for clients that connect to us via TLS. We would like to have a front "cluster" to be able to handle all these connections in balanced way. All clients register to us and the do calls that we statefully forward to the provider. On our test we have already created a single kamailio that successfully accepts and handles registration, do the control and routing to next provider with dispatcher and also do the accounting with cdrs. We want to separate this controlling, routing, accounting and dispatching from the front end and create a front "cluster" to offload the TLS and also be redundant to as many connections by adding more kamailio's. What would be the best kamailio module and topology as solution to this requirement? Thank you in advance. Kind Regards, Angelo

7 12

Fwd: Catching an overall timeout after dropping replies
by Marrold 15 Sep '21

15 Sep '21

Hi, This is partially related to a previous thread - https://lists.kamailio.org/pipermail/sr-users/2021-August/113128.html We are using the TSILO module to suspend transactions whilst we wait for mobile app users to register after receiving a push notification. In the time we are waiting for the app to register, other UAs might reject the call (DND, Busy, etc). To avoid this failure response being relayed to the caller, we use t_drop_replies in the failure_route. Eventually if all devices have rejected the call Kamailio times out and sends a 408 Request Timeout. Is there any way we can catch this error, and replace it with the last response we got from an actual UA? t_is_expired() and t_branch_timeout() don't seem to apply as they're on the *called* side. Is there another option to catch this overall timeout? As a side note, failure_route is quite limited in the actions that can take place there. Is it safe to access variables and update a htable? Thanks Matthew

1 0

Re: [SR-Users] Empty Subnets in Permissions Module
by Henning Westerholt 14 Sep '21

14 Sep '21

Hello, please keep the list in CC. Let’s look into the two issues one by one: 1) I had to explicitly configure the parameter: modparam("permissions", "mask_col", "mask") Although the documentation suggests "mask" is the default - the JSON output from "kamctl address dump" did not output this value on K5.5. (On K5.3 it outputted properly) Do you get an error if you do not specify the mask_col like this, or something else? From the source code the default should be “mask”. When I run the "kamcmd permissions.subnetDump" on Kamailio 5.3, it returns everything as expected - including the 0.0.0.0/0<http://0.0.0.0/0> subnets. However, when running the same commands on Kamailio 5.5, it only returns a small subset (of only 20) subnets/groups - and the selection does not appear to follow a logical selection criteria. Additionally, it does not return any groups with a 0.0.0.0/0<http://0.0.0.0/0> subnet either. It seems that the behaviour has changed regarding the “0” subnet, checkout the docs: https://kamailio.org/docs/modules/devel/modules/permissions.html#permission… It will convert them to 32/128 respectively. Can you see a 0.0.0./32 in your dump? This was changed in commit f376c82a9f8 during an extension for text files. Maybe Daniel can comment here if this was done by purpose. Otherwise, you can open an issue on our tracker about it. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Tom Dworakowski <dworakowski.tom(a)gmail.com> Sent: Tuesday, September 14, 2021 5:00 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Empty Subnets in Permissions Module Hello Henning, Thank you for looking into this for me. I made two interesting discoveries this morning: 1) I had to explicitly configure the parameter: modparam("permissions", "mask_col", "mask") Although the documentation suggests "mask" is the default - the JSON output from "kamctl address dump" did not output this value on K5.5. (On K5.3 it outputted properly) 2) When I run the "kamcmd permissions.subnetDump" on Kamailio 5.3, it returns everything as expected - including the 0.0.0.0/0<http://0.0.0.0/0> subnets. However, when running the same commands on Kamailio 5.5, it only returns a small subset (of only 20) subnets/groups - and the selection does not appear to follow a logical selection criteria. Additionally, it does not return any groups with a 0.0.0.0/0<http://0.0.0.0/0> subnet either. From my logs - I have noted this: Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4353, 0.0.0.0, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <3769, 0.0.0.0, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4355, 0.0.0.0, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4359, 0.0.0.0, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <1955, 84.XX.XX.66, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4361, 91.X.X.231, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4361, 91.X.X.33, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4361, 91.X.X.34, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4363, 80.X.X.25, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4363, 85.X.X.124, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4363, 212.X.X.19, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4365, 0.0.0.0, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4367, 0.0.0.0, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <4371, 0.0.0.0, 0> inserted into address hash table Sep 14 03:07:25 webrtc kamailio[5407]: 0(5407) DEBUG: permissions [address.c:118]: reload_address_insert(): Tuple <3991, 0.0.0.0, 0> inserted into address hash table At the moment of querying group id 3983 (where there is only 0.0.0.0/0<http://0.0.0.0/0>), the function returns false: DEBUG: permissions [address.c:671]: allow_source_address(): looking for <3983, [IPv4 in hex, reversed octet order], 62281> However, None of those addresses appear in the "kamcmd permissions.subnetDump" output. Moreover, if "my" group has the address 0.0.0.0/0<http://0.0.0.0/0> listed as an approved address - it will fail the test; but if I register 0.0.0.0/1<http://0.0.0.0/1> it will let me through (as my IP is < 128.0.0.0), kamcmd permissions.subnetDump will display this address. My thoughts are that there might be another table that is not being populated - or there is a filter during the import that either drops 0.0.0.0/0<http://0.0.0.0/0> or filters it out completely? Regards, Tom On Tue, Sep 14, 2021 at 4:10 AM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Tom, I’ve done a quick comparison of the main function and the called function. On a first view it looked identically, but I looked only a few levels deep. Do you have maybe some means to reproduce this on a test system? Then it would be probably interesting to look to the DEBUG logging of this cases. Maybe you can compare if you spot some obvious differences from the logic. Cheers, Henning From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Tom Dworakowski Sent: Tuesday, September 14, 2021 4:10 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Empty Subnets in Permissions Module Greetings all! I have two deployments of Kamailio: one running version 5.3 and one 5.5 with practically identical configurations, same (MySQL and REDIS) data sources. We have customers that we assign an ACL "group" to, where the ID of this group resolves to records in the "address" table in our MySQL database - using the "grp" field. On the box running Kamailio 5.5, we have noticed that if a group has ip_addr=0.0.0.0, mask=0, port=0 - and we try to run the allow_source_address() - it will return false, thus failing this phase of the authentication process. However, on Kamailio 5.3 we are not seeing this issue, i.e. if a customer is assigned a group where the ACL is 0.0.0.0/0<http://0.0.0.0/0> - it will let him through. Has something changed that I'm not aware of? Any suggestions on how to resolve this? My best, Tom

2 1

Capturing response from uac_reg module
by Ross McKillop 14 Sep '21

14 Sep '21

Hi, What's the best way to capture the response from REGISTER requests automatically initiated by the uac_reg module? I was looking to use event_route[uac:reply] {} but that is only triggered if the evroute flag is set on the initial call made to uac_req_send() Is there a way to set this globally so that uac:reply is called for all requests initiated for peers in the uacreg table or a better way to capture the response? Best, Ross

3 4

Kamailio 5.4.3 crash at SSL error
by Igor Olhovskiy 14 Sep '21

14 Sep '21

Hello. Kamailio 5.4.3 crashes at CentOS 7 with following error logs Sep 13 13:15:11 /usr/local/sbin/kamailio[44592]: ERROR: <core> [core/tcp_read.c:1500]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f3b93734cf8 r: 0x7f3b93734e20 (-1) Sep 13 13:15:12 /usr/local/sbin/kamailio[44594]: NOTICE: tls [tls_domain.c:747]: sr_ssl_ctx_info_callback(): SSL handshake started Sep 13 13:15:12 kernel: traps: kamailio[44594] general protection ip:7f3ba8fb3c09 sp:7ffe2c3aff70 error:0 in libssl.so.1.0.2k[7f3ba8f87000+67000] Sep 13 13:15:12 /usr/local/sbin/kamailio[44597]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 38 Sep 13 13:15:12 /usr/local/sbin/kamailio[44597]: CRITICAL: <core> [core/pass_fd.c:85]: recv_all(): 1st recv on 41 failed: Connection reset by peer Sep 13 13:15:12/usr/local/sbin/kamailio[44597]: CRITICAL: <core> [core/tcp_main.c:3543]: handle_tcp_child(): read from tcp child 2 (pid 44594, no 19) Connection reset by peer [104] Sep 13 13:15:12/usr/local/sbin/kamailio[44561]: ALERT: <core> [main.c:777]: handle_sigs(): child process 44594 exited by a signal 11 Sep 13 13:15:12/usr/local/sbin/kamailio[44561]: ALERT: <core> [main.c:780]: handle_sigs(): core was generated Sep 13 13:15:12 /usr/local/sbin/kamailio[44561]: INFO: <core> [main.c:802]: handle_sigs(): terminating due to SIGCHLD I'm pretty sure these logs are mostly not usable, but maybe they will point to some bugs. To add, this happened just occasionally and I can't reproduce it at the moment. -- Regards, Igor

2 3

Manage blacklisting on network errors
by Marat Gareev 14 Sep '21

14 Sep '21

Hello! I have a simple config for routing requests with failover and blacklisting on 408, 480 and 503 codes from servers. This is a part of config: # Wrapper for relaying requests route[RELAY] { # the base event routes t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); t_on_failure("MANAGE_FAILURE"); if (!t_relay()) { sl_reply_error(); } exit; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_NOTICE", "$rr ($rs) [$cs] ($ci) $si:$sp - $ua\n"); if ( t_check_status("(503)|(408)|(480)") ) { xlog("L_WARN", "Server will be blacklisted: $si:$sp ($rs)\n"); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { if ( !t_check_status("(503)|(408)|(480)") ) { exit; } ds_mark_dst("IP"); # blacklist if (t_is_canceled()) exit; if (!ds_next_domain()) { send_reply("503", "Service Unavailable"); exit; } route(RELAY); } If there is a timeout or network error on the server side, it is blacklisted. How can such cases be managed and how can they be logged? I didn't find this on the module page: https://www.kamailio.org/docs/modules/stable/modules/tm.html

2 2

Question about suspected memory leak presence module
by Albert Petit 14 Sep '21

14 Sep '21

Hi, I am suspecitng presence module might be having a memory leak on a production server used by an end user (not have access to it). The output of command kamcmd mod.stats presence shm is Module: presence { mem_copy_subs(148): 25720 mem_copy_subs_noc(214): 1672201904 mem_copy_subs_noc(251): 678406560 add_event(181): 88 shm_copy_event(57): 40 shm_copy_event(50): 312 add_event(156): 40 add_event(149): 456 new_shtable(66): 5767168 new_shtable(53): 262144 init_evlist(289): 16 Total: -1938302848 } I am not sure what mem_copy_subs_noc is meaning. And final output is negative, SHM is incfreasing all the time but not sure of the root cause is this a reasonable hint that a memory leak is related to presence?

2 3

Empty Subnets in Permissions Module
by Tom Dworakowski 14 Sep '21

14 Sep '21

Greetings all! I have two deployments of Kamailio: one running version 5.3 and one 5.5 with practically identical configurations, same (MySQL and REDIS) data sources. We have customers that we assign an ACL "group" to, where the ID of this group resolves to records in the "address" table in our MySQL database - using the "grp" field. On the box running Kamailio 5.5, we have noticed that if a group has ip_addr=0.0.0.0, mask=0, port=0 - and we try to run the allow_source_address() - it will return false, thus failing this phase of the authentication process. However, on Kamailio 5.3 we are not seeing this issue, i.e. if a customer is assigned a group where the ACL is 0.0.0.0/0 - it will let him through. Has something changed that I'm not aware of? Any suggestions on how to resolve this? My best, Tom

2 1

Kamailio dispatcher -> FreeSWITCH transfers handling.
by Тимур Ситдиков 11 Sep '21

11 Sep '21

Hi list! :-) I'm a total newbie in Kamailio. Trying to study) Sorry for lots of text, I'm simply don't know how to explain my questions shorter :-D I want to configure Kamailio as dispatcher, registrar and NAT traversal in front of FS farm. I've managed to set up Kamailio and Fswitches. Now I need to handle "special cases" with confs, REFER transfers etc Giovanni Maruzzelli once presented solution for this, with following logic: ------------------------------------------------------------------------------------------------------- 1) Kamailio by default distribute calls to various FSs - dispatch or load_balance 2) Each time Kamailio sends a call to a FS, it writes down which destination "number" is going to which FS server and from which UserID 3) if another call comes for the same (or related) destination, or is originated from the same UserID then Kamailio sends the call to that same FS server, bypassing default distribution algorithm ------------------------------------------------------------------------------------------------------- Sounds good and helpful, but I'm stuck :-( *Can someone please share an example or point me how and in what form should I write down call info,and how can I control it (delete on BYE or CANCEL)?* Actually there are 2 questions It seems to me that recordings can be made using HTABLEs but here's a problem: ===================================================================================== ============================================================================================= 1) PROBLEM WITH WRITING TO HTABLE ON ATT TRANSFER After the dispatcher sets $du - I write an htable entry $sht(fsdest=>$fU:$rU) = $du; Existing entries are removed on catching 'BYE' for normal clearing, and with failure_route for all failures. BYE logic is: IN WITHINNDLG route if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { if (sht_match_name("fsdest", "eq", "$fU:$rU")) { $sht(fsdest=>$fU:$rU) = $null; //BYE A->B } else { $sht(fsdest=>$rU:$fU) = $null; //BYE A<-B } } handle_ruri_alias(); route(RELAY); } } Another entry is setting up if blind transfer is made. If INVITE is coming from one of FSs, and have 'Referred-By' field. IN DIALOG route: if (is_present_hf("Referred-By")) { $sht(fsdest=>$fU:$rU) = $si; } For now it's working. Entries are written and deleted properly. *Problem with Attended transfer.* 1) A --> B = htable entry A:B=xxxx 2) A --> C = htable entry A:C=xxxx(checked and set same FS addr because $fU is present (If only it's worked as it should)) 3) Att transfer (Refer $rU B, referred-by - A. Refer-To - C) B --> C = all related entries deleted from table over BYE logic. 4) No entry B --> C (no INVITE was sent, bridged by REFER) So, logic for check active calls in htable is broken now, because we don't have entry B-->C *How can I handle writing to htable in Attended transfer case?* If another Invite will come FROM or TO B or C = there will be no htable entry. I'm not sure this is even a correct way. Surely missing something. ============================================================================================= ============================================================================================= 2) PROBLEM WITH IDENTIFYING TABLE ENTRIES Anyway, the main problem is with identifying those htables to set proper $du. If INVITE is coming TO FSs I need to check if caller ID ($fU) or dest ($rU) is present in htables. Form I need to write to hash must be like $fU:$rU Entry must be unique or it will be: 1) rewrited anytime a new invite is going to the same dest. 2) deleted anytime BYE or failure will come eg: If $sht(fsdest=>$rU) = $dd 1) 1000 --> 3200 = htable entry 3200 = xxxx 2) 1005 --> 3200 = htable entry rewriten to the same 3200 = xxxx When 1000 or 1005 leaving conf, it sends BYE and removes entry. So, next call to 3200 may be sent to the wrong FS. Ok, we have a unique entry ($fU:$rU). Checking for active destination: if (sht_match_name("fsdest", "re", ":$rU")) {} If there's an htable with $rU in key - we got a match. Now we need to assign a value from htable to $du. But, I don't know how to get the name of this entry. There's always different $fU: (or $rU if we check for the same caller ID ($fU)). eg: 1000 --> 3200 = htbale 1000 --> 3200 New call 1005 --> 3200 Check sht_match name ":3200" - got match We need to set: $du = $sht(fsdest=>$fU:$rU) But it will be $sht(fsdest=>1005:3200) and we don't have this entry. We need (fsdest=>1003:3200) But i don't see any way to get needed $fU. *Is there any way to search or identify these tables?* ============================================================================================= ============================================================================================= Can someone help me with this, please? :-) Or maybe there is a better way to do this? *My kamailio.cfg routing block in pastebin, if neededhttps://pastebin.com/1PZ0QSDW <https://pastebin.com/1PZ0QSDW>* Thank you! Regards, Tim

4 5

Info: Kamailio World 2021 starting soon - live stream
by Daniel-Constantin Mierla 10 Sep '21

10 Sep '21

Hello, to sync better on time, Kamailio World 2021 online edition is starting in about 2 hours from now (12:50 UTC Welcome and 13:00UTC the first session) - the agenda is available at: * https://kamailioworld.com/k09-online/ In short: participation is free, no registration is needed. The sessions are live streamed to YouTube KamailioWorld channel: * https://www.youtube.com/c/KamailioWorld/ YouTube chat next to the live stream can be used for addressing questions to speakers -- we will monitor it and ask the speaker inside the video conferencing room. For community interaction via text chatting, you can join the Matrix room #kamailio:matrix.kamailio.dev : * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev Questions for speakers can also be addressed in this chat room. If anyone wants to participate in the live video conferencing room, the URL is: * https://meet.kamailioworld.com/live Just note that we may have to limit the number of participants in the live session in order to ensure proper streaming quality. See you soon online! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

5 9

RPC usrloc ul.add problem
by Pepelux 09 Sep '21

09 Sep '21

Hi all I have problems adding a new location by RPC command. *# kamailio -v* version: kamailio 5.5.0 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 8.3.0 A simple example, adding an AOR which expires in 1 hour: *# date ; kamcmd ul.add kam_location 1002-test1(a)sipserver.com <1002-test1(a)sipserver.com> sip:1002-test1@192.168.89.82 <sip%3A1002-test1(a)192.168.89.82> 3600 0.0 . 0 0 0 .* Wed 08 Sep 2021 01:58:00 PM CEST Location record has been inserted correctly: *# date ; kamcmd ul.dump | grep test -A24* Wed 08 Sep 2021 01:58:02 PM CEST AoR: 1002-test1(a)sipserver.com HashID: -597689288 Contacts: { Contact: { Address: sip:1002-test1@192.168.89.82 Expires: 3598 Q: 0.000000 Call-ID: dfjrewr12386fd6-343(a)kamailio.rpc CSeq: 1 User-Agent: kamailio SIP Router - RPC Server Received: [not set] Path: [not set] State: CS_SYNC Flags: 0 CFlags: 0 Socket: [not set] Methods: 0 Ruid: ulcx-61389f47-2e44-01 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: 0 Keepalive: 0 Last-Keepalive: 1631102280 KA-Roundtrip: 0 Last-Modified: 0 } } } But a few seconds later, the location record has gone: * # date ; kamcmd ul.dump | grep test -A24* Wed 08 Sep 2021 01:58:26 PM CEST In kamailio logs I can see that: * Sep 8 13:58:00 sipserver /usr/sbin/kamailio[11844]: DEBUG: usrloc [ul_rpc.c:637]: ul_rpc_add(): path == 0 -> unset Sep 8 13:58:00 sipserver /usr/sbin/kamailio[11844]: DEBUG: usrloc [ul_rpc.c:652]: ul_rpc_add(): ret: 10 table:kam_location aor:1002-test1@sipserver.com <aor%3A1002-test1(a)sipserver.com> contact:sip:1002-test1@192.168.89.82 <contact%3Asip%3A1002-test1(a)192.168.89.82> expires:3600 dtemp:0.000000 path: flags:0 bflags:0 methods:0 Sep 8 13:58:00 sipserver /usr/sbin/kamailio[11844]: DEBUG: usrloc [ul_rpc.c:660]: ul_rpc_add(): q:0 Sep 8 13:58:00 sipserver /usr/sbin/kamailio[11844]: DEBUG: usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:58:00 sipserver /usr/sbin/kamailio[11844]: DEBUG: usrloc [ucontact.c:1868]: uldb_insert_attrs(): no location attributes* Sep 8 13:58:03 sipserver /usr/sbin/kamailio[11827]: DEBUG: {1 81337 REGISTER 1955770655-18357-1(a)BJC.BGI.B.BBF REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:03 sipserver /usr/sbin/kamailio[11827]: DEBUG: {1 81337 REGISTER 1955770655-18357-1(a)BJC.BGI.B.BBF REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:03 sipserver /usr/sbin/kamailio[11827]: DEBUG: {1 81337 REGISTER 1955770655-18357-1(a)BJC.BGI.B.BBF REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:58:04 sipserver /usr/sbin/kamailio[11829]: DEBUG: {1 141 REGISTER 1aa357700bd0696702c40340595d89e7(a)185.99.186.215 REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:04 sipserver /usr/sbin/kamailio[11829]: DEBUG: {1 141 REGISTER 1aa357700bd0696702c40340595d89e7(a)185.99.186.215 REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:04 sipserver /usr/sbin/kamailio[11829]: DEBUG: {1 141 REGISTER 1aa357700bd0696702c40340595d89e7(a)185.99.186.215 REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:58:06 sipserver /usr/sbin/kamailio[11852]: DEBUG: {1 114 REGISTER 5ne5ebv674tptk0p1uj4gg REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:06 sipserver /usr/sbin/kamailio[11852]: DEBUG: {1 114 REGISTER 5ne5ebv674tptk0p1uj4gg REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:06 sipserver /usr/sbin/kamailio[11852]: DEBUG: {1 114 REGISTER 5ne5ebv674tptk0p1uj4gg REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:58:23 sipserver /usr/sbin/kamailio[11828]: DEBUG: {1 12484 REGISTER 1382185578-19234-1(a)BJC.BGI.B.BGC REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:23 sipserver /usr/sbin/kamailio[11828]: DEBUG: {1 12484 REGISTER 1382185578-19234-1(a)BJC.BGI.B.BGC REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:23 sipserver /usr/sbin/kamailio[11828]: DEBUG: {1 12484 REGISTER 1382185578-19234-1(a)BJC.BGI.B.BGC REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes * Sep 8 13:58:23 sipserver /usr/sbin/kamailio[11835]: DEBUG: usrloc [urecord.c:377]: wb_timer(): tcp connection has been lost, expiring contact sip:1002-test1@192.168.89.82 <sip%3A1002-test1(a)192.168.89.82> Sep 8 13:58:23 sipserver /usr/sbin/kamailio[11835]: DEBUG: usrloc [urecord.c:387]: wb_timer(): Binding '1002-test1(a)sipserver.com <1002-test1(a)sipserver.com>','sip:1002-test1@192.168.89.82 <sip%3A1002-test1(a)192.168.89.82>' has expired Sep 8 13:58:23 sipserver /usr/sbin/kamailio[11835]: DEBUG: usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes* Sep 8 13:58:25 sipserver /usr/sbin/kamailio[11830]: DEBUG: {1 185389 REGISTER 1487655759-19321-1(a)BJC.BGI.B.BBB REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:25 sipserver /usr/sbin/kamailio[11830]: DEBUG: {1 185389 REGISTER 1487655759-19321-1(a)BJC.BGI.B.BBB REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:25 sipserver /usr/sbin/kamailio[11830]: DEBUG: {1 185389 REGISTER 1487655759-19321-1(a)BJC.BGI.B.BBB REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:58:33 sipserver /usr/sbin/kamailio[11853]: DEBUG: {1 3360 REGISTER pgmirc5oirbekeaj3tmrf7 REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:33 sipserver /usr/sbin/kamailio[11853]: DEBUG: {1 3360 REGISTER pgmirc5oirbekeaj3tmrf7 REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:33 sipserver /usr/sbin/kamailio[11853]: DEBUG: {1 3360 REGISTER pgmirc5oirbekeaj3tmrf7 REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:58:44 sipserver /usr/sbin/kamailio[11829]: DEBUG: {1 5645 REGISTER 89744034-6071-1(a)BJC.BGI.A.BJ REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:44 sipserver /usr/sbin/kamailio[11829]: DEBUG: {1 5645 REGISTER 89744034-6071-1(a)BJC.BGI.A.BJ REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:44 sipserver /usr/sbin/kamailio[11829]: DEBUG: {1 5645 REGISTER 89744034-6071-1(a)BJC.BGI.A.BJ REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:58:44 sipserver /usr/sbin/kamailio[11830]: DEBUG: {1 5683 REGISTER 1974024198-16543-1(a)BJC.BGI.B.BDE REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:44 sipserver /usr/sbin/kamailio[11830]: DEBUG: {1 5683 REGISTER 1974024198-16543-1(a)BJC.BGI.B.BDE REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:44 sipserver /usr/sbin/kamailio[11830]: DEBUG: {1 5683 REGISTER 1974024198-16543-1(a)BJC.BGI.B.BDE REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:58:50 sipserver /usr/sbin/kamailio[11829]: DEBUG: {1 144820 REGISTER 381139623-5062-1(a)BJC.BGI.B.BAD REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:50 sipserver /usr/sbin/kamailio[11829]: DEBUG: {1 144820 REGISTER 381139623-5062-1(a)BJC.BGI.B.BAD REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:50 sipserver /usr/sbin/kamailio[11829]: DEBUG: {1 144820 REGISTER 381139623-5062-1(a)BJC.BGI.B.BAD REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:58:59 sipserver /usr/sbin/kamailio[11828]: DEBUG: {1 84973 REGISTER 467700456-5060-1(a)BJC.BGI.B.BCI REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:58:59 sipserver /usr/sbin/kamailio[11828]: DEBUG: {1 84973 REGISTER 467700456-5060-1(a)BJC.BGI.B.BCI REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:58:59 sipserver /usr/sbin/kamailio[11828]: DEBUG: {1 84973 REGISTER 467700456-5060-1(a)BJC.BGI.B.BCI REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:59:01 sipserver /usr/sbin/kamailio[11854]: DEBUG: {1 116 REGISTER 5ne5ebv674tptk0p1uj4gg REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:59:01 sipserver /usr/sbin/kamailio[11854]: DEBUG: {1 116 REGISTER 5ne5ebv674tptk0p1uj4gg REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:59:01 sipserver /usr/sbin/kamailio[11854]: DEBUG: {1 116 REGISTER 5ne5ebv674tptk0p1uj4gg REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes Sep 8 13:59:04 sipserver /usr/sbin/kamailio[11830]: DEBUG: {1 81338 REGISTER 1955770655-18357-1(a)BJC.BGI.B.BBF REGISTER} usrloc [ucontact.c:1801]: uldb_delete_attrs_ruid(): trying to delete location attributes Sep 8 13:59:04 sipserver /usr/sbin/kamailio[11830]: DEBUG: {1 81338 REGISTER 1955770655-18357-1(a)BJC.BGI.B.BBF REGISTER} usrloc [ucontact.c:72]: ucontact_xavp_store(): trying to clone per contact xavps Sep 8 13:59:04 sipserver /usr/sbin/kamailio[11830]: DEBUG: {1 81338 REGISTER 1955770655-18357-1(a)BJC.BGI.B.BBF REGISTER} usrloc [ucontact.c:1857]: uldb_insert_attrs(): trying to insert location attributes What am I doing wrong? It is supposed that the connection is by UDP. I also tried to insert the socket value as UDP, but it had the same problem. Thanks!

3 6

Topos does not add record-route on invite
by Angelo Sipper 09 Sep '21

09 Sep '21

Hello, We have the following setup on Kamailio 5.3.7 (x86_64/linux) and I am trying to use topos module on Kamailio2 in order to hide our topology but, mostly to reduce the invite size as most of the times we have fragmentation on invites from Kamailio1 as Kamailio1 it adds 2 record-routes and two via as it handles TLS from one side and UDP on the other side and we have found no wat to avoid all this info even with topos enabled on Kamailio1 there is no change on this. UA (OverTLS) -> Kamailio1 (Over UDP)-> Kamailio2 (Over UDP)-> Provider UA :50.50.50.50:5065 (TLS) Kamailio1: 170.170.170.210:5061 (TLS),170.170.170.210:5060 (UDP) Kamailio2: 170.170.170.170:5060 (UDP) Provider: 200.200.200.193:5060 (UDP) Kamailio1 does not run topos. Kamailio2 does run topos and rr with bellow settings: # ------ topos ------ modparam("topos", "db_url", DBURL) modparam("topos", "mask_callid", 0) modparam("topos", "sanity_checks", 0) modparam("topos", "branch_expire", 130) # 3 Min modparam("topos", "dialog_expire", 10800) # 3 Hours modparam("topos", "clean_interval", 60) modparam("topos", "event_mode", 3) modparam("topos", "contact_host", " sbc181.myserver.com") # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) modparam("rr", "ignore_sips", 1) # 0 = do not append from tag to the RR, 0 = append from tag to the RR modparam("rr", "append_fromtag", 0) # dual RR 0 = No, 1 = Yes when needed 2 = Always modparam("rr", "enable_double_rr", 0) The problem is that on the invite from Kamailio2 to provider, topos adds only VIA kai new contact. No record route at all. So the provider sends multiple 200 ok SDPs to wrong IPs. If I disable topos module on kamailio2 everything works fine except that we have huge invite as another VIA and another record route is added having now three of each and fragmentation is present. *Invite from Kamailio1 to Kamailio2* ------------------------------------------------- 2021/09/08 21:41:57.698823 170.170.170.210:5060 -> 170.170.170.181:5060 INVITE sip:1234567890@sip.myserver.com:5060 SIP/2.0 Record-Route: <sip:170.170.170.210;r2=on;lr;ftag=1b973153-4034-4480-89d3-c47b411bd81e;vsf=AAAAAFpUAgEBBwEAAXABAG5BR0EAQUBCZXJ2b2ljZS5ldQ--;vst=AAAAAAoMDgsBBQcGAgIAdENbQG4AFxNUXAUaGQYXWAocY2UuZXU-;did=ff2.96a;nat=yes> Record-Route: <sip:170.170.170.210:5061 ;transport=tls;r2=on;lr;ftag=1b973153-4034-4480-89d3-c47b411bd81e;vsf=AAAAAFpUAgEBBwEAAXABAG5BR0EAQUBCZXJ2b2ljZS5ldQ--;vst=AAAAAAoMDgsBBQcGAgIAdENbQG4AFxNUXAUaGQYXWAocY2UuZXU-;did=ff2.96a;nat=yes> Via: SIP/2.0/UDP 170.170.170.210;branch=z9hG4bKd703.e095cdba1da1a4b1599119c9135b7838.0;i=8eb41 Via: SIP/2.0/TLS 50.50.50.50:5065 ;received=50.50.50.50;rport=31090;branch=z9hG4bKPj9196e5f9-977c-4a53-a4c0-c0ab82b095da;alias From: 987654321 <sip: 987654321(a)sip.myserver.com >;tag=1b973153-4034-4480-89d3-c47b411bd81e To: <sip: 1234567890(a)sip.myserver.com> Contact: <sip:customer1@50.50.50.50:5065 ;transport=TLS;alias=50.50.50.50~31090~3> Call-ID: cd64414e-77d5-45b4-a52d-74ec7fb5c7ef CSeq: 15776 INVITE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 69 Content-Type: application/sdp Content-Length: 231 P-Asserted-Identity: <sip: 987654321(a)sip.myserver.com> v=0 o=- 391641952 391641952 IN IP4 170.170.170.216 s=Asterisk c=IN IP4 170.170.170.216 t=0 0 m=audio 29478 RTP/AVP 8 0 18 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv a=rtcp:29479 a=ptime:20 *Invite from Kamailio2 to Provider* ------------------------------------------------- 2021/09/08 21:41:57.715881 170.170.170.181:5060 -> 200.200.200.193:5060 INVITE sip: 1234567890@sbc181.myserver.com:5060 SIP/2.0 Via: SIP/2.0/UDP 170.170.170.181;branch=z9hG4bKd703.d45db1fd910c43c5e7c4f8711a5fca92.0 From: 987654321 <sip: 987654321(a)sip.myserver.com >;tag=1b973153-4034-4480-89d3-c47b411bd81e To: <sip:1234567890@sip.myserver.com> Call-ID: cd64414e-77d5-45b4-a52d-74ec7fb5c7ef CSeq: 15776 INVITE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 68 Content-Type: application/sdp Content-Length: 231 P-Asserted-Identity: <sip: 987654321(a)sip.myserver.com> Contact: <sip:btpsh-c9-613903ef-21d75-1@sbc181.myserver.com> v=0 o=- 391641952 391641952 IN IP4 170.170.170.184 s=Asterisk c=IN IP4 170.170.170.184 t=0 0 m=audio 11450 RTP/AVP 8 0 18 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv a=rtcp:11451 a=ptime:20 *Bellow you can see the flow* ---------------------------------------- 170.170.170.210:5060 170.170.170.170:5060 200.200.200.193:5060 qqqqqqqqqqwqqqqqqqqq qqqqqqqqqqwqqqqqqqqq qqqqqqqqqqwqqqqqqqqq 21:41:57.698823 x INVITE (SDP) x x +0.003788 x qqqqqqqqqqqqqqqqqqqqqqqqqq> x x 21:41:57.702611 x 100 trying -- your call is x x +0.013270 x <qqqqqqqqqqqqqqqqqqqqqqqqqq x x 21:41:57.715881 x x INVITE (SDP) x +0.018529 x x qqqqqqqqqqqqqqqqqqqqqqqqqq> x 21:41:57.734410 x x 100 Trying x +0.804083 x x <qqqqqqqqqqqqqqqqqqqqqqqqqq x 21:41:58.538493 x x 183 Session Progress (SDP) x +0.014557 x x <qqqqqqqqqqqqqqqqqqqqqqqqqq x 21:41:58.553050 x 183 Session Progress (SDP) x x +0.483907 x <qqqqqqqqqqqqqqqqqqqqqqqqqq x x 21:41:59.036957 x x 183 Session Progress (SDP) x +0.012009 x x <<<qqqqqqqqqqqqqqqqqqqqqqqq x 21:41:59.048966 x 183 Session Progress (SDP) x x +0.988036 x <<<qqqqqqqqqqqqqqqqqqqqqqqq x x 21:42:00.037002 x x 183 Session Progress (SDP) x +0.010960 x x <<<qqqqqqqqqqqqqqqqqqqqqqqq x 21:42:00.047962 x 183 Session Progress (SDP) x x +1.989085 x <<<qqqqqqqqqqqqqqqqqqqqqqqq x x 21:42:02.037047 x x 183 Session Progress (SDP) x +0.010947 x x <<<qqqqqqqqqqqqqqqqqqqqqqqq x 21:42:02.047994 x 183 Session Progress (SDP) x x +3.989064 x <<<qqqqqqqqqqqqqqqqqqqqqqqq x x 21:42:06.037058 x x 183 Session Progress (SDP) x +0.008860 x x <<<qqqqqqqqqqqqqqqqqqqqqqqq x 21:42:06.045918 x 183 Session Progress (SDP) x x +1.178240 x <<<qqqqqqqqqqqqqqqqqqqqqqqq x x 21:42:07.224158 x CANCEL x x +0.002725 x qqqqqqqqqqqqqqqqqqqqqqqqqq> x x 21:42:07.226883 x x CANCEL x +0.000372 x x qqqqqqqqqqqqqqqqqqqqqqqqqq> x 21:42:07.227255 x 200 canceling x x +0.016902 x <qqqqqqqqqqqqqqqqqqqqqqqqqq x x 21:42:07.244157 x x 200 OK x +0.004590 x x <qqqqqqqqqqqqqqqqqqqqqqqqqq x 21:42:07.248747 x x 487 Request Terminated x +0.003547 x x <qqqqqqqqqqqqqqqqqqqqqqqqqq x 21:42:07.252294 x x ACK x +0.003737 x x qqqqqqqqqqqqqqqqqqqqqqqqqq> x 21:42:07.256031 x 487 Request Terminated x x +0.001161 x <qqqqqqqqqqqqqqqqqqqqqqqqqq x x 21:42:07.257192 x ACK x x x qqqqqqqqqqqqqqqqqqqqqqqqqq> x TCP is out of question from the provider. Any help to use topos or any other module to reduce the size of our invites to avoid fragmentation will be much appreciated! Regards, Angelo

2 1

manually append_branch to lookup result
by David Villasmil 07 Sep '21

07 Sep '21

Hello guys, I'm trying to dynamically add a branch after doing lookup. The user is found, but in some cases I need to add a branch and do parallel forking. So i'm basically doing: route[LOCATION] { if (!lookup("location")) { .... } if (something) { route(BRANCH_TO_EXTRA); } route(RELAY); } route[BRANCH_TO_EXTRA] { $fs = MY_SOCKET; append_branch("sip:$tU@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port)); return; } For some reason only the branch appended is being used (I have append_branches=1) Ideas? David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 5

via header not removed wit hremove_hf()
by Richard Robson 07 Sep '21

07 Sep '21

during testing I need to remove a Via header This is the code snippet route[REPLACE_VIA] { xlog ("L_INFO", "[REPLACE_VIA] Start of block\n"); xlog ("L_INFO", "[REPLACE_VIA] $hdrc(Via) Via headers\n"); if(remove_hf("Via")) { xlog ("L_INFO", "[REPLACE_VIA] removed Via\n"); } else { xlog ("L_INFO", "[REPLACE_VIA] did not remove Via\n"); } xlog ("L_INFO", "[REPLACE_VIA] $hdrc(Via) Via headers\n"); and this is the logs output Sep 7 08:00:00 INFO: _LOG: 3a547f956d65fe985ba189916047fc320ae150ad: [REPLACE_RURI] Start of block Sep 7 08:00:00 INFO: _LOG: 3a547f956d65fe985ba189916047fc320ae150ad: [REPLACE_VIA] Start of block Sep 7 08:00:00 INFO: _LOG: 3a547f956d65fe985ba189916047fc320ae150ad: [REPLACE_VIA] 1 Via headers Sep 7 08:00:00 INFO: _LOG: 3a547f956d65fe985ba189916047fc320ae150ad: [REPLACE_VIA] removed Via Sep 7 08:00:00 INFO: _LOG: 3a547f956d65fe985ba189916047fc320ae150ad: [REPLACE_VIA] 1 Via headers From the logs it looks like the remove_hf returns true and we have the header removed, but its still counting the Via header and the Via Header is still present when the INVIE is releyed on Regards, Richard

2 4

Kamailio and robotics repo
by Yuriy Gorlichenko 07 Sep '21

07 Sep '21

Following the promise during the Kamailio world 2021 I share the source code of the Kamailoi and robotics repo: https://github.com/arigativa/robolive

2 1

Manage multiple RTP streams with different TO-tags (forking)
by B. Tietz 07 Sep '21

07 Sep '21

Hello, following situation. I have a Kamailio (5.4) using rtpengine to loadbalance calls. If a call from Alice comes in, Kamailio decides to send the call to Carrier B from Bob. Bobs Phone is ringing and the carrier B send a 183 Session Progress with SDP and To-tag=abcd. The SDP has G722 as codec and port 1234. A few moments later carrier B send a second 183 Session Progress with SDP and TO-tag=fghi. The SDP has G711 as codec and port 5678. This is done, to play some funky music as ringtone -.- If Bob answers the call, carrier B sends a 200 OK WITHOUT SDP and TO-tag=abcd. So this should instruct our Kamailio to switch to the first G722 and port 1234. But sadly, this is just not working as expected. We tried to set the flags media-handover and port-latching for the rtpengine options and additionally set a to-tag when using rtpenging_manage. But this doesn't solve the codec change, so we have only audio when Bob answers the call, but no ringtone-music. If we allow G711 only in the outgoing INVITE to Bob, we have also tha ringtone-muisic, because there is no codec-change. Carrier B tells us, they are using a fork-mechanism. Is there something we can do, to support the codec change in 183? Or enforce carrier B to send SDP in 200 OK? Or anything else? Carrier B can not change anything in the ringtone-music-backend. They are stuck on G711. Thanks!

5 4

Re: [SR-Users] t_drop_replies not working with t_suspend in failure route
by Huber Andreas 03 Sep '21

03 Sep '21

Hi, We’re still using kamailio 4.4 but we’ll be migrating to 5.0 soon. Cool so it will be fixed when we migrate ! Thanks, Andreas From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Federico Cabiddu Sent: vendredi 12 mai 2017 11:56 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] t_drop_replies not working with t_suspend in failure route Hi, which version are you using? A similar case had been reported some months ago and it should be fixed in 5.0. Regards, Federico On Fri, May 12, 2017 at 11:44 AM, Huber Andreas <andreas.huber(a)nagra.com<mailto:andreas.huber@nagra.com>> wrote: Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We’d like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 1

How to drop negative final response and send ACK
by Koray Vatansever 02 Sep '21

02 Sep '21

Hi, I have a problem with the following scenario: I have web and mobile sip clients for the same username. If both clients are idle, INVITE is forked and both clients ring. If web client is busy and mobile client is online and registered, the mobile client rings and the caller hears ringback tone. If web client is busy and mobile client is offline, before the mobile client receives push notification and registers, web client sends 486 busy response and caller hears busy tone. In this last scenario, I want to wait some time for the mobile client to register and ring. I tried to drop 486 response in onreply_route[x], but I found that final responses cannot be dropped in this route. I dropped 486 in reply_route, but I couldn't send an ACK for this transaction. Is there a way to implement this scenario? I would appreciate if you have any suggestions. Thanks, Koray

2 2

dns_srv_lb on or yes
by Juha Heinanen 01 Sep '21

01 Sep '21

Wiki tells: dns_srv_lb = yes | no (default no) and doc/tutorials/dns.txt tells: dns_srv_lb = on | off Which one is correct or does it matter? -- Juha

2 3

Info: Tomorrow Live - Kamailio World 2021 - Online
by Daniel-Constantin Mierla 31 Aug '21

31 Aug '21

Hello, couple of details for those that want to watch live or even participate to Kamailio World Online Conference, Sep 1-2, 2021 - the schedule is available at: * https://www.kamailioworld.com/k09-online/ There is no registration required, participation is completely free. You can watch live the presentations starting with 13:00UTC (14:00 London, 15:00 Berlin/Paris/Rome/Madrid, 09:00 New York) via KamailioWorld youtube channel: * https://www.youtube.com/c/KamailioWorld If you want to participate to the live video conferencing room, the link is: * https://meet.kamailioworld.com/live It is recommended to do it only if you want to address questions and interact with the speakers, otherwise the capacity of the video conferencing system may be filled and we will have to restrict the access in order to ensure good quality for presentations. You can also address questions via youtube chat that will be available next to the live stream or Matrix room: * #kamailio:matrix.kamailio.dev * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev To interact with other community member, join the Matrix chat room. No registration is required for Matrix to join the room, however the web client at the link above does not support the guests feature, you will have to use another client application that supports it or make an account with any federated Matrix server, like the one from element.io: * https://app.element.io/#/welcome Looking forward to chatting with many of you during the next two days! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Add PIDF-LO object to INVITE
by Adarsh Chauhan 31 Aug '21

31 Aug '21

Hi All, I am trying to add PIDF-LO XML to INVITE, however i have had little success so far. My end goal is to create an invite like mentioned in https://docs.microsoft.com/en-us/openspecs/office_protocols/ms-sipre/4010da… I have the following config: request_route{ xlog("L_DBG","REQUEST RECEIVED \n ($mb) \n"); if (is_method("INVITE")) { xinfo("$ua\n"); regex_substring("$ua", "([a-f0-9]{12})",0, 1, "$var(asd)"); if($var(asd)!="") { http_client_query("http://127.0.0.1/location/$var(asd)","$var(result)"); xdbg("$rc ,[$var(result)]\n"); if(!has_body("application/pidf+xml")) { set_body_multipart("$rb", "application/sdp", "delimiter"); msg_apply_changes(); append_body_part($(var(result){s.unquote}), "application/pidf+xml"); msg_apply_changes(); } }else { xerr("NO MAC ADDRESS FOUND\n"); } } route("forward"); } which achieves the following but doesn't have the xml in the correct format. I have explored most of the modules but couldnt find a handy function for this. INVITE sip:service@10.0.1.133:5060 SIP/2.0 Via: SIP/2.0/TCP 169.254.172.2;branch=z9hG4bK3de6.d3b4cddbcbdba59a3e190c41b06ada15.0;i=1 From: sipp <sip:sipp@1.1.1.1:45907>;tag=21113SIPpTag001 To: service <sip:service@10.0.1.133:5060> Call-ID: 1-21113(a)1.1.1.1 CSeq: 1 INVITE Contact: sip:sipp@1.1.1.1:45907 Max-Forwards: 70 Subject: Performance Test Content-Length: 937 Content-Type: multipart/mixed;boundary="delimiter" Mime-Version: 1.0 --delimiter Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 172.31.87.143 s=- c=IN IP4 1.1.1.1 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --delimiter Content-Type: application/pidf+xml <?xml version=\"1.0\" encoding=\"UTF-8\"?>\n<presence xmlns:xsd=\"http:/ /www.w3.org/2001/XMLSchema\" xmlns:xsi=\"http://www.w3.org/2001/ XMLSchema-instance\" entity=\"sip:U SER9(a)SBSBVR00Test0m.ABC.com\" xmlns=\"urn:ietf:params:xml:ns:pidf\">\n<tuple id=\"tuple0\">\n<status>\n<geopriv xmlns=\"urn:ietf:params:xml:ns:pidf:geopriv10\">\n<civicAddre ss xmlns=\"urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr\">\n['<STS>Ave</STS>\\n', '<A1>WA</A1>\\n', '<A3>Redmond</A3>\\n', '<RD>163rd</RD>\\n', '<PC>98052</PC>\\n', '<POD >NE</POD>\\n', '<NAM>Contoso Corporation </NAM>\\n', '<HNO>3910</HNO>\\n', '<country>US</country>\\n']\n</civicAddress>\n</geopriv>\n</status>\n</tuple>\n</presence> --delimiter-- Regards, Adarsh Chauhan

1 0

What does "tls.reload" actually do?
by Sebastian Damm 30 Aug '21

30 Aug '21

Hi, I noticed a strange behavior on some of our proxy servers, all running Kamailio 5.3.8. After running for some time (weeks), our monitoring system sporadically starts reporting errors. The check connects via tls and registers to an Asterisk behind the proxy server. When this happens, the Kamailio log shows the following line: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1409441B:SSL routines:ssl3_read_bytes:tlsv1 alert decrypt error When restarting Kamailio, the problem goes away only to come back after some weeks uptime again. On one host, I tried to find something using kamcmd, and I don't know why but I also issued "tls.reload". And from that point, the monitoring system has not reported the system as faulty anymore. I repeated the same thing on other hosts when the problem occured there, all with the same result. "tls.reload" helps. But from the documentation, I don't know why. Does anybody have an explanation for it? Regards, Sebastian

4 8

Slack Group
by Daniel Donoghue 26 Aug '21

26 Aug '21

Hello everyone Does anybody know if the Slack workspace at kamailio.slack.com (as mentioned on https://www.kamailio.org/w/contact-us/) is still active? If it is, who should we contact for access? With every blessing, — Daniel Donoghue

2 1

pua dialogindo doesn't not generate PUBLISH when routes is written on kemi framework(python)
by Zhan Bazarov 26 Aug '21

26 Aug '21

Hi everyone. I have an issue, when I loaded the kemi python engine - module pua_dialoginfo stopped sending publishes... Has anyone faced the same issue? kamailio version is 5.4.5

1 0

no transaction found so that cant step into route-block
by fran.interpla.56906 26 Aug '21

26 Aug '21

1 0

no transaction found so that cant step into route-block
by fran.interpla.56906 26 Aug '21

26 Aug '21

1 0

Kamailio v5.5.2 Released
by Daniel-Constantin Mierla 25 Aug '21

25 Aug '21

Hello, Kamailio SIP Server v5.5.2 stable release is out. This is a maintenance release of the latest stable branch, 5.5, that includes fixes since the release of v5.5.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.2. For more details about version 5.5.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/08/kamailio-v5-5-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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

3 2

MS Teams outbound calls. Called number normalization
by Володимир Іванець 25 Aug '21

25 Aug '21

Hello all! I'm testing integration with MS Teams. I treat calls from Teams to the phone system the same way as calls from phones. And would like to ask if someone has already dealt with this. My test account location is set to the United States. When I make a call to "200", an INVITE request from Teams to Kamailio will have "+1200" in the To URI user. I could just strip "+1" but when I make a call to "100", Teams sends "+100". So in this case I need to strip only one character. The problem is that I don't know when I need to remove one or two characters. And it gets more complicated when I change my test account location. For some countries, Teams prepend 3 characters. What do to make sure phone numbers are consistent? I thought this behavior could be disabled in Teams but could not find anything related yet. Thank you! Regards, Volodymyr Ivanets

1 0

Releasing Kamailio v5.5.2
by Daniel-Constantin Mierla 25 Aug '21

25 Aug '21

Hello, I am considering to release Kamailio v5.5.2 sometime next week, likely on Wednesday (August 25, 2021). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

[VoLTE] 401 unauthorized error
by 오택경 24 Aug '21

24 Aug '21

Hi. I am implementing the VoLTE setup with the dockerized project (https://github.com/herlesupreeth/docker_open5gs https://github.com/herlesupreeth/docker_open5gs). I have almost done to run the VoLTE service, but 401 unauthorized error in sip and auth-pending error in fhoss have occured. How can I fix this problem? I will share the discussion note in which I tried to solve some problems including the above one. : https://github.com/herlesupreeth/docker_open5gs/issues/55 https://github.com/herlesupreeth/docker_open5gs/issues/55 Very thanks, Taekkyung Oh.

3 7

Race conditions upon quick reinvite
by David VILLAUME 24 Aug '21

24 Aug '21

Hello, We encounter sometimes reinvite from some provider really quickly after 200 OK ACK so sometime there a race conditions where the re-invite exits Kamailio before the ACK leading to some Freeswitch “500 Overlapping Requests”. [cid:image001.png@01D798FA.6F5B9D80] I tried to catch the 500 on failure-route unsuccessfully as transaction is already established. Do you have any idea on how I could deal with this behavour ? Running 5.3.6 with 8 childs modules used :(tm + dialog) No costly action performed on ack that can explain slowliness. Regards, David

3 3

kazoo error
by 崔群 24 Aug '21

24 Aug '21

Dear sir: I have some problems。the kamailio version is 5.3.0. My linux version is centos 7. the kamailio log: ERROR: kazoo [kz_amqp.c:2238]: kz_amqp_send_ex(): Failed to publish -9 : a socket error occurred ERROR: kazoo [kz_amqp.c:2707]: kz_amqp_publisher_send(): error sending publish to zone : local , connection id : 1, uri : amqp://admin The connection between kamailio and rabbitmq was interrupted, which caused the queue of MQ to be blocked and the message could not be consumed。

2 1

Some questions about drouting module
by Андрей Яри н 23 Aug '21

23 Aug '21

Hello! I test module on Centos 7, kamailio version 5.5 and found some strange things: 1) Definition of table dr_gateways at https://kamailio.org/docs/modules/5.5.x/modules/drouting.html#idm98 - in DB there is field 'attrs' and only notice about it is module param 3.8. attrs_avp but it doesnt describe table field. In OpenSIPS docs i found description * attributes (not used by DR engine, but only pushed to script level when routing to this GW) In kamailio this field is deprecated or its only docs misses param? 2) Previous moment was found when i try to resolve: WARNING: drouting [dr_load.c:529]: dr_load_routing_info(): route <1> does not exist "Definition of dr_rules table" says that field has type of integer, but realy in database its varchar and i think its old docs. But i use KEMI question is - how i can tell module that it must be called in KEMI, not in native? 3) Every time when i call 'kamcmd drouting.reload' i get message CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: bad pointer 0x7f3e4cea04c0 (out of memory block!) called from core: core/utils/tmrec.c: tmrec_free(393) - ignoring I rollback to 5.4 and still get this error. As i understand its old module and its can be internal error - how critical this error can be? 4) Also planned commands like drouting.dump gws|gw_lists|rules? -- С уважением, Ярин Андрей Анатольевич инженер, группы управления телефонией Группа компаний «Интерсвязь» e-mail: yarin-aa(a)intersvyaz.net

2 3

Kamailio as basic dispatcher for load balancing
by Philipp Trenz 23 Aug '21

23 Aug '21

Hey there, I’m Philipp, an IT Systems Engineer from Potsdam, Germany and want to deepen my knowledge in SIP communication on scale. Currently, my setup is quite simple: A single stateless Asterisk instance, fully managed via ARI, which registers to multiple VoIP providers, only processes incoming calls to play audio files (no outgoing or conference calls). Now, with growing load on the single Asterisk instance, I would like to have one or two Kamailio instances, which will balance the load of incoming calls to multiple Asterisk instances. On which Asterisk instance they end up is irrelevant for me, as the calls can be processed on every instance. I looked into some Kamailio and Astricon conference speeches and checked out some tutorials. But still I’m not quite sure where to start. So here are some basic questions which would really help me out to get going: As I don’t want to have registrations of all my Asterisk instances at the VoIP providers, I should use the UAC module to let Kamailio do the registering to the VoIP providers and register my Asterisk instances to Kamailio, right? Will Kamailio automatically apply new outbound registrations when added to the database or do I have to trigger that manually? Should I use one or two instances of Kamailio, how hard is it to configure them fail-safe? For writing my first config file, should I start blank or from the standard config, what’s best practice? Thanks in advance, looking forward to your replies! Philipp

3 5

How tell if rtp packets are being relayed through rtpengine in kamailio
by Faiz Fazaluddin 23 Aug '21

23 Aug '21

Hi, I want to see if rtp packets are being relayed through kamailio. I am attmepting to connect rtpengine with kamailio, but am not sure if it is working, how can I tell? I have the kamailio server on a different VM than the clients. In wireshark, no rtp packets show on the server VM and rtp packets only seem to be going directly through the client VM. Rtp engine mentions being unable to find hashtables when attempting to implement a delete_node command. Thank you very much for your help, Faiz

3 2

Register Forwarding
by Thomas Grande 21 Aug '21

21 Aug '21

Hi there! I bumped into this post to perform forwarding of REGISTER requests and then saving a local cache on 2xx replies from main Registrar: https://lists.kamailio.org/pipermail/sr-users/2020-October/110779.html I think I understand all the steps described, but some features I need are missing: - How to change the Via headers to perform topology hiding? I understand TOPOS and TOPOH do not work on these types of messages. - Change contact header so that registrar responses traverse the Kamailio box. (Use textops I suppose?) - Besides, is this approach still the best in comparison to OpenSIPs mid-registrar module? Thanks in advance, -- *Thomás Alimena Del Grande* Engenharia - Aligera Tel. 51 3500-0121

3 2

record_route() from websocket UA to tls UA
by Juha Heinanen 20 Aug '21

20 Aug '21

I noticed that when jsSIP UA that has registered over wss calls another SIP UA that has registered over tls, record_route() adds only one Route URI to outgoing INVITE (example below). This causes BYE to fail. This issue may be caused by the fact that both UAs register over the same Kamailio tls listening socket. Still two Route URIs should be added in the same way as is done when one UA registers over tcp and the other over udp. -- Juha --------------------------------------------------------------------- 09:25:43.639833 wss:87.95.73.155:19458 wss:192.27.134.1:5061 INVITE sip:foo@test.tutpro.com SIP/2.0 Via: SIP/2.0/WSS jovobf9n4svm.invalid;branch=z9hG4bK2639638 Max-Forwards: 69 To: <sip:foo@test.tutpro.com> From: " Test" <sip:test@test.tutpro.com>;tag=f7lpsuo7cb Call-ID: ovhd9fu1fl1a66nec011 CSeq: 1387 INVITE Contact: <sip:test@test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5> ... 09:25:43.649462 tls:192.27.134.1:5061 tls:87.95.73.155:19461 INVITE sip:foo-0x793ee87a90@10.158.141.103:38378;transport=tls SIP/2.0 Record-Route: <sip:192.27.134.1:5061;transport=ws;sn=ext_tls;lr;n2;dtlsf=avp;pm=0;ice> Via: SIP/2.0/TLS 192.27.134.1:5061;branch=z9hG4bK637c.7e15ef5b84ea054cfc8ea3d4e860521f.0 Via: SIP/2.0/WSS jovobf9n4svm.invalid;rport=19458;received=87.95.73.155;branch=z9hG4bK2639638 Max-Forwards: 68 To: <sip:foo@test.tutpro.com> From: " Test" <sip:test@test.tutpro.com>;tag=f7lpsuo7cb Call-ID: ovhd9fu1fl1a66nec011 CSeq: 1387 INVITE Contact: <sip:test@test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5>

2 11

install latest sems-server on debian 10 buster
by Mojtaba 18 Aug '21

18 Aug '21

Hello there, Does anybody have experience installing the latest SEMS-Server (1.6.0) on debian 10 buster? I tried to install both 1.6.0. and 1.7-dev and some issues are occurred, but i installed the version 1.3.1 on debian 8 before. Any help would be appreciated. Thanks with best regards -- --Mojtaba Esfandiari.S

4 11

How to insert user=phone header INVITE TO FROM
by César Matheus 18 Aug '21

18 Aug '21

Hi list. mY provider asks me to insert user = phone in my header. because it requires to receive my INVITE with user = phone in TO and FROM However, I add the user = phone in my configuration like this: $ ru = "sip:" + $ rU + "@" + $ sel (cfg_get.pstn1.gw_ip) + ":" + $ sel (cfg_get.pstn1.gw_port) + "; user = phone"; but only the INVITE appears, not in the TO or FROM. INVITE sip: 09872323232(a)X.Y.X.K: 5060; user = phone SIP / 2.0 From: <sip: XXXXYYYYY(a)172.20.12.4: 5060>; tag = 15552444533876 To: <sip: 09872323232(a)172.20.12.2: 5060> You can help me solve it. Thanks. -- César Matheus

3 2

crash in timer
by Juha Heinanen 17 Aug '21

17 Aug '21

Any idea what could cause the crash shown in the bt below? This is with Kamailio 5.5. -- Juha [New LWP 12061] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". bt Core was generated by `/usr/bin/sip-proxy -f /etc/sip-proxy/sip-proxy.cfg -P /run/sip-proxy/sip-proxy.'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x0000563d3b390c5d in timer_list_expire (t=1584008014, h=0x7f7a1e76bb40, slow_l=0x7f7a1e76d680, slow_mark=63909) at core/timer.c:846 846 core/timer.c: No such file or directory. (gdb) bt full #0 0x0000563d3b390c5d in timer_list_expire (t=1584008014, h=0x7f7a1e76bb40, slow_l=0x7f7a1e76d680, slow_mark=63909) at core/timer.c:846 tl = 0x7f7a226ecbd8 ret = 0 #1 0x0000563d3b391219 in timer_handler () at core/timer.c:922 saved_ticks = 1584008014 run_slow_timer = 0 i = 421 __func__ = "timer_handler" #2 0x0000563d3b391742 in timer_main () at core/timer.c:961 No locals. #3 0x0000563d3b2c94de in main_loop () at main.c:1839 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=46.182.160.20:5060\000\000\000\000\003\000\000\000 \000\000\000\000\071CU\356xL\375\000\000\000\000\000\000\000\000\210\300m;=V\000\000\000\000\000\000\000\000\000\000\260\334\371Jz\177\000\000\263\331r;=V\000\000\000\000\000 \000\000\000\000\340\b3;\374\177\000\000\t\365U;=V\000" nrprocs = 8 woneinit = 1 __func__ = "main_loop" #4 0x0000563d3b2d44aa in main (argc=17, argv=0x7ffc3b330e28) at main.c:3053 cfg_stream = 0x563d3b9ea290 c = -1 r = 0 tmp = 0x7ffc3b331f3f "" tmp_len = 1260628273 port = 32634 proto = 0 ahost = 0x0 aport = 0 --Type <RET> for more, q to quit, c to continue without paging--c options = 0x563d3b6dece0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 1970794883 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x0 p = 0xffff00001f80 <error: Cannot access memory at address 0xffff00001f80> st = {st_dev = 21, st_ino = 21649, st_nlink = 2, st_mode = 16832, st_uid = 109, st_gid = 115, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1620846563, tv_nsec = 71986646}, st_mtim = {tv_sec = 1628122861, tv_nsec = 116913769}, st_ctim = {tv_sec = 1628122861, tv_nsec = 616909103}, __glibc_reserved = {0, 0, 0}} tbuf = "\000\000\000\000\000\000\000\000\b\000\000\000\000\000\000\000\060\067&Kz\177\000\000\000\000\000\000\000\000\000\000\240\241>;\374\177\000\000\364:$Kz\177\000\000\b\000\000\000\000\000\000\000\230\242>;\374\177\000\000\060\067&Kz\177\000\000\310\v3;\374\177\000\000\304\v3;\374\177", '\000' <repeats 18 times>, "\230\242>;\374\177\000\000\250\241>;\374\177\000\000'\256\002Kz\177\000\000&\260be\000\000\000\000\060\r3;\374\177\000\000\300\212\225\001\000\000\000\000\220\f3;\374\177\000\000\200\f3;\374\177\000\000\310\v3;\374\177\000\000X7&Kz\177\000\000\001", '\000' <repeats 31 times>... option_index = 0 long_options = {{name = 0x563d3b6e0f3f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x563d3b6dc4e1 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x563d3b6e0f44 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x563d3b6e0f4a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x563d3b6e0f50 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x563d3b6e0f59 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x563d3b6e0f63 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x563d3b6e0f6d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x563d3b6e0f78 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x563d3b6e0f81 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x563d3b6e0f8c "debug", has_arg = 1, flag = 0x0, val = 1032}, {name = 0x563d3b6e0f92 "cfg-print", has_arg = 0, flag = 0x0, val = 1033}, {name = 0x563d3b6e0f9c "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __func__ = "main"

1 0

Info: First group of presentations at Kamailio World Online 2021
by Daniel-Constantin Mierla 17 Aug '21

17 Aug '21

Hello, the first group of presentations selected for the next Kamailio World Online, September 1-2, 2021, has been published, aiming to provide a good knowledge base of using Kamailio for different deployment types: segregation of networks with an SBC-like role, controlling robots, RTP streams management with RTPEngine, configuration variables or transformations, SIP attacks handling and asynchronous SIP routing. You can find more details at: * https://kamailioworld.com/k09-online/ The schedule will be completed during the next days, with another group of interesting sessions and open discussion panels. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

No docker hub automated builds
by Sergey Safarov 14 Aug '21

14 Aug '21

I found that not more automated builds available on docker hub https://www.docker.com/blog/changes-to-docker-hub-autobuilds/ Looks like we need to make some changes for the alpine docker image. Maybe use github actions for this task? https://docs.docker.com/ci-cd/github-actions/ Sergey

1 0

RTPEngine: Async connection/communication between Kamailio and RTPEngine
by Mitesh Thakkar 12 Aug '21

12 Aug '21

Hi, Couple of week before I have posted feature request - https://github.com/kamailio/kamailio/issues/2807 As suggested, I am trying to figure out how I can achieve it with async or sworker modules. Can someone help me to understand how to use those modules to achieve async connectivity? Regards, Miteshkumar Thakkar

3 3

auth challenge freeswitch with Zopier
by Joli Martinez 10 Aug '21

10 Aug '21

Hello, I am using dsiprouter as a public facing server for pass through registrations to a fusion pbx box. This setup works great and has for a long time now. The issue I am having is that we are switching soft phone clients to copier. Zopier uses a push server. When an incoming call comes into the freeswitch server, it fires off an invite to the Zopier registered client. For some reason ONLY in this scenario Kamailio is doing an auth challenge to Freeswitch preventing to call to be sent to the soft phone. How can I stop this from happening? Other soft phones and hard phones work great with inbound calls except for Zopier. thanks,

2 1

Integration with multiple MS Teams instances
by Володимир Іванець 10 Aug '21

10 Aug '21

Hello all! I was able to connect Kamailio with MS Teams and now trying to add one more Teams instance. It looks like I have some misconfiguration or there is a bug. My test server has 2 domain records pointing at it (kamailio.domain1.com and kamailio.domain2.com). My tls.cfg configuration file looks like this. As you can see the Default section is configured with a kamailio.domain1.com sertificate: *[server:default]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain1.com/server/key.pem <http://kamailio.domain1.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain1.com/server/cert.pem <http://kamailio.domain1.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain1.com/CA/cert.pem <http://kamailio.domain1.com/CA/cert.pem>* *[client:default]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain1.com/server/key.pem <http://kamailio.domain1.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain1.com/server/cert.pem <http://kamailio.domain1.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain1.com/CA/cert.pem <http://kamailio.domain1.com/CA/cert.pem>* *[server:172.16.30.206:5062 <http://172.16.30.206:5062>]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain1.com/server/key.pem <http://kamailio.domain1.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain1.com/server/cert.pem <http://kamailio.domain1.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain1.com/CA/cert.pem <http://kamailio.domain1.com/CA/cert.pem>* *server_name = "kamailio.domain1.com <http://kamailio.domain1.com>"* *server_id = "**"kamailio.domain1.com <http://kamailio.domain1.com>"* *[client:172.16.30.206:5062 <http://172.16.30.206:5062>]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain1.com/server/key.pem <http://kamailio.domain1.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain1.com/server/cert.pem <http://kamailio.domain1.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain1.com/CA/cert.pem <http://kamailio.domain1.com/CA/cert.pem>* *[server:172.16.30.206:5063 <http://172.16.30.206:5063>]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain2.com/server/key.pem <http://kamailio.domain2.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain2.com/server/cert.pem <http://kamailio.domain2.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain2.com/CA/cert.pem <http://kamailio.domain2.com/CA/cert.pem>* *server_name = "kamailio.domain2.com <http://kamailio.domain2.com>"* *server_id = "**"kamailio.domain2.com <http://kamailio.domain2.com>"* *[client:172.16.30.206:5063 <http://172.16.30.206:5063>]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain2.com/server/key.pem <http://kamailio.domain2.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain2.com/server/cert.pem <http://kamailio.domain2.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain2.com/CA/cert.pem <http://kamailio.domain2.com/CA/cert.pem>* The dispatcher configuration table looks like this: +----+-------+----------------------------------------------+-------+----------+--------------------------------------------------------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+----------------------------------------------+-------+----------+--------------------------------------------------------------------+-------------+ | 1 | 1 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 3 | socket=tls:172.16.30.206:5062;ping_from=sip:kamailio.domain1.com | MS Teams 1 | | 2 | 2 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 3 | socket=tls:172.16.30.206:5063;ping_from=sip:kamailio.domain2.com | MS Teams 2 | +----+-------+----------------------------------------------+-------+----------+--------------------------------------------------------------------+-------------+ When Kamailio is started only connection with the first trunk is established: *# kamcmd tls.list* *{* * id: 1* * timeout: 0* * src_ip: 52.114.75.24* * src_port: 5061* * dst_ip: 172.16.30.206* * dst_port: 0* * cipher: ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH Au=RSA Enc=AESGCM(256) Mac=AEAD* * ct_wq_size: 0* * enc_rd_buf: 0* * flags: 2* * state: established* *}* *{* * id: 2* * timeout: 0* * src_ip: 52.114.75.24* * src_port: 7810* * dst_ip: 172.16.30.206* * dst_port: 5062* * cipher: AES256-GCM-SHA384 TLSv1.2 Kx=RSA Au=RSA Enc=AESGCM(256) Mac=AEAD* * ct_wq_size: 0* * enc_rd_buf: 0* * flags: 2* * state: established* *}* *{* * id: 3* * timeout: 596* * src_ip: 52.114.75.24* * src_port: 7811* * dst_ip: 172.16.30.206* * dst_port: 5062* * cipher: AES256-GCM-SHA384 TLSv1.2 Kx=RSA Au=RSA Enc=AESGCM(256) Mac=AEAD* * ct_wq_size: 0* * enc_rd_buf: 0* * flags: 2* * state: established* *}* Here is what I can see in Kamailio log file when it sends an OPTIONS request to the second trunk. Kamailio uses Default tls configuration and MS Teams don't accept it: *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: ALERT: <script>: == TRACE. tm:local-request. fs is tls:172.16.30.206:5063 <http://172.16.30.206:5063>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tm [uac.c:352]: t_run_local_req(): apply new updates without Via to sip msg* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/msg_translator.c:1796]: check_boundaries(): no multi-part body* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): SIP Request:* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:612]: parse_msg(): method: <OPTIONS>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): uri: <sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK169b.6411b4c3000000000000000000000000.0>; state=16* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [47]; uri=[sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (47)[<sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls>^M* *], to tag (0)[]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <10> <OPTIONS>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=0* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): SIP Request:* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:612]: parse_msg(): method: <OPTIONS>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): uri: <sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK169b.6411b4c3000000000000000000000000.0>; state=16* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [47]; uri=[sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (47)[<sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls>^M* *], to tag (0)[]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <10> <OPTIONS>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tm [uac.c:189]: uac_refresh_hdr_shortcuts(): cseq: [CSeq: 10]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/tcp_main.c:1993]: tcp_send(): no open tcp connection found, opening new one* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 52.114.75.24* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/tcp_main.c:1175]: tcpconn_new(): on port 5061, type 3, socket -1* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/tcp_main.c:1498]: tcpconn_add(): hashes: 2831:67:0, 1* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_server.c:199]: tls_complete_init(): completing tls connection initialization* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_server.c:162]: tls_get_connect_server_name(): xavp with outbound server name not found* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_server.c:142]: tls_get_connect_server_id(): xavp with outbound server id not found* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_server.c:228]: tls_complete_init(): Using initial TLS domain TLSc<default> (dom 0x7f35509da688 ctx 0x7f3550b7a568 sn [])* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_domain.c:1177]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7f3550b7a568: (nil)* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_domain.c:747]: sr_ssl_ctx_info_callback(): SSL handshake started* *...* If I change the Default configuration to use kamailio.domain2.com certificate, the second trunk will connect but the first one will fail. I tried to set "$xavp(tls=>server_name)" and "$xavp(tls[0]=>server_id)" variables to the event_route[tm:local-request] section but log still stated that server Name and ID were not found. Can someone please point me in the right direction, how can I make Kamailio use the correct certificates when establishing multiple TLS connections? Thanks a lot! Regards, Volodymyr Ivanets

3 12

Ringing Timeout
by Edward Romanenco 09 Aug '21

09 Aug '21

Hey all, I'm trying to find the best way for settings a timeout for the 'ringing' stage of a call - meaning - I would like to wait for 20 seconds between receiving status 180 / 183 / early media from the remote end and 200, if I fail to receive the expected response in this timeframe - call should be dropped. Most of the documented timeouts I saw count between the first invite to any of the aforementioned stages - but I'm actually looking for the opposite. Edward

2 1

help
by Said Hassani 08 Aug '21

08 Aug '21

Dear Team, I would like to know how to link Kamailio with openIMSCore?I would like to use Kamailio as an Application server.Please guide me or if there is someone who has worked on this subject, I would like to benefit from his help.

3 4

XCAP server with simservs support
by H Yavari 05 Aug '21

05 Aug '21

Hi to all, As I found there isn't any support for this AUID : simservs.ngn.etsi.org in Kamailio.Anyone can confirm that we can't use Kamailio as mmtel for IMS with the current xcap_server module?!!Thank you. Regards,Hossein

2 2

no debug trace of incoming request
by Juha Heinanen 05 Aug '21

05 Aug '21

Is it somehow possible that ngrep shows incoming INVITE arriving over TCP to Kamailio's listening address and port, but there is no debug trace of the request (e.g. receive_msg(): --- received sip message ...) in syslog? -- Juha

3 3

Issues with TOPOS when not relaying an in-dialog request
by Marrold 05 Aug '21

05 Aug '21

Hi, I am currently using TOPOS to make Kamailio behave more like a B2BUA from the clients perspective. It's working well, however I have discovered a scenario where it fails. To aid with some interoperability I am sending (sl_send_reply) 200 OK to an in-dialog SUBSCRIBE request during a call. Once this has happened, the subsequent BYE from the B leg goes to the incorrect (Private) IP. If I disable this SUBSCRIBE, or relay it to the B leg, the BYE goes to the expected IP address. Does anyone know where I am going wrong? I have tried not calling record_route() for these messages but the result is the same. I've attached the ladder diagram of the call below. Note the subscribe is not relayed in this scenario. Thanks! [image: image.png]

4 8

Re: [SR-Users] How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 04 Aug '21

04 Aug '21

> Serdar, > > Have you tried to clean variables before calling new ds_select_domain(), > that are using by dispatcher module failover? > > Like > https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.… <https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…> > and so on? > > But as I got, you're saying, that calling ds_select_domain() with > different setid's in a case of fail, not really fails, but using "old" > available destinations from previous attempt? > > Regards, > Igor Igor, also thanks for your interest. > Hello, > > delete the xavps based on the names you set via modparams xavp_dst and > xavp_ctx. > > Cheers, > Daniel I remove xavp value using "xavp_rm("_dsdst_")", i supposed that my problem was solved but in my other tests, i recognised that xavp_rm removed the first index of "xavp(_dsdst_)" not all previous destinations(indexes). after researching, i found a post at https://lists.kamailio.org/pipermail/sr-users/2020-May/109192.html and i removed all ellements of list as below, if(defined $xavp(_dsdst_)) { while($xavp(_dsdst_[0]) != $null) { xlog("L_INFO", "--- Loaded Dispatchers --- Grp : $xavp(_dsdst_[0]=>grp)\n"); xlog("L_INFO", "--- Loaded Dispatchers --- Uri : $xavp(_dsdst_[0]=>uri)\n"); $xavp(_dsdst_[0]) = $null; } } it worked but i am not sure it is a best solution. Is there another simple way to delete all indexes of xavp? Best regards, Serdar

1 0

Converting Cisco SIP NOTIFY DTMF to SIP INFO (Binary operations)
by Marrold 04 Aug '21

04 Aug '21

Hi, I am using an old Cisco 2811 with an FXS card and trying to get it to convert between POTS and SIP. Mostly for "fun". For the most part it works, however there are some problems with DTMF in the call. Because the end device uses pulse dialling it seems that only the Cisco Proprietary SIP NOTIFY method of sending DTMF is supported. It sends 4 bytes in the message body that indicate the DTMF digit sent. I'd like to convert it to the more standard SIP INFO method. The docs about the encoding are here - https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/1… I looked at the transformations but couldn't find anything relevant. Is it possible to extract the binary from the body and decode it? Thanks Matthew

1 0

which is better for B2BUA
by Mojtaba 04 Aug '21

04 Aug '21

Hello there, According to , https://lists.kamailio.org/pipermail/sr-users/2016-March/092058.html, which talked about B2BUA (just signalling) in Kamailio. As i have experienced working with SEMS, freeswitch and Kamailio while using B2BUA feature, Each of them have pros and cons: 1- The sems is a light sip engine server with several applications (like as sbc) for using b2bua. All incoming and outgoing calls could go to sems server for doing b2bua like this: Incoming<=======>Kamailio<========>Sems<========>Kamailio<=======>outgoing 2- In sems, you could disable rtp realying. It forces sems to work just as b2bua without anchoring RTP 3- Easy to use different active profiles in routing. Just a couple of things there are in SEMS. For example, the sems adds itself (local IP) in Via header, and it couldn't be common in b2bua. like this: . . . INVITE sip:200@cloud.domain.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bK2f53.07a3fd9edaa8c8d609ab2ac6b01a087f.0 *================================>Kamailio public IP* *Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKsPRMNast;rport=5080 ===============================>private loopback sems ip* From: <sip:100@cloud.domain.com ;transport=UDP>;tag=6FFDB493-60EABB3600016ECD-379F9700 To: <sip:200@cloud.domain.com;transport=UDP> CSeq: 10 INVITE Call-ID: Y2M1ODQxNTZmMjdkZWZjN2U5MmMyYjBmN2Y2OGY1ODQ._leg2 Route: <sip:mo@127.0.0.1:5060;lr> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 3.3.25608 r25552 Content-Type: application/sdp Contact: <sip:127.0.0.1:5080;transport=udp> Content-Length: 246 . . . For this reason, Is there a way to avoid this issue? I know it is possible to do this by using other modules like textops and retransformation ? And why in Kamailio, there is no b2bua module to perform all b2bua functionality, yet? Thanks with Best Regards. -- --Mojtaba Esfandiari.S

6 9

How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 03 Aug '21

03 Aug '21

Hello all, I have a trouble that is related to the way of usage of dispatcher module. I am working with Kamailio 5.3.2 and using dispatcher module as load balancer to route calls to the media gateways. My dispatching routes are below as simply, route[DISPATCH] { if(ds_select_domain("2", "4")) { route(MYRELAY); } else { send_reply("503","Service Unavailable - No MGW"); exit; } } route[MYRELAY] { t_on_failure("MYFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if ($T_reply_code == 408 || $T_reply_code == 503) { if(ds_next_domain()) { route(MYRELAY); } else { send_reply("503","Service Unavailable"); exit; } } else { if(ds_select_domain("6", "4")) { route(MYVMRELAY); } else { send_reply(486,"Busy"); exit; } } } route[MYVMRELAY] { t_on_failure("MYVMFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYVMFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if(ds_next_domain()) { route(MYVMRELAY); } else { send_reply("503","Service Unavailable"); exit; } } I have multiple media gateways(setid=2) and voicemail servers(setid=6). As can be seen from the configuration, after first routing, for transaction reply code except 408 or 503, I am routing call to the voicemail server using dispatcher. In that phase, dispatcher module is remembering previously loaded destinations that come from first ds_select_domain and module try to route these destinations in the case that all voicemail servers are unavailable. So basically, I want to unload all destinations comes from previous ds_select_domain. From the documentation, i tried ds_load_update and ds_load_unset methods but not worked for me. Is there any way to overcome that problem or any right usages of these methods. Thank you, Serdar

2 2

Re: [SR-Users] How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 02 Aug '21

02 Aug '21

Hello Daniel, I removed xavps using with xavp_rm and problem solved. Thanks for your helping Serdar > Hello, > > delete the xavps based on the names you set via modparams xavp_dst and > xavp_ctx. > > Cheers, > Daniel > > On 30.07.21 11:14, Serdar GÜÇLÜER wrote: > >/Hello all, />//>/I have a trouble that is related to the way of usage of dispatcher > module. />//>/I am working with Kamailio 5.3.2 and using dispatcher module as load />/balancer to route calls to the media gateways. />//>/My dispatching routes are below as simply, />//>/route[DISPATCH] { />/ if(ds_select_domain("2", "4")) { />/ route(MYRELAY); />/ } else { />/ send_reply("503","Service Unavailable - No MGW"); />/ exit; />/ } />/} />/route[MYRELAY] { />/ t_on_failure("MYFAILURE"); />/ if(!t_relay()) { />/ sl_reply_error(); />/ } />/ exit; />/} />/failure_route[MYFAILURE] { />/ route(NATMANAGE); />/ revert_uri(); />/ if (t_is_canceled()) { />/ exit; />/ } />/ if ($T_reply_code == 408 || $T_reply_code == 503) { />/ if(ds_next_domain()) { />/ route(MYRELAY); />/ } else { />/ send_reply("503","Service Unavailable"); />/ exit; />/ } />/ } else { />/ if(ds_select_domain("6", "4")) { />/ route(MYVMRELAY); />/ } else { />/ send_reply(486,"Busy"); />/ exit; />/ } />/ } />/} />/route[MYVMRELAY] { />/ t_on_failure("MYVMFAILURE"); />/ if(!t_relay()) { />/ sl_reply_error(); />/ } />/ exit; />/} />/failure_route[MYVMFAILURE] { />/ route(NATMANAGE); />/ revert_uri(); />/ if (t_is_canceled()) { />/ exit; />/ } />/ if(ds_next_domain()) { />/ route(MYVMRELAY); />/ } else { />/ send_reply("503","Service Unavailable"); />/ exit; />/ } />/} />//>//>/I have multiple media gateways(setid=2) and voicemail />/servers(setid=6). As can be seen from the configuration, />/after first routing, for transaction reply code except 408 or 503, I />/am routing call to the voicemail server using dispatcher. />/In that phase, dispatcher module is remembering previously loaded />/destinations that come from first ds_select_domain />/and module try to route these destinations in the case that all />/voicemail servers are unavailable. />/So basically, I want to unload all destinations comes from previous />/ds_select_domain. />//>/From the documentation, i tried ds_load_update and ds_load_unset />/methods but not worked for me. />//>/Is there any way to overcome that problem or any right usages of these />/methods. />//>/Thank you, />//>/Serdar />//>//>/__________________________________________________________ />/Kamailio - Users Mailing List - Non Commercial Discussions />/* sr-users at lists.kamailio.org > <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> />/Important: keep the mailing list in the recipients, do not reply only > to the sender! />/Edit mailing list options or unsubscribe: />/* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> / > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 1

Access via script to location data in the memory
by Social Boh 31 Jul '21

31 Jul '21

Hello List, is there a way to access location data from kamailio script? I mean data I can see with the command: kamctl ul show Regards -- --- I'm SoCIaL, MayBe

3 11

Guidelines on E164 conversions
by Melissa Frasik 30 Jul '21

30 Jul '21

Hi all, I'm running into some issues with E164 conversions and was wondering if there was a decent standardized method for conversions. Conversions using $rU seem to be working great, but $fU is behaving badly. This is what I have so far: xlog("$ci - caller id was $avp(src=>callerid)"); if($avp(src=>callerid) =~ "^\+"){ xlog("$ci - caller id already E164"); }else if($(rU{s.substr,0,$(avp(src=>countrycode){s.len})}) == $avp(src=>countrycode)){ $avp(src=>callerid) = "+" + $avp(src=>callerid); $fU = $avp(src=>callerid); $avp(uacreplacefromdisplay) = $fU; $avp(uacreplacefromuri) = $fu; xlog("$ci - Adding plus to make caller id e164"); }else{ $fU = "+" + $avp(src=>countrycode) + $fU; $avp(uacreplacefromdisplay) = $fU; $avp(uacreplacefromuri) = $fu; xlog("$ci - converting domestic caller id number to $fU ($fu) - $avp(uacreplacefromdisplay) $avp(uacreplacefromuri)"); } Previously, I'd found a suggestion mentioning using uac_replace_from in the route[RELAY] (which is the purpose of the uacreplacefrom avps), but $fU does not appear to be changing its value. Is there something I'm doing incorrectly?

2 4

How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 30 Jul '21

30 Jul '21

Hello all, I have a trouble that is related to the way of usage of dispatcher module. I am working with Kamailio 5.3.2 and using dispatcher module as load balancer to route calls to the media gateways. My dispatching routes are below as simply, route[DISPATCH] { if(ds_select_domain("2", "4")) { route(MYRELAY); } else { send_reply("503","Service Unavailable - No MGW"); exit; } } route[MYRELAY] { t_on_failure("MYFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if ($T_reply_code == 408 || $T_reply_code == 503) { if(ds_next_domain()) { route(MYRELAY); } else { send_reply("503","Service Unavailable"); exit; } } else { if(ds_select_domain("6", "4")) { route(MYVMRELAY); } else { send_reply(486,"Busy"); exit; } } } route[MYVMRELAY] { t_on_failure("MYVMFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYVMFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if(ds_next_domain()) { route(MYVMRELAY); } else { send_reply("503","Service Unavailable"); exit; } } I have multiple media gateways(setid=2) and voicemail servers(setid=6). As can be seen from the configuration, after first routing, for transaction reply code except 408 or 503, I am routing call to the voicemail server using dispatcher. In that phase, dispatcher module is remembering previously loaded destinations that come from first ds_select_domain and module try to route these destinations in the case that all voicemail servers are unavailable. So basically, I want to unload all destinations comes from previous ds_select_domain. From the documentation, i tried ds_load_update and ds_load_unset methods but not worked for me. Is there any way to overcome that problem or any right usages of these methods. Thank you, Serdar

2 1

Setting up uacreg
by Duncan Turnbull 30 Jul '21

30 Jul '21

Hi there I am a new user of Kamailio and we are trying to use it to be as a front end for our asterisk pbx. We are running on Ubuntu 18.04 and Kamailio 5.3.8 with Siremis Rather than proxying the request through to asterisk we are trying to use uacreg to send a login to asterisk. Asterisk will think all the users are appear from the proxy but thats okay. Initially this is just for external users but eventually all phones etc will register via Kamailio and we will have the trunks there (and split them across another kamailio but thats another job) If I add a user to the uacreg then when I register to Kamailio it sends a register request but to the realm in the uacreg table and the matching port Kamailio is running on. Is this because somewhere we have set Kamailio to directly proxy on and we need to turn that off first? This is our uacreg table mysql> select * from uacreg; +----+--------+------------+------------+------------+-----------+-----------+---------------+---------------+----------+--------------------+---------+-------+-----------+--------+ | id | l_uuid | l_username | l_domain | r_username | r_domain | realm | auth_username | auth_password | auth_ha1 | auth_proxy | expires | flags | reg_delay | socket | +----+--------+------------+------------+------------+-----------+-----------+---------------+---------------+----------+--------------------+---------+-------+-----------+--------+ | 1 | testuser | testuser | ourdomain.com | 88 | 10.8.8.20 | 10.8.8.20 | 88 | password | '' | sip:10.8.8.20:5060 | 360 | 0 | 3 | | +----+--------+------------+------------+------------+-----------+-----------+---------------+---------------+----------+--------------------+---------+-------+-----------+--------+ 1 row in set (0.00 sec) All pointer, guides and recommendations will be welcome Thanks very much Cheers Duncan

3 8

Guidance tracking down "qm_free(): BUG: freeing already freed pointer"
by Anthony Joseph Messina 29 Jul '21

29 Jul '21

I'm seeking guidance on how to track down "qm_free(): BUG: freeing already freed pointer" which occurs only on branched calls. These errors don't crash so I don't get any core dumps. The different log entries below are the result of me selectively disabling sections of the script that apply in branch route, all to no avail. I'm running Kamailio on the current tip of the 5.5 branch (1f9f6fff6e). I'm reviewing https://www.kamailio.org/wiki/tutorials/troubleshooting/memory in the meantime as a place to start. version: kamailio 5.5.1-5.git1f9f6fff6e.fc34 (x86_64/linux) 7abebb flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f04b8f47e90), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f04b8f06c70), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f08e8f9cf80), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afafa60d8), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/sdp/sdp.c: init_p_payload_attr(183) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afae2a018), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afae431e8), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afafa3900), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f5fd4044ef0), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7ffbb2d2e2f0), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f6d1a50b1d8), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc19c165c70), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc19c17adb8), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc19c125790), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4d969c9d48), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc29bdca3d0), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc2c2eccc80), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc2c2ec52c0), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f910c04e240), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa351286b18), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa3512a7ae8), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa0777d9e70), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa0777d7a80), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4f2c92cf20), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4f2c8afa00), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4f2c92aa48), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fe55825eb30), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/sdp/sdp.c: init_p_payload_attr(183) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f6526db8378), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fcd9b465980), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7ff7ca92ff60), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f978a2dc938), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f978a2b7eb8), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa1556c1610), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fd81dfbba78), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fb2750d9988), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fb2750e1b28), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f32f7b45d30), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f35849e1940), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fd3a2c04280), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f5de857a040), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring -- Anthony - https://messinet.com CCED 8BFB 88D9 18F6 9F68 93DC 45C4 A8CE A64E 88E3

2 6

BYE not accepted by MS TEAMS
by João Vitor Arruda 29 Jul '21

29 Jul '21

Hello all, I recently built an SBC to integrate Kamailio with MS Teams following Henning Westerholt blob post and all the other awesome tips I was able to find here in the list. Everything is working great. The only missing piece I would like some thoughts on is about why MS Teams is not accepting the BYE sent from Kamailio. It eventually replies with "504 Server Time-out" The below trace shows the entire call. |||||||||||||||||||| ==================== tag: rcv pid: 3381 process: 63 time: 1627070235.086307 date: Fri Jul 23 19:57:15 2021 proto: tls ipv4 srcip: 52.114.76.76 srcport: 4672 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ INVITE sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone;transport=tls SIP/2.0 FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 TO: <sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bK60895c91 RECORD-ROUTE: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> CONTACT: <sip:api-du-b-usea2.pstnhub.microsoft.com:443 ;x-i=a82f1d71-4e21-42a8-8389-2ae26e54610f;x-c=c9b2aa4f658f5788aa7090bef0f5e35e/d/8/b14037a741dd46a79c74dce15912782c> CONTENT-LENGTH: 1103 MIN-SE: 300 SUPPORTED: timer USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.EUNO.0 CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY PRIVACY: id SESSION-EXPIRES: 3600 v=0 o=- 599625 0 IN IP4 127.0.0.1 s=session c=IN IP4 52.113.58.181 b=CT:10000000 t=0 0 m=audio 50122 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118 c=IN IP4 52.113.58.181 a=rtcp:50123 a=ice-ufrag:fTy/ a=ice-pwd:tQKGN0Szi7Q/vTdf6SiXUSpj a=rtcp-mux a=candidate:1 1 UDP 2130706431 52.113.58.181 50122 typ srflx raddr 10.0.33.230 rport 50122 a=candidate:1 2 UDP 2130705918 52.113.58.181 50123 typ srflx raddr 10.0.33.230 rport 50123 a=candidate:2 1 tcp-act 2121006078 52.113.58.181 49152 typ srflx raddr 10.0.33.230 rport 49152 a=candidate:2 2 tcp-act 2121006078 52.113.58.181 49152 typ srflx raddr 10.0.33.230 rport 49152 a=label:main-audio a=mid:1 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:PC00Mgk39GjTYJ+mUSNQg++TU6zJKar4VYDj4/ET|2^31 a=sendrecv a=rtpmap:104 SILK/16000 a=rtpmap:9 G722/8000 a=rtpmap:103 SILK/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=ptime:20 |||||||||||||||||||| ==================== tag: snd pid: 3381 process: 63 time: 1627070235.089106 date: Fri Jul 23 19:57:15 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 100 trying -- your call is important to us FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 TO: <sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bK60895c91;rport=4672 Server: kamailio (5.3.7 (x86_64/linux)) Content-Length: 0 |||||||||||||||||||| ==================== tag: snd pid: 3291 process: 18 time: 1627070235.111825 date: Fri Jul 23 19:57:15 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 t:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e CSEQ: 1 INVITE m:<sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp> User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Accept:application/sdp Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces u:talk,hold,conference,refer l:0 Via: SIP/2.0/TLS 52.114.76.76:5061;rport=4672;branch=z9hG4bK60895c91 Record-Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on>,<sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XBranch: z9hG4bKe7d4.5ae277881edb9c04a7ac65af016e582e.0 |||||||||||||||||||| ==================== tag: snd pid: 3298 process: 21 time: 1627070245.043621 date: Fri Jul 23 19:57:25 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 183 Session Progress FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 t:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e CSEQ: 1 INVITE m:<sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp> User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces u:talk,hold,conference,refer c:application/sdp Content-Disposition:session l:323 Via: SIP/2.0/TLS 52.114.76.76:5061;rport=4672;branch=z9hG4bK60895c91 Record-Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on>,<sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XBranch: z9hG4bKe7d4.5ae277881edb9c04a7ac65af016e582e.0 v=0 o=FreeSWITCH 1627045233 1627045234 IN IP4 SBC_IP_ADDR s=FreeSWITCH c=IN IP4 SBC_IP_ADDR t=0 0 m=audio 49692 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eMGqj0jubamtvLsI/E/YSj8UqNKgUB5DTHGWc7PY a=nortpproxy:yes |||||||||||||||||||| ==================== tag: snd pid: 3293 process: 19 time: 1627070255.938770 date: Fri Jul 23 19:57:35 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 200 OK FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 t:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e CSEQ: 1 INVITE m:<sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp> User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces u:talk,hold,conference,refer c:application/sdp Content-Disposition:session l:323 Via: SIP/2.0/TLS 52.114.76.76:5061;rport=4672;branch=z9hG4bK60895c91 Record-Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on>,<sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XBranch: z9hG4bKe7d4.5ae277881edb9c04a7ac65af016e582e.0 v=0 o=FreeSWITCH 1627045233 1627045234 IN IP4 SBC_IP_ADDR s=FreeSWITCH c=IN IP4 SBC_IP_ADDR t=0 0 m=audio 49692 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eMGqj0jubamtvLsI/E/YSj8UqNKgUB5DTHGWc7PY a=nortpproxy:yes |||||||||||||||||||| ==================== tag: rcv pid: 3370 process: 57 time: 1627070255.957735 date: Fri Jul 23 19:57:35 2021 proto: tls ipv4 srcip: 52.114.76.76 srcport: 4672 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ ACK sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp SIP/2.0 FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 TO: <sip:+5515XXXXXXXXX@SBC-FQDN:5061>;user=phone;tag=e55N8paBUgZac CSEQ: 1 ACK CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKe3747d68 CONTACT: <sip:api-du-b-usea2.pstnhub.microsoft.com:443 ;x-i=a82f1d71-4e21-42a8-8389-2ae26e54610f;x-c=c9b2aa4f658f5788aa7090bef0f5e35e/d/8/b14037a741dd46a79c74dce15912782c> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.EUNO.0 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY Route: <sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> P-SR-XUID: atpsh-60fb1b5d-d35-1 |||||||||||||||||||| ==================== tag: snd pid: 3302 process: 23 time: 1627070275.228363 date: Fri Jul 23 19:57:55 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.75.24 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ BYE sip:52.114.76.76:4672;x-i=a82f1d71-4e21-42a8-8389-2ae26e54610f;x-c=c9b2aa4f658f5788aa7090bef0f5e35e/d/8/b14037a741dd46a79c74dce15912782c SIP/2.0 Record-Route: <sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=e55N8paBUgZac;lr=on> Record-Route: <sip:192.168.1.253:5060;r2=on;ftag=e55N8paBUgZac;lr=on> Via: SIP/2.0/TLS SBC_IP_ADDR:5061;branch=z9hG4bK92fc.fcb147cf8e5e6e6e960d0f2401ad6383.0 v:SIP/2.0/UDP 192.168.1.254;received=192.168.1.254;rport=5060;branch=z9hG4bKgvXrQrcNjNj3m f:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac t:"Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 i:c9b2aa4f658f5788aa7090bef0f5e35e CSeq:38981345 BYE User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces Reason:Q.850;cause=16;text="NORMAL_CLEARING" Max-Forwards: 64 l:0 Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XUID: atpsh-60fb1b5d-d35-1 |||||||||||||||||||| ==================== tag: rcv pid: 3385 process: 65 time: 1627070300.423499 date: Fri Jul 23 19:58:20 2021 proto: tls ipv4 srcip: 52.114.75.24 srcport: 5061 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 504 Server Time-out FROM: <sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac TO: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 CSEQ: 38981345 BYE CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e VIA: SIP/2.0/TLS SBC_IP_ADDR:5061;branch=z9hG4bK92fc.fcb147cf8e5e6e6e960d0f2401ad6383.0 CONTENT-LENGTH: 0 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY SERVER: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.EUWE.3 Via: SIP/2.0/UDP 192.168.1.254;received=192.168.1.254;rport=5060;branch=z9hG4bKgvXrQrcNjNj3m Record-Route: <sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=e55N8paBUgZac;lr=on>,<sip:192.168.1.253:5060 ;r2=on;ftag=e55N8paBUgZac;lr=on> P-SR-XBranch: z9hG4bK92fc.fcb147cf8e5e6e6e960d0f2401ad6383.0

2 2

Accessing Transformations in KEMI
by Ben Kaufman 28 Jul '21

28 Jul '21

Sorry to bother on this, but is it possible to access Kamailio transformations in KEMI, and if so, how? In Kamailio's native scripting language I could do: $var(pai_user) = $(hdr(P-Asserted-Identity){tobody.user}); In KEMI, I can easily get the full header like this: KSR.hdr.get("P-Asserted-Identity") But I don't see how to access the transformations. Am I missing it? Ben Kaufman

3 6

Password for pi_framework.xml - Xhttp_pi module
by Abdirahman Osman 27 Jul '21

27 Jul '21

Hello everyone, In the xhttp_pi file of pi_framework.xml,is there a way I can pass the password to the xhttp_pi module as a pre-processing define? <db_url id="postgres">postgres://kamailiorw:kamailiorw@localhost /kamailiorw</db_url> Thank you! Abdirahman

2 1

Info: Kamailio World Online - Sep 1-2, 2021 - Celebrating 20 Years Of Development
by Daniel-Constantin Mierla 26 Jul '21

26 Jul '21

Hello, given the current situation across the world, no physical Kamailio World Conference is planned in 2021. However, this year the project celebrates 20 years of development -- according to commit history, the first one was done on the 3rd of September, 2021: > ❯ git log --reverse > commit 512dcd9838f5aeb0bfccd02cfc156067b7f02b71 > Author: Andrei Pelinescu-Onciul <andrei(a)iptel.org> > Date: Mon Sep 3 21:27:11 2001 +0000 > Initial revision To mark this notable milestone, we are going to organize an online event during Sep 1-2, 2021, similar to the one from 2020 -- the website for the event is available at: * https://www.kamailioworld.com/k09-online/ Mark the date in your calendar, more details about the content of the event will be published in the near future! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

New user checking on SIP proxying use case
by Nick Hurlburt 22 Jul '21

22 Jul '21

Hello all, My name is Nick Hurlburt, and work with a nonprofit tech organization that provides a software platform for child helplines around the world. I am a new user to Kamailio, and I'm interested in figuring out if my use case is appropriate, getting some tips on how to go about it, and possibly enlisting some expert help. Our service is built on top of Twilio, but we are working in some countries where Twilio doesn't have phone numbers. So in those areas, we are working with local telecoms on ways to integrate using a SIP connection. In one particular example, the telecom company can send SIP INVITEs but can only address them to a static IP. However, Twilio needs to receive the invite addressed to a fully-qualified domain name (eg, example-setup.sip.twilio.com). The idea arose to use Kamailio as a proxy to accept the static IP-addressed request and then rewrite the request to pass it along to a domain name. Then once the connection is established between the two endpoints, the media can flow directly between them. I have a software engineering background but still coming up to speed when it comes to VoIP technology. I'd be interested in: - does this sound like a good use case for Kamailio? - is there any specific documentation you could point me to for this use case? (I have been looking at the Kamailio documentation, but a quick pointer could save a lot of time in digging through it) - if anyone from the community would be open to helping us develop a prototype/proof-of-concept instance of this, especially if you're open to contributing to a nonprofit on a volunteer or discounted rate, I'd love to talk more Thank you, Nick

4 4

Kamailio/rtpengine not populating mos avp stats?
by Anthony Joseph Messina 22 Jul '21

22 Jul '21

I'm using Kamailio compiled from the current 5.5 tip (1f9f6fff6e), and rtpengine compiled from the current master (6e160da4). While rtpengine is working quite well, within the past month or so, I've noticed that Kamailio isn't populating the mos* related AVPs. I see continued upstream rtpengine work stream handling, etc. -- should I wait or try to debug? Thanks -A -- Anthony - https://messinet.com CCED 8BFB 88D9 18F6 9F68 93DC 45C4 A8CE A64E 88E3

3 6

How to get Stir/Shaken certificates
by ryan embgrets 22 Jul '21

22 Jul '21

Greetings. Does someone know where one can get stir/shaken certificates? Are they same as letsencrypt or do I need to obtain them from some other authority. If someone could provide me with a link to that authority it will be very helpful. Thanks again for your help and time. Ryan

4 5

Question about MS Teams integration
by Володимир Іванець 22 Jul '21

22 Jul '21

Hello all! Sorry, this is not a direct Kamailio question. I would like to ask for some help from those who already did this. I'm trying to integrate my testKamailio installation with MS Teams. I added the "mydomain.com" to the Domains section (I kept the " mydomain.onmicrosoft.com"). When I tried to create SBC with a " kamailio-dev2.mydomain.com" FQDN I got this error message: We* can't use the "kamailio-dev2.mydomain.com <http://kamailio-dev2.mydomain.com>" domain as it hasn't be set up in the organization. Try again. If you continue to have problems, contact Microsoft customer support.* There were no issues creating a dummy SBC with a " kamailio-dev2.mydomain.onmicrosoft.com" FQDN. Does anyone know how to set up a domain in the organization from MS perspective? Thank you very much! Regards, Volodymyr Ivanets.

4 9

serial forking kamailio - twilio
by Filippo Graziola 21 Jul '21

21 Jul '21

Hi all, I have a question for my kamailio configuration which works like this: - INVITE arrives to kamailio - HTTP async request to API endpoint - Resume, use of rtjson module for parallel forking and loop to kamailio itself - One of the invite is "transformed" in serial forking with rtjson module and it is sent with a delay to TWILIO SIP Domain (landline/mobile phone numbers) Twilio is giving me some troubles because it answers with a 200 OK but the call it is still in establishing/ringing on the "landline" side and as expected when reaching the timeout (no answer from phone number on TWILIO side) kamailio is receiving a BYE. Now the question, how to handle next number to call through Twilio? I was thinking of using rtjson_next_route() when handling the in-dialog BYE but as it is not a failure route so it does not work, or I'm doing it wrong. Thanks in advance for any help/hint. Filippo

2 2

Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record
by Henning Westerholt 20 Jul '21

20 Jul '21

Hello, I have not tested it, but the dialog database will probably only help in a restart scenario with regarding to dialogs, at least with a standard cfg. Regarding the example below, this would mean starting the primary Kamailio again, then the cdrs should be generated during the hangup from the primary. You can test also to restart the secondary Kamailio, it should then also read the dialogs from the replicated table and generate cdrs at hangup. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Social Boh <social(a)bohboh.info> Sent: Tuesday, July 20, 2021 2:02 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record Hello, with dialog db_mode 1 and Primary/Primary MariaDB Replication If I switch Off primary Kamailio during a call and move the call to Replica and then hangup, no entry on acc_cdrs table. acc table is ok. Regards --- I'm SoCIaL, MayBe El 20/07/2021 a las 4:09 a. m., Henning Westerholt escribió: Hello, Usage of DB to store dialogs. See https://kamailio.org/docs/modules/devel/modules/dialog.html#dialog.p.db_mode – in particular mode 1 and 3. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org><mailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Social Boh Sent: Tuesday, July 20, 2021 12:55 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org><mailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record Hello what means "DB loading" ? Regards --- I'm SoCIaL, MayBe El 19/07/2021 a las 9:25 a. m., Henning Westerholt escribió: Hello, so, you are saying that in a DMQ replicated case the acc CDR generation is not working in failover cases. I just looked briefly to the code; others might have more details. But I think this is because the acc module do not register to a callback function for dialog loading on startup for the DMQ replication. This is done for DB loading, but seems to be not implemented yet for DMQ synchronisation. You could open a feature request on our github tracker for this. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org><mailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Social Boh Sent: Monday, July 19, 2021 1:58 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org><mailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record Hi, I'm using 5.4.6 I switch off Kamailio on the first server and keepalived move the virtual IP on the second server so First Kamailio is off. This during the call. Before switch off Kamailio and during the call I can see data dialog on both server but in the LOG /var/log/messages in my case: server 1 INVITE server 2 BYE acc_cdrs table empty on both servers. acc table first server INVITE; acc table second server BYE Regards --- I'm SoCIaL, MayBe El 18/07/2021 a las 2:33 p. m., M Arqum CH escribió: Hi SoCIal, which Kamailio version you are using. did you able to get the dialog information using the dialog command. https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_l… when you received bye on secondary server, is that server recognized the bye. On Sun, Jul 18, 2021 at 3:38 AM Social Boh <social(a)bohboh.info<mailto:social@bohboh.info>> wrote: Hello, if during a call I stop Primary Kamailio the Secondary Kamailio take de Virtual IP but don't save call data: In the Primary database, acc table, I have INVITE data but not BYE data; no data in acc_cdrs for the call. In the Secondary Kamailio, acc table, I have BYE data but not INVITE data. in the file LOG no data about call cdr. Maybe a solution is write call data whene the call end? Regards --- I'm SoCIaL, MayBe El 17/07/2021 a las 11:34 a. m., M Arqum CH escribió: Make sure that your table columns matched the field's you are trying to insert, please check the syslog for acc module log, acc module print the cdr in log facility too, if acc module facing any issue while writing cdr in db, it will throw error in logs. Hope this will help you. On Sat, Jul 17, 2021, 7:33 PM Social Boh <social(a)bohboh.info<mailto:social@bohboh.info>> wrote: Hello, i tested with DMQ module and happens the same. DMQ module loaded before dialog module and on dialog module parameters: modparam("dialog", "enable_dmq", 1) During a call I can see on both Kamailio servers the dialog data but when switch off the active kamailio server and the second kamailio server take the control and hangup the call, no data acc_cdrs table. Regards --- I'm SoCIaL, MayBe El 17/07/2021 a las 1:24 a. m., M Arqum CH escribió: Hi Social, Replication of dialog using dmq will solve your problem, you need to load dmq module before loading dialog module and enable replication of dialogs in dialog module, please check dialog module conf option for more details. Cheers On Fri, Jul 16, 2021, 7:32 PM Social Boh <social(a)bohboh.info<mailto:social@bohboh.info>> wrote: Im' testinf Kamailio HA with Keepalived. I have a problem with acc_cdrs table in this situation: A call B and when the two users are in the call I switch off the active Kamailio. Keepalived pass the Virtual IP to the secondary Kamailio and audio still work because I'm using a remote RTPENGINE server for both Kamailio. The problem is when the call end. On the acc table i can see INVITE and BYE entries but on the acc_cdrs table nothing. Second problem the dialog still present on the table dialog in a Primary/Primary replication. Is there a way to solve it? Thank you Regards -- --- I'm SoCIaL, MayBe __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

Kamailio KeepAlived HA acc_cdrs table record
by Social Boh 20 Jul '21

20 Jul '21

Im' testinf Kamailio HA with Keepalived. I have a problem with acc_cdrs table in this situation: A call B and when the two users are in the call I switch off the active Kamailio. Keepalived pass the Virtual IP to the secondary Kamailio and audio still work because I'm using a remote RTPENGINE server for both Kamailio. The problem is when the call end. On the acc table i can see INVITE and BYE entries but on the acc_cdrs table nothing. Second problem the dialog still present on the table dialog in a Primary/Primary replication. Is there a way to solve it? Thank you Regards -- --- I'm SoCIaL, MayBe

3 11

Kamailio python_exec getting randomly TypeError
by Voip support 19 Jul '21

19 Jul '21

Dear Kamailio users, I am trying to use a python script using the module app_python3 to do some billing calculations. The only argument i want to pass to the python script is duration which i calculate in event_route[dialog:start] { $dlg_var('call_start') = $TS; event_route[dialog:end] { $var('v_duration') = $TS - $dlg_var('call_start'); So my $var('v_duration') has the duration of the call in seconds. Then i execute from my kamailio.cfg: python_exec("calculateCost","$var('v_duration')"); The script works randomly - sometimes all is good and sometimes it throw error : app_python3 [python_support.c:156]: python_handle_exception(): apy_exec: calculateCost(4): Unhandled exception in the Python code: TypeError: an integer is required (got type NoneType) The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio-python.py", line 282, in calculateCost duration = int(duration) It looks like the kamailio log show calculateCost(4) so argument 4 is passed to python function but once it executes it appear like there is no string/integer but NoneType argument. I cannot figure this out why this randomly happens - maybe it's an app_python3 module issue? Looking to hear back from you! BR, Tom

2 1

please help to configure tls in kamailio for webrtc client like simpl5
by ThanhTruong 19 Jul '21

19 Jul '21

Dear, I am trying to make a test to see how can webrtc work in kamailio and tls. I was try some link : 1 <https://github.com/havfo/WEBRTC-to-SIP>. https://github.com/havfo/WEBRTC-to-SIP <https://github.com/havfo/WEBRTC-to-SIP> 2. https://telecom.altanai.com/2018/09/04/kamailio-webrtc-sip-server/ <https://telecom.altanai.com/2018/09/04/kamailio-webrtc-sip-server/> And Generate CA from the link: https://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates <https://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates> Then, I tried to login with simpl5 and i got the message back Disconnected: Failed to connet to the server I do not know how can i fix that issue, look like TLS configure has issue with that. the log is showing something like Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094416:SSL routines:ssl3_read_bytes:sslv3 alert certificate unknown Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 27.65.214.194 Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 172.31.44.170 Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: <core> [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f4233c60fc0 r: 0x7f4233c610e8 (-1) My tls.cfg is : [server:default] method = SSLv23 verify_certificate = yes require_certificate = yes private_key = /etc/certs/my.domain/key.pem certificate = /etc/certs/my.domain/cert.pem ca_list = /etc/certs/demoCA/cert.pem [client:default] verify_certificate = yes require_certificate = yes Please help me to register sipml5 or how can i create CA for TLS in kamailio to use with webrtc. Thank all.

4 14

Active Profile for sipreply in SEMS
by Mojtaba 17 Jul '21

17 Jul '21

Hello, Does anybody know how i could change white filter in siprelpy in sems? In Kamailio, I got "200 OK" from UE, then add a new header (SBC: mt-reply) and send it to sems server. The sems server remove it: [#7f410fbfb700/4560] [inplaceHeaderFilter, HeaderFilter.cpp:226] DEBUG: erasing header 'sbc' by whitelist I really don't want to remove it. Actually if I knew how i active profile for sip replying in sems, I would be able to solve this. Thanks with regards. -- --Mojtaba Esfandiari.S

2 2

get defined names in kemi script
by Riccardo Villa 16 Jul '21

16 Jul '21

Hello guys, do you know if it’s possible for a kemi script to know if a name is defined (with #!define) in kamailio.cfg? It would be useful to modify kemi script logic depending on feature/module presence. Thanks in advance for your help. Regards, Riccardo -- -- NOTICE: This electronic mail transmission may contain confidential information and is intended only for the person(s) named. Any use, copying, or disclosure by any other person is strictly prohibited. If you have received this transmission in error, please notify the sender via e-mail.

3 6

rtpengine play_media
by Дмитрий Плаксюк 15 Jul '21

15 Jul '21

Greetings. I'm trying to set up annoncement playback for both parts of the call. I have to play the file after the connection is established. Stop_media (all) must be executed before playing It turns out the construction: .... block_media ("all"); play_media ("all file = / etc / kamailio / sounds / mg1.wav"); Can't find a way to do unblock_media ("all"); after the file has finished playing. There are no sip messages in this process. The call has already been established.

1 0

DMQ node list
by Aidar Kamalov 14 Jul '21

14 Jul '21

Hello, I have a kamailio cluster with two nodes kamailio-1.local: 10.101.2.119 kamailio-2.local: 10.101.2.247 with dmq configs: kamailio-1: modparam("dmq", "server_address", "sip:MY_IP4_ADDR:5060") modparam("dmq", "notification_address", "kamailio-2.local:5060") kamailio-2: modparam("dmq", "server_address", "sip:MY_IP4_ADDR:5060") modparam("dmq", "notification_address", "kamailio-1.local:5060") All works fine, but kamcmd dmq.list_nodes on kamailio-1.local show me: { host: 10.101.2.247 port: 5060 resolved_ip: 10.101.2.247 status: active last_notification: 0 local: 0 } { host: kamailio-1.local port: 5060 resolved_ip: 10.101.2.119 status: active last_notification: 0 local: 0 } { host: kamailio-2.local port: 5060 resolved_ip: 10.101.2.247 status: active last_notification: 0 local: 0 } { host: 10.101.2.119 port: 5060 resolved_ip: 10.101.2.119 status: active last_notification: 0 local: 1 } and kamailio-2.local show me: { host: kamailio-2.local port: 5060 resolved_ip: 10.101.2.247 status: active last_notification: 0 local: 0 } { host: 10.101.2.119 port: 5060 resolved_ip: 10.101.2.119 status: active last_notification: 0 local: 0 } { host: kamailio-1.local port: 5060 resolved_ip: 10.101.2.119 status: active last_notification: 0 local: 0 } { host: 10.101.2.247 port: 5060 resolved_ip: 10.101.2.247 status: active last_notification: 0 local: 1 } So there are only two hosts in the cluster, but there are 4 records in dmq.list_nodes. Also in sip trace I can see that kamailio send DMQ messages to itself too (because it ip marked as local: 0 I think). Is it ok? Is it normal behavior? What should I do to fix it? Should multi_notify and DNS name with 2 A records fix it (and why?)? -- Aydar A. Kamalov

2 1

Re: [SR-Users] Subscriber Management using JSONRPC over HTTP
by Surabhi Garg 14 Jul '21

14 Jul '21

Hi Alex, By subscriber management I mean, auth_db module which actually works on subscriber table to add or alter sip account for authentications. Auth_db Module (kamailio.org)<https://kamailio.org/docs/modules/4.4.x/modules/auth_db.html#auth_db.f.auth…> So, similar like usrloc module which you mentioned having JSONRPC commands to alter Location table. I am looking for JSONRPC Command for subscriber module as well. As far as my understanding, currently there is no JSONRPC command available for subscriber table. So, if this is currently a known not offered feature / limitation shall we raise it as feature request if others also agree to its usefulness. What I'm seeking is more of an acknowledgment that this feature doesn't exist & have it's benefit. As I can see other community members either ends up directly interacting with DB for subscriber table provisioning. Maybe I can contribute on this too. Surabhi Garg Mobile - 91 783 888 7523 | Skype - surabhi.garg9 ________________________________ From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Alex Balashov <abalashov(a)evaristesys.com> Sent: 09 July 2021 01:27 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Subscriber Management using JSONRPC over HTTP I am not 100% sure what you mean by subscriber management, but if you mean what I think you mean, these might be the droids you’re looking for: https://kamailio.org/docs/modules/5.5.x/modules/usrloc.html#idm771 — Sent from mobile, with due apologies for brevity and errors. On Jul 8, 2021, at 3:55 PM, Surabhi Garg <surabhi.garg(a)outlook.com> wrote: Hi Team, Greetings! I was trying to use HTTP to do the subscriber management (CRUD operations) & for that I am using JSONRPC-S module (JSONRPC over HTTP). But like other modules e.g. dispatcher, location I am unable to find any RPC commands for doing subscriber management (checked in auth_db) module. So is there a way to achieve subscriber management using JSONRPC over HTTP. Few HTTP examples which I am using for dispatcher module are given below. Similar commands I am looking for subscriber management as well. curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.add\", \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.remove\”, \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" Surabhi Garg Mobile - 91 783 888 7523 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 2

DMQ_USRLOC module and calls
by Social Boh 13 Jul '21

13 Jul '21

Hello, I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) and (B) The two servers share the same domain and using Amazon Route53 to distributes 50% requests each Server. The problem is when I try to make a call from user A (registered on kamailio A) and the user B (registered on Kamailio B). The call follow this flow: User A --> KamailioA --> UserB go directly to userB because KamailioA know, via DMQ_USRLOC IP and port of UserB. With some Softphone the call work with audio without problem; with other Softphone USERB never answer the INVITE sends from Kamailio. Is it possible use this flow? USER A --> KamailioA --> KamailioB --> User B to resolve this kind of problem or is there other available option? Thank you -- --- I'm SoCIaL, MayBe

5 15

Change routing for Register
by H Yavari 13 Jul '21

13 Jul '21

Hi to all, In a scenario, we need to change the destination of users' registration requests when they send the first 'Register' to Kamailio-A and it replies 401 to the user, next time the user sends 'Register' to Kamailio-B. User------Register-------> Kamailio-A User<-----401--------------Kamailio-A User-------Register--------> Kamailio-B RFC says we can't use the Record-Route for register because it doesn't create a dialog.Is there any option, header that Kamailio-A puts in the 401, and after that users send register to the Kamailio-B? Thank you.BR,Hossein

3 3

Kamailio v5.4.6 Released
by Daniel-Constantin Mierla 13 Jul '21

13 Jul '21

Hello, Kamailio SIP Server v5.4.6 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.6. Note that 5.4 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.5, with v5.5.0 being release out of it. For more details about version 5.4.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/06/kamailio-v5-4-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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 Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

3 2

Using the dialog module. dialog was not destroyed with BYE message
by woenho＠gmail.com 08 Jul '21

08 Jul '21

According to the documentation a dialog is automatically destroyed when a BYE is received. But, dialog was not destroyed. What can I do? I am testing with kamailio 5.5 version

3 2

Subscriber Management using JSONRPC over HTTP
by Surabhi Garg 08 Jul '21

08 Jul '21

Hi Team, Greetings! I was trying to use HTTP to do the subscriber management (CRUD operations) & for that I am using JSONRPC-S module (JSONRPC over HTTP). But like other modules e.g. dispatcher, location I am unable to find any RPC commands for doing subscriber management (checked in auth_db) module. So is there a way to achieve subscriber management using JSONRPC over HTTP. Few HTTP examples which I am using for dispatcher module are given below. Similar commands I am looking for subscriber management as well. curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.add\", \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.remove\”, \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" Surabhi Garg Mobile - 91 783 888 7523

2 1

How to check If Alias is already Set in Contact Header
by Juha Heinanen 08 Jul '21

08 Jul '21

Transformations (e.g. nameaddr.uri and uri.param.name) on $ct are your friends. -- Juha

2 2

How to check If Alias is already Set in Contact Header
by Terry Tailor 08 Jul '21

08 Jul '21

Hi, I'm having the following scenario UA <---> Prox1 <----> Proxy2 <-----> B2BUA <------> UA Proxy2 and B2BUA are connected on LAN and Proxy1 is adding his own Contact Alias and Proxy 2 is adding his own Contact Alias I'm adding Alias via using the following function set_contact_alias("1") as it will remove other contact alias but is causing the problem in the case of BYE coming from B2BUA. Is there is a way to find if the contact alias is already set? How I can replace the local IP alias with domain name when request is coming from B2BUA and alias is added by Proxy2. Here are logs to show problem in 200 sent by B2BUA and proxy2 add alias 10.44.44.2~6000~1 Now in ACK it is causing the problem SIP/2.0 200 OK Via: SIP/2.0/UDP sip.prox1.example.com:5060 ;rport=5060;received=3.18.1.21;branch=z9hG4bK2d66.a89039bedfe3d1113132f04c94b8a757.0 Via: SIP/2.0/UDP 192.168.1.168:50279 ;received=109.8.2.1;rport=45607;branch=z9hG4bKPj84875d08eafc48bfa6c29de5fe245cea Record-Route: <sip:sip.proxy2.example;lr;did=e84.87e2;rtp=bridge> Record-Route: <sip:sip.prox1.example.com;lr;did=e84.7452;rtp=bridge> From: "8001" <sip:8001@sip.example.com>;tag=0b9915abae654e21845fd4d7f577ede9 To: <sip:2501@sip.example.com>;tag=as1f4f6b1c Call-ID: a0095b9868924b0681225a8884a06c22 CSeq: 3687 INVITE Server: Voip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:2501@10.44.44.2:6000;alias=10.44.44.2~6000~1> Content-Type: application/sdp Require: timer Content-Length: 281 Ack for 200 ACK sip:2501@10.44.44.2:6000 SIP/2.0 Via: SIP/2.0/UDP sip.proxy2.example.com:5060 ;branch=z9hG4bK2d66.3c10a3b45e5c8caacbc5853d5e6418b2.0 Via: SIP/2.0/UDP sip.proxy1.example.com:5060;rport=5060;received= 3.18.1.21 ;branch=z9hG4bK2d66.788312006d5f7aedc058435cd203fed3.0 Via: SIP/2.0/UDP 192.168.1.168:50279;received= 109.8.2.1 ;rport=45607;branch=z9hG4bKPjf003c3bfe4ba423e9a0dedd647511dd3 Max-Forwards: 68 From: "8001" <sip:8001@sip.example.com>;tag=0b9915abae654e21845fd4d7f577ede9 To: <sip:2501@sip.example.com>;tag=as1f4f6b1c Call-ID: a0095b9868924b0681225a8884a06c22 CSeq: 3687 ACK Content-Length: 0 Thanks in advance

1 0

SECSIPID Identity Decode
by Daniel W. Graham 06 Jul '21

06 Jul '21

I am unable to base64url decode the json payload in identity header generated by secsipid. (Using python for test) decoded_payload = url64.decode(‘payload’) UnicodeDecodeError: 'utf-8' codec can't decode byte 0xc2 in position 27: invalid continuation byte Header decodes fine this way but not payload. Is this an issue with the payload encoding? Kamailio 5.5 Daniel W. Graham, CTO CMSInter.net LLC DIRECT (989) 400-4230 INTERNET | TELEPHONE | MANAGED IT

5 28

I found bug in sipdump module.
by woenho 06 Jul '21

06 Jul '21

2 1

Re: [SR-Users] SECSIPID Identity Decode
by Daniel W. Graham 03 Jul '21

03 Jul '21

This was already pointed out, $mb. Thanks, -dan On Jul 2, 2021, at 4:40 PM, Daniel W. Graham <dan(a)cmsinter.net> wrote:

1 0

crash when sending reply
by Juha Heinanen 02 Jul '21

02 Jul '21

Kamailio 5.5 crashed when sending reply (bt full below). Before the crash there had been several mysql related error messages ("Lost connection to MySQL server during query" and "Too many connections") in syslog when async worker tried to insert accounting data. So perhaps this core dump has something to do with that. -- Juha Reading symbols from /usr/bin/sip-proxy...done. [New LWP 21324] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/bin/sip-proxy -f /etc/sip-proxy/sip-proxy.cfg -P /run/sip-proxy/sip-proxy.'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007fd274ce3b6e in run_trans_callbacks_internal (cb_lst=0x7fd2532503a8, type=512, trans=0x7fd253250330, params=0x7fff9b8eb2b0) at t_hooks.c:254 254 t_hooks.c: No such file or directory. (gdb) bt full #0 0x00007fd274ce3b6e in run_trans_callbacks_internal (cb_lst=0x7fd2532503a8, type=512, trans=0x7fd253250330, params=0x7fff9b8eb2b0) at t_hooks.c:254 cbp = 0x10 backup_from = 0x5613ccfb37f0 <def_list+16> backup_to = 0x5613ccfb37f8 <def_list+24> backup_dom_from = 0x5613ccfb3800 <def_list+32> backup_dom_to = 0x5613ccfb3808 <def_list+40> backup_uri_from = 0x5613ccfb37e0 <def_list> backup_uri_to = 0x5613ccfb37e8 <def_list+8> backup_xavps = 0x5613ccf56270 <_xavp_list_head> backup_xavus = 0x5613ccf56278 <_xavu_list_head> backup_xavis = 0x5613ccf56280 <_xavi_list_head> __func__ = "run_trans_callbacks_internal" #1 0x00007fd274ce41f7 in run_trans_callbacks_with_buf (type=512, rbuf=0x7fd253250400, req=0x7fd252d6dd20, repl=0x7fd2758bf0b0, flags=0) at t_hooks.c:303 params = {req = 0x7fd252d6dd20, rpl = 0x7fd2758bf0b0, param = 0x7fd254760f60, code = 486, flags = 0, branch = 0, t_rbuf = 0x7fd253250400, dst = 0x7fd253250450, send_buf = { s = 0x7fd25325b9f0 "@", len = 579}} trans = 0x7fd253250330 #2 0x00007fd274c88c15 in relay_reply (t=0x7fd253250330, p_msg=0x7fd2758bf0b0, branch=0, msg_status=486, cancel_data=0x7fff9b8eb580, do_put_on_wait=1) at t_reply.c:2133 relay = 0 save_clone = 0 buf = 0x7fd275ca0ff0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX.com;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"... res_len = 579 relayed_code = 486 relayed_msg = 0x7fd2758bf0b0 reply_bak = 0x7fd2510f1000 bm = {to_tag_val = {s = 0x7fd25216e4c0 "@\003", len = 1377234752}} totag_retr = 0 reply_status = RPS_COMPLETED uas_rb = 0x7fd253250400 to_tag = 0x7fd25216e820 reason = {s = 0x7fd25216e830 "0", len = 1394935248} onsend_params = {req = 0x9b8eb420, rpl = 0x7fd274c6b10d <futex_get+40>, param = 0x1, code = 1359984720, flags = 32722, branch = 0, t_rbuf = 0x7fff9b8eb460, dst = 0xa0679800, send_buf = { s = 0x19b8eb450 <error: Cannot access memory at address 0x19b8eb450>, len = 1359984720}} ip = {af = 2609820752, len = 32767, u = {addrl = {94643040895493, 0}, addr32 = {3436528133, 22035, 0, 0}, addr16 = {16901, 52437, 22035, 0, 0, 0, 0, 0}, addr = "\005B\325\314\023V\000\000\000\000\000\000\000\000\000"}} __func__ = "relay_reply" #3 0x00007fd274c8de6b in reply_received (p_msg=0x7fd2758bf0b0) at t_reply.c:2680 msg_status = 486 last_uac_status = 180 ack = 0x7fd25216e830 "0" ack_len = 459 branch = 0 reply_status = 32722 onreply_route = 2 cancel_data = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 1977055608}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 1977055608}}}} uac = 0x7fd2532505c0 t = 0x7fd253250330 lack_dst = {send_sock = 0x40, to = {s = {sa_family = 49168, sa_data = "\207u\322\177\000\000\220\266\216\233\377\177\000"}, sin = {sin_family = 49168, sin_port = 30087, sin_addr = { s_addr = 32722}, sin_zero = "\220\266\216\233\377\177\000"}, sin6 = {sin6_family = 49168, sin6_port = 30087, sin6_flowinfo = 32722, sin6_addr = {__in6_u = { __u6_addr8 = "\220\266\216\233\377\177\000\000L=\215s\322\177\000", __u6_addr16 = {46736, 39822, 32767, 0, 15692, 29581, 32722, 0}, __u6_addr32 = {2609821328, 32767, 1938636108, 32722}}}, sin6_scope_id = 1971831024}, sas = {ss_family = 49168, __ss_padding = "\207u\322\177\000\000\220\266\216\233\377\177\000\000L=\215s\322\177\000\000\360\300\207u\322\177", '\000' <repeats 13 times>, "\b\000\000\000\000\060Z\264u\322\177\000\000\213\t\342\316\023V\000\000;\b\342\316\023V\000\000\020\313\320\000\000\000\000\000\020\000\000\000\000\000\000\000I\001\000\000\000\000\000\000\000\230g\240\210\347A\357\300\266\216\233\377\177\000\000\000\230g\240\210\347", <incomplete sequence \357>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 0 '\000', proto_pad0 = 0 '\000', proto_pad1 = 0} backup_user_from = 0x5613ccfb37f0 <def_list+16> backup_user_to = 0x5613ccfb37f8 <def_list+24> backup_domain_from = 0x5613ccfb3800 <def_list+32> backup_domain_to = 0x5613ccfb3808 <def_list+40> backup_uri_from = 0x5613ccfb37e0 <def_list> backup_uri_to = 0x5613ccfb37e8 <def_list+8> backup_xavps = 0x5613ccf56270 <_xavp_list_head> backup_xavus = 0x5613ccf56278 <_xavu_list_head> backup_xavis = 0x5613ccf56280 <_xavi_list_head> replies_locked = 1 branch_ret = 0 prev_branch = 1971830800 blst_503_timeout = 32722 hf = 0xb5c750 onsend_params = {req = 0x7fff9b8eb5e0, rpl = 0xef41e788a0679800, param = 0x30, code = 0, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x7fd27d972cb0 <__syslog>, send_buf = { s = 0x5613ccec69b3 "INFO", len = -1603823616}} ctx = {rec_lev = 0, run_flags = 2, last_retcode = -1, jmp_env = {{__jmpbuf = {0, -5133089567146157739, 0, 140542026919088, 94643042412979, 536870912, -5133089567232140971, -1505551737941960363}, __mask_was_saved = 0, __saved_mask = {__val = {140735803210560, 13683472, 7765887371, 140735803209568, 94643041089583, 0, 17240315422543681536, 0, 0, 0, 0, 94643042412979, 17240315422543681536, 140735803209872, 94643041372964, 2866715017336}}}}} bctx = 0x7fd27587c010 keng = 0x0 ret = 0 evname = {s = 0x7fd274d3696f "on_sl_reply", len = 11} __func__ = "reply_received" #4 0x00005613ccbba0d4 in do_forward_reply (msg=0x7fd2758bf0b0, mode=0) at core/forward.c:764 new_buf = 0x0 dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 0, __ss_padding = '\000' <repeats 117 times>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 0 '\000', proto_pad0 = 0 '\000', proto_pad1 = 0} new_len = 0 r = 2 ip = {af = 1, len = 6356993, u = {addrl = {140735803210560, 140541891965104}, addr32 = {2609822528, 32767, 1972105392, 32722}, addr16 = {47936, 39822, 32767, 0, 61616, 30091, 32722, 0}, addr = "@\273\216\233\377\177\000\000\260\360\213u\322\177\000"}} s = 0xef41e788a0679800 <error: Cannot access memory at address 0xef41e788a0679800> len = -1685145520 __func__ = "do_forward_reply" #5 0x00005613ccbbc23c in forward_reply (msg=0x7fd2758bf0b0) at core/forward.c:865 No locals. #6 0x00005613ccc5eea9 in receive_msg ( buf=0x5613cee206f0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"..., len=667, rcv_info=0x7fd253b7cad8) at core/receive.c:587 msg = 0x7fd2758bf0b0 ctx = {rec_lev = 0, run_flags = 0, last_retcode = 1, jmp_env = {{__jmpbuf = {0, -5133089566915471019, 0, 140542026919088, 94643042412979, 536870912, -5133089567001454251, -1505551737941960363}, __mask_was_saved = 0, __saved_mask = {__val = {94643073835016, 65535, 17240315422543681536, 18446744073709551536, 2, 0, 140542026919088, 94643042412979, 536870912, 140735803211024, 94643040635195, 140735803211696, 65535, 65536, 4095, 140735803211580}}}}} bctx = 0x0 ret = 1 tvb = {tv_sec = 0, tv_usec = 0} tve = {tv_sec = 0, tv_usec = 0} diff = 0 inb = {s = 0x5613cee206f0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"..., len = 667} netinfo = {data = {s = 0x5613cee206f0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX;tag=as6964f29d\r\nCal"..., len = 667}, rcv = 0x7fd253b7cad8, dst = 0x0} keng = 0x0 evp = {data = 0x7fff9b8eba90, obuf = {s = 0x0, len = 0}, rcv = 0x7fd253b7cad8, dst = 0x0, req = 0x0, rpl = 0x0, rplcode = 0, mode = 0} cidlockidx = 0 cidlockset = 0 errsipmsg = 0 exectime = 0 __func__ = "receive_msg" #7 0x00005613ccd1cf15 in receive_tcp_msg (tcpbuf=0x7fd253b7ce60 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"..., len=667, rcv_info=0x7fd253b7cad8, con=0x7fd253b7cac0) at core/tcp_read.c:1424 buf = 0x5613cee206f0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"... bsize = 65535 blen = 65535 __func__ = "receive_tcp_msg" #8 0x00005613ccd1f554 in tcp_read_req (con=0x7fd253b7cac0, bytes_read=0x7fff9b8ebf34, read_flags=0x7fff9b8ebf3c) at core/tcp_read.c:1607 bytes = 667 total_bytes = 667 resp = 1 size = 0 req = 0x7fd253b7cbe8 dst = {send_sock = 0x7fff9b8ebec0, to = {s = {sa_family = 32062, sa_data = "\303\314\023V\000\000p\276\216\233\377\177\000"}, sin = {sin_family = 32062, sin_port = 52419, sin_addr = {s_addr = 22035}, sin_zero = "p\276\216\233\377\177\000"}, sin6 = {sin6_family = 32062, sin6_port = 52419, sin6_flowinfo = 22035, sin6_addr = {__in6_u = {__u6_addr8 = "p\276\216\233\377\177\000\000\030\000\000\000\000\000\000", __u6_addr16 = {48752, 39822, 32767, 0, 24, 0, 0, 0}, __u6_addr32 = {2609823344, 32767, 24, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 32062, __ss_padding = "\303\314\023V\000\000p\276\216\233\377\177\000\000\030", '\000' <repeats 15 times>, "\225q\\\037P\000\000\000\070,\214u\322\177\000\000\000\244\374\314\023V\000\000\000\000\000\000\001 \000\000\070,\214u\322\177\000\000\001\000\000\000\000\000\000\000\300ʷS\322\177\000\000\200\276\216\233\377\177\000\000\070,\214u\322\177\000\000\200\277\216\233\377\177\000\000\b\000\000\000\000\000\000", __ss_align = 20}}, id = 1, send_flags = {f = 1, blst_imask = 0}, proto = 31 '\037', proto_pad0 = 0 '\000', proto_pad1 = 0} c = 49 '1' ret = 0 __func__ = "tcp_read_req" #9 0x00005613ccd23eb4 in handle_io (fm=0x7fd2758c2c38, events=1, idx=-1) at core/tcp_read.c:1857 ret = 0 n = 1404553920 read_flags = RD_CONN_SHORT_READ con = 0x7fd253b7cac0 s = 1404554328 resp = 1 t = 526152085 ee = 0x0 __func__ = "handle_io" #10 0x00005613ccd0dd1e in io_wait_loop_epoll (h=0x5613ccfca240 <io_w>, t=2, repeat=0) at core/io_wait.h:1070 n = 1 r = 0 fm = 0x7fd2758c2c38 revents = 1 __func__ = "io_wait_loop_epoll" #11 0x00005613ccd25981 in tcp_receive_loop (unix_sock=39) at core/tcp_read.c:1978 __func__ = "tcp_receive_loop" #12 0x00005613ccb8561b in tcp_init_children (woneinit=0x7fff9b8ec338) at core/tcp_main.c:5139 r = 2 i = 7 reader_fd_1 = 39 pid = 0 si_desc = "tcp receiver (generic)\000\000\205\220\274\314\023V\000\000\020Î\233\377\177\000\000\301\200\313\314\000\000\000\000AS\000\000\000\000\000\000۸\352\314\023V\000\000\246\000\216\233\377\177\000\000\034\352\340\314\023V\000\000\340\302\216\233\001\000\000\000\360\311\017Q\322\177\000\000\340\302\216\233\377\177\000\000\340\227:S\001\000\000\000\000Î\233\377\177\000\000\274\353\340\314\023V\000" si = 0x0 __func__ = "tcp_init_children" #13 0x00005613cca62bba in main_loop () at main.c:1857 i = 8 pid = 21313 si = 0x0 si_desc = "udp receiver child=7 sock=XXX.XXX.XXX.XXX:5060\000\000\000\003\000\000\000 \000\000\000\000\230g\240\210\347A\357\000\000\000\000\000\000\000\000\210P\347\314\023V\000\000\000\000\000\000\000\000\000\000\260,\227}\322\177\000\000\263i\354\314\023V\000\000\000\000\000 \000\000\000\000\260Ď\233\377\177\000\000\t\205\317\314\023V\000" nrprocs = 8 woneinit = 1 __func__ = "main_loop" #14 0x00005613cca6d4aa in main (argc=17, argv=0x7fff9b8ec9f8) at main.c:3053 cfg_stream = 0x5613cecb6290 c = -1 r = 0 tmp = 0x7fff9b8ecf3f "" tmp_len = 2109797928 port = 32722 proto = 2109962736 ahost = 0x0 aport = 0 options = 0x5613cce77ce0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 3769828628 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x0 p = 0x0 st = {st_dev = 21, st_ino = 17087, st_nlink = 2, st_mode = 16832, st_uid = 109, st_gid = 115, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1620845793, tv_nsec = 887985330}, st_mtim = {tv_sec = 1625090397, tv_nsec = 157181302}, st_ctim = {tv_sec = 1625090401, tv_nsec = 197141517}, __glibc_reserved = {0, 0, 0}} tbuf = "\000\000\000\000\000\000\000\000\b\000\000\000\000\000\000\000\060\207\303}\322\177\000\000\000\000\000\000\000\000\000\000\240q\237\233\377\177\000\000\364\212\301}\322\177\000\000\b\000\000\000\000\000\000\000\230r\237\233\377\177\000\000\060\207\303}\322\177\000\000\230ǎ\233\377\177\000\000\224ǎ\233\377\177", '\000' <repeats 18 times>, "\230r\237\233\377\177\000\000\250q\237\233\377\177\000\000'\376\237}\322\177\000\000&\260be\000\000\000\000\000Ɏ\233\377\177\000\000\300\212\225\001\000\000\000\000`Ȏ\233\377\177\000\000PȎ\233\377\177\000\000\230ǎ\233\377\177\000\000X\207\303}\322\177\000\000\001", '\000' <repeats 31 times>... option_index = 0 long_options = {{name = 0x5613cce79f3f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x5613cce754e0 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x5613cce79f44 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x5613cce79f4a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x5613cce79f50 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x5613cce79f59 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x5613cce79f63 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x5613cce79f6d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x5613cce79f78 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x5613cce79f81 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x5613cce79f8c "debug", has_arg = 1, flag = 0x0, val = 1032}, {name = 0x5613cce79f92 "cfg-print", has_arg = 0, flag = 0x0, val = 1033}, {name = 0x5613cce79f9c "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __func__ = "main"

3 9

UAC_AUTH with RTPEngine NAT
by Paul Smith 02 Jul '21

02 Jul '21

Hi, [Note I posted this yesterday from the wrong email address.] I am struggling to see why my SDP is not being set correctly on the INVITE to my supplier with Proxy-Authentication when I use uac_auth(). The initial INVITE to my supplier has correct SDP set by rtpengine_manage. Then supplier replies with a 407. My failure route correctly handles the Auth, and also calls NATMANAGE again... but this time the SDP is unchanged and the private IP and original media information from the original device is relayed to my supplier. kamailio.cfg isbased on the default config. Running kamailio 5.4.6 and using example from uac module for uac_auth. My failure route calls NATMANAGE. Is there anything special about uac_auth? Do I need some extra magic to apply the message body changes after I have run rtpengine_manage(). I can see that the NATMANAGEr test for nat_uac_test("8") is true, and rtpengine_manage() s being called. But the outgoing SDP is not changed. Thanks for any hints! Paul ==================== Extract of kamailio.cfg failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } route(NATMANAGE); xlog("L_INFO","In failure route, just finshed NATMANAGE and now body is $mb"); if(t_check_status("401|407")) { # $avp(auser) = "test"; # $avp(apass) = "test"; # $avp(apass) = "36d0a02793542b4961e8348347236dbf"; if (uac_auth()) { t_relay(); } exit; } }

3 3

UAC_AUTH with RTPEngine NAT
by Paul Smith 01 Jul '21

01 Jul '21

Hi, I am struggling to see why my SDP is not being set correctly on the INVITE to my supplier with Proxy-Authentication when I use uac_auth(). The initial INVITE to my supplier has correct SDP set by rtpengine_manage. Then supplier replies with a 407. My failure route correctly handles the Auth, and also calls NATMANAGE again... but this time the SDP is unchanged and the private IP and original media information from the original device is relayed to my supplier. kamailio.cfg isbased on the default config. Running kamailio 5.4.6 and using example from uac module for uac_auth. My failure route calls NATMANAGE. Is there anything special about uac_auth? Do I need some extra magic to apply the message body changes after I have run rtpengine_manage(). ==================== Extract of kamailio.cfg failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } route(NATMANAGE); xlog("L_INFO","In failure route, just finshed NATMANAGE and now body is $mb"); if(t_check_status("401|407")) { # $avp(auser) = "test"; # $avp(apass) = "test"; # $avp(apass) = "36d0a02793542b4961e8348347236dbf"; if (uac_auth()) { t_relay(); } exit; } }

1 0

Kamailio BSD license
by stevecarellccie＠yahoo.com 01 Jul '21

01 Jul '21

Hello, I am trying to use Kamailio in our application which would be deployed in certain environment in addition to a cloud infrastructure. As I understand it, parts of the Kamailio code are covered under the BSD license - are you able to share what those are? I am trying to determine if Kamailio can be used in a customer environment or should it be strictly restricted to cloud environments due to the GPL v2 licenses Thanks,Steve

2 1

install stirshaken module by apt at Debian 10
by Yuriy Nasida 30 Jun '21

30 Jun '21

Hi, I know that it's possible to compile manually but what about apt method to install stirshaken module? I am checking the list of modules using: apt search kamailio But... Nothing looks like stirshaken. Thus, only several modules available and for others we should compile manually ? Please advise Thanks

2 1

Websocket In-dialoge SIP routing failed post network loss due to aliases
by Shahid Hussain 30 Jun '21

30 Jun '21

Hi, Websocket module documentation has a code reference to use aliases for SIP routing. However, aliases will not work in the following setup and situation. 1. Kamailio is configured with active and standby node 2. Ping is implemented from webclient and kamailio responds with pong. 3. Two clients ClinetA and ClientB registered themselves to Kamailio. 4. After SIP negotiation (INVITE-200OK) each client learnt about below aliases. Alias of ClientA: alias=172.27.6.98~58336 Alias of ClientB: alias=172.27.6.98~58342 So normally if ClientB wants to send SIP message to ClientA, SIP URI from ClientB looks like below ACK sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws SIP/2.0 4. Call is in a connected state. Following is the issue. i. Switchover (or network lost or reboot) at Kamailio happened ii. Due to ping pong both the clients detected network loss individually and re-registered themselves. iii. Aliases of both the clients got changed. New Alias of ClientA: alias=172.27.6.98~ 58346 New Alias of ClientB: alias=172.27.6.98~ 58348 iv. But ClientB doesn’t know that ClientA also re-registered and ClientA’s alias got changed and vice-versa v. Because of this ClientB still sends BYE with Initial alias BYE sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws SIP/2.0 Would like to know what is the recommended solution for this problem using alias or is it a limitation of using alias? Thanks & Regards, Shahid

4 11

Re: [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy
by Peter Manley 29 Jun '21

29 Jun '21

Thanks Muhammad, That seems to have done the trick. I am now able to bridge WebRTC to UDP. Regards, Peter Manley M S shaheryarkh at gmail.com<mailto:sr-users%40lists.kamailio.org?Subject=Re%3A%20%5BSR-Users%5D%20Guidance%20building%20a%20WebRTC%20to%20Asterisk%20Intermediate%0A%20Proxy&In-Reply-To=%3CCAFZQphyZE_tGB9%3D5tFQBtXAv5twMh%3D9aCNFAvveGf0-L%3Dy_qwQ%40mail.gmail.com%3E> Sat Jun 26 03:57:27 CEST 2021 * Previous message (by thread): [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy<https://lists.kamailio.org/pipermail/sr-users/2021-June/112799.html> * Messages sorted by: [ date ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/date.html#112800> [ thread ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/thread.html#112800> [ subject ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/subject.html#112800> [ author ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/author.html#112800> ________________________________ The problem seems to that the kamailio is trying to route it over UDP port using WSS transport, which will never work. You are creating a transport bridge (UDP <-> WSS i.e. stateless transport <-> stateful transport) as well media bridge (RTP <-> SRTP), so it is going to be tricky. I strongly discourage it since things get messy for sequential requests. But you can try the following, Here are some PATH module parameters and functions, you will need to setup. 1. You will need to enable "use_received" parameter to ensure connection tracking. 2. It seems you may have enabled "received_format", so you make sure you do NAT detection and use "handle_ruri_alias" for initial requests. 3. You must also enabled "enable_r2", but with caution, since both ends (i.e. UAC and UAS) can be using same kamailio (proxy) socket for in/out. 4. Make sure to use "add_path_received" instead of "add_path". 5. If using OUTBOUND module then make sure it is loaded BEFORE the PATH module. Thank you. -- Muhammad Shahzad Shafi Burraq Technologies Tel: +49 176 99 83 10 85 On Sat, Jun 26, 2021 at 12:47 AM Peter Manley <peter.manley at kombea.com<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> wrote: > Hello, > > > > I'm working on building an RTP Proxy which will allow both regular SIP to > SIP transactions and WebRTC to SIP transactions to an Asterisk Server. > > > > SIP UAC < ---UDP/SIP --- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > > Asterisk Server > > > > AND > > > > WebRTC UAC < ---WSS--- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > > Asterisk Server > > > > The Kamailio/RTPEngine Proxy is stateless (does not use the tm or > registerer modules), and uses the Path module to replace the Route header > with a Path header. > > > > It works grep for SIP/UDP to SIP/UDP transactions. > > > > HOWEVER... It only partially works for WSS to SIP/UDP transactions. The > REGISTER/401(Unauthorized), REGISTER/200(OK) transaction works well. > > > > When the Asterisk server sends a SIP Options message, I get the following > error: > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: INFO: <script>: => OPTIONS < > sip:l4umfumh at 192.0.2.97<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>;transport=ws;alias=209.141.117.212~59751~6 -> > <null> (IP:172.16.0.48:sp) > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> > [core/msg_translator.c:2815]: via_builder(): TCP/TLS connection (id: 0) for > WebSocket could not be found > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> > [core/msg_translator.c:2003]: build_req_buf_from_sip_req(): could not > create Via header > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> > [core/forward.c:550]: forward_request(): building failed > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> > [core/receive.c:425]: receive_msg(): error while trying script > > > > It appears that the WebSocket corresponding to the WSS leg cannot be > located when the request is initiated from the Asterisk Server. > > > > Any thoughts on how I can get this to work in the other direction > (Asterisk to Kamailio)? I've been looking at the WebRTC to SIP Client and > Server (https://github.com/havfo/WEBRTC-to-SIP), but that doesn't quite > do what I want it to do (i.e. it appears to act as a registrar server and > doesn't forward REGISTER requests on to another registrar). > > > > > > Thanks and regards, > > > > *Peter H. Manley, Ph.D.* > > Chief Software Architect > > KomBea Corporation > > 3400 North Ashton Blvd, Suite 470 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > Lehi, Utah 84043 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > www.kombea.com<http://www.kombea.com> > <https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> > > > > Office:+1.801.610.5808 > > Mobile +1.801.243.7773 > > Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

1 0

Kamailio + RTPEngine - Destinations Behind NAT
by Edward Romanenco 29 Jun '21

29 Jun '21

Hi everyone, I'm using Kamailio (Ver 5.5.1, Public IP X.X.X.11, Private IP 172.18.0.20) + RTPEngine (Ver 9.4.1.1, Public IP X.X.X.7, Private IP 172.18.0.50) alongside Freeswitch (Private IP 172.18.0.40) in my backend and delivering calls to a mobile application (call flow is Freeswitch -> Kamailio + RTPEngine -> Applicaiton). While calls made to NATted destination generally works fine - from time to time, we've experiencing one-sided audio due to the rtpengine sending the media stream to the remote end's private IP instead of his public one. For instance, in the attached logs - media was sent towards 192.168.1.93 instead of 79.179.1.2, which is the expected destination. Problem appears almost randomly; I can make 20 fine calls in a row for the same destination until experiencing it, and I didn't find any difference as it comes to the SIP invite exchange between the two. Has anyone ever experienced the same behavior and can share some general instructions on how to troubleshoot it? Relevant Kamailio config section is attached. Invite Kamailio->Device INVITE sip:972587102881@79.179.1.2:56308;transport=TLS;ob SIP/2.0 Record-Route: <sip:X.X.X.11:443;transport=tls;r2=on;lr=on;ftag=cvcHy5UcS9pjj> Record-Route: <sip:172.18.0.20:5099;r2=on;lr=on;ftag=cvcHy5UcS9pjj> Via: SIP/2.0/TLS X.X.X.11:443;branch=z9hG4bK24bf.52b1dd68e26010ba103bd8d6fdb53cf4.0 Via: SIP/2.0/UDP 172.18.0.40;received=172.18.0.40;rport=5060;branch=z9hG4bKH234Q58HpHNKK Max-Forwards: 66 From: "972522149596" <sip:FreeSWITCH@172.18.0.20:5099>;tag=cvcHy5UcS9pjj To: <sip:972587102881@172.18.0.20:5099> Call-ID: be55bd1e-537d-123a-65a9-0242ac120028 CSeq: 37932220 INVITE Contact: <sip:gw+kam-in@172.18.0.40:5060;transport=udp;gw=kam-in;alias=172.18.0.40~5060~1> User-Agent: FreeSWITCH-mod_sofia/1.10.6-release+git~20210325T131609Z~1ff9d0a60e~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 567 X-SESSION-ID: 721533765 X-INCOMING: PSTN X-from-freeswitch: +972522149596 X-FS-Support: update_display,send_info Remote-Party-ID: "972522149596" <sip:972522149596@172.18.0.20:5099>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1624937621 1624937622 IN IP4 X.X.X.7 s=FreeSWITCH c=IN IP4 X.X.X.7 t=0 0 m=audio 30018 RTP/AVP 102 101 a=rtpmap:102 opus/48000/2 a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 a=rtpmap:101 telephone-event/48000 a=fmtp:101 0-16 a=sendrecv a=rtcp:30019 a=ptime:20 a=ice-ufrag:bKPZt0dW a=ice-pwd:H4qyL7mPUOvoiAjosiLasXAdzj a=candidate:41sXGEsTafxHi8ms 1 UDP 2130706431 X.X.X.7 30018 typ host a=candidate:41sXGEsTafxHi8ms 2 UDP 2130706430 X.X.X.7 30019 typ host Response Device->Kamailio SIP/2.0 200 OK Via: SIP/2.0/TLS X.X.X.11:443;received=X.X.X.11;branch=z9hG4bK24bf.52b1dd68e26010ba103bd8d6fdb53cf4.0 Via: SIP/2.0/UDP 172.18.0.40;rport=5060;received=172.18.0.40;branch=z9hG4bKH234Q58HpHNKK Record-Route: <sip:X.X.X.11:443;transport=tls;lr;r2=on;ftag=cvcHy5UcS9pjj> Record-Route: <sip:172.18.0.20:5099;lr;r2=on;ftag=cvcHy5UcS9pjj> Call-ID: be55bd1e-537d-123a-65a9-0242ac120028 From: "972522149596" <sip:FreeSWITCH@172.18.0.20>;tag=cvcHy5UcS9pjj To: <sip:972587102881@172.18.0.20>;tag=5f75877e-58c2-44f3-a5b3-43f8786c42a1 CSeq: 37932220 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:972587102881@79.179.1.2:56308;transport=TLS;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 348 v=0 o=- 3833960833 3833960834 IN IP4 192.168.1.93 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 96 120 c=IN IP4 192.168.1.93 b=TIAS:96000 a=rtcp:4003 IN IP4 192.168.1.93 a=sendrecv a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:120 telephone-event/48000 a=fmtp:120 0-16 a=ssrc:816694098 cname:72cdb5f53fc2e8fb Kamailio NAT Detect/Manage Routes # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT xinfo("NATDETECT: Adding the rport parameter to the first Via header of the received message\n"); force_rport(); xinfo("NATDETECT: Trying to guess if client's request originated behind a nat. Testing 19.\n"); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xinfo("NATDETECT: REGISTER request. Creating a URI consisting of the source IP, port, and protocol and storing the URI in an Attribute-Value-Pair.\n"); fix_nated_register(); } else { if(is_first_hop()) { xinfo("NATDETECT: Adding an ;alias=ip~port~transport parameter to the contact URI containing the received ip, port, and transport protocol.\n"); set_contact_alias(); } } xinfo("NATDETECT: Setting flag FLT_NATS.\n"); setflag(FLT_NATS); } else { xinfo("NATDETECT: Didn't detect natted request by any of the given heuristics\n"); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xinfo("NATMANAGE: This is a request with to-tag that has 'nat=yes', setting branch flag FLB_NATB.\n"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { xinfo("NATMANAGE: No NAT flag detected for this request\n"); return; } else { xinfo("NATMANAGE: NAT flag detected for this request\n"); } if(!is_present_hf("x-purpose")) { xinfo("NATMANAGE: x-purpose header does not appear in this request, SDP is searched for occurrence of RFC1918 or RFC6598 addresses.\n"); if(nat_uac_test("8")) { xinfo("NATMANAGE: Found occurrence of RFC1918 or RFC6598 addresses. Checking if there is a match of source address or uri with an address in the group of the dispatcher-list.\n"); if(ds_is_from_list()) { xinfo("NATMANAGE: There is a match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin direction=priv direction=pub"); } else { xinfo("NATMANAGE: No match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin direction=pub direction=priv"); } } else { xinfo("NATMANAGE: No occurrence of RFC1918 or RFC6598 addresses. Checking if there is a match of source address or uri with an address in the group of the dispatcher-list.\n"); if(ds_is_from_list()) { xinfo("NATMANAGE: There is a match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin trust-address direction=priv direction=pub"); } else { xinfo("NATMANAGE: No match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin trust-address direction=pub direction=priv"); } } } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xinfo("NATMANAGE: This is a request without to-tag and top-executed route block is branch_route, adding record route param ;nat=yes.\n"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xinfo("NATMANAGE: Adding an ;alias=ip~port~transport parameter to the contact URI containing the received ip, port, and transport protocol.\n"); set_contact_alias(); } } } #!endif return; } RTPEngine Logs 2021-06-29T13:07:05.440658779Z [1624972025.439547] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Received command 'offer' from 172.18.0.20:53880 2021-06-29T13:07:05.440690853Z [1624972025.439641] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Dump for 'offer' from 172.18.0.20:53880: { "supports": [ "load limit" ], "sdp": "v=0 2021-06-29T13:07:05.440695330Z o=FreeSWITCH 1624937621 1624937622 IN IP4 172.18.0.40 2021-06-29T13:07:05.440698353Z s=FreeSWITCH 2021-06-29T13:07:05.440700926Z c=IN IP4 172.18.0.40 2021-06-29T13:07:05.440703539Z t=0 0 2021-06-29T13:07:05.440706179Z m=audio 34404 RTP/AVP 102 101 2021-06-29T13:07:05.440708722Z a=rtpmap:102 opus/48000/2 2021-06-29T13:07:05.440711294Z a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 2021-06-29T13:07:05.440714053Z a=rtpmap:101 telephone-event/48000 2021-06-29T13:07:05.440716746Z a=fmtp:101 0-16 2021-06-29T13:07:05.440719258Z a=ptime:20 2021-06-29T13:07:05.440721823Z ", "ICE": "force", "direction": [ "priv", "pub" ], "replace": [ "session-connection", "origin" ], "call-id": "be55bd1e-537d-123a-65a9-0242ac120028", "received-from": [ "IP4", "172.18.0.40" ], "from-tag": "cvcHy5UcS9pjj", "command": "offer" } 2021-06-29T13:07:05.441069790Z [1624972025.439694] NOTICE: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] Creating new call 2021-06-29T13:07:05.441088698Z [1624972025.439809] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Default sink codec is opus/48000/2 2021-06-29T13:07:05.441091829Z [1624972025.439815] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Output DTMF payload type is 101 2021-06-29T13:07:05.441094966Z [1624972025.439821] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for opus/48000/2 2021-06-29T13:07:05.441097777Z [1624972025.439828] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec opus/48000/2 2021-06-29T13:07:05.441100540Z [1624972025.439833] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for opus/48000/2 2021-06-29T13:07:05.441103390Z [1624972025.439839] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for telephone-event/48000 2021-06-29T13:07:05.441106153Z [1624972025.439844] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec telephone-event/48000 2021-06-29T13:07:05.441108913Z [1624972025.439849] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for telephone-event/48000 2021-06-29T13:07:05.441111679Z [1624972025.440116] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 172.18.0.40:34404 2021-06-29T13:07:05.441124990Z [1624972025.440124] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 172.18.0.40:34405 2021-06-29T13:07:05.441663481Z [1624972025.441074] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Replying to 'offer' from 172.18.0.20:53880 (elapsed time 0.001414 sec) 2021-06-29T13:07:05.441694377Z [1624972025.441103] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Response dump for 'offer' to 172.18.0.20:53880: { "sdp": "v=0 2021-06-29T13:07:05.441701068Z o=FreeSWITCH 1624937621 1624937622 IN IP4 X.X.X.7 2021-06-29T13:07:05.441706421Z s=FreeSWITCH 2021-06-29T13:07:05.441711414Z c=IN IP4 X.X.X.7 2021-06-29T13:07:05.441716274Z t=0 0 2021-06-29T13:07:05.441721177Z m=audio 30018 RTP/AVP 102 101 2021-06-29T13:07:05.441726204Z a=rtpmap:102 opus/48000/2 2021-06-29T13:07:05.441731246Z a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 2021-06-29T13:07:05.441736873Z a=rtpmap:101 telephone-event/48000 2021-06-29T13:07:05.441742017Z a=fmtp:101 0-16 2021-06-29T13:07:05.441827012Z a=sendrecv 2021-06-29T13:07:05.441844744Z a=rtcp:30019 2021-06-29T13:07:05.441848881Z a=ptime:20 2021-06-29T13:07:05.441852444Z a=ice-ufrag:bKPZt0dW 2021-06-29T13:07:05.441856394Z a=ice-pwd:H4qyL7mPUOvoiAjosiLasXAdzj 2021-06-29T13:07:05.441860642Z a=candidate:41sXGEsTafxHi8ms 1 UDP 2130706431 X.X.X.7 30018 typ host 2021-06-29T13:07:05.441864955Z a=candidate:41sXGEsTafxHi8ms 2 UDP 2130706430 X.X.X.7 30019 typ host 2021-06-29T13:07:05.441868967Z ", "result": "ok" } 2021-06-29T13:07:07.001082810Z [1624972027.000176] DEBUG: [core] timer run time = 0.000029 sec 2021-06-29T13:07:07.817899628Z [1624972027.816949] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] >>> in_ssrc changed for: :0 new: 30adc352 2021-06-29T13:07:07.817932021Z [1624972027.816987] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] >>> out_ssrc changed for 172.18.0.40:34405 new: 30adc352 2021-06-29T13:07:07.817936388Z [1624972027.816998] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 201 2021-06-29T13:07:07.817939853Z [1624972027.817025] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] RR from 30adc352 about 0: FL 0 TL 0 HSR 0 J 0 LSR 0 DLSR 0 2021-06-29T13:07:07.817942689Z [1624972027.817030] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] No payload type known for RTCP RR, discarding 2021-06-29T13:07:07.817945696Z [1624972027.817036] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 202 2021-06-29T13:07:07.817948500Z [1624972027.817124] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] Forward to sink endpoint: 172.18.0.40:34405 2021-06-29T13:07:07.818762433Z [1624972027.817995] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 201 2021-06-29T13:07:07.818790347Z [1624972027.818031] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] RR from 30adc352 about 0: FL 0 TL 0 HSR 0 J 0 LSR 0 DLSR 0 2021-06-29T13:07:07.818794422Z [1624972027.818035] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] No payload type known for RTCP RR, discarding 2021-06-29T13:07:07.818797834Z [1624972027.818042] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 202 2021-06-29T13:07:07.818801095Z [1624972027.818122] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] Forward to sink endpoint: 172.18.0.40:34405 2021-06-29T13:07:07.847992398Z [1624972027.846284] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Received command 'answer' from 172.18.0.20:47817 2021-06-29T13:07:07.848048472Z [1624972027.846325] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Dump for 'answer' from 172.18.0.20:47817: { "supports": [ "load limit" ], "sdp": "v=0 2021-06-29T13:07:07.848058346Z o=- 3833960833 3833960834 IN IP4 192.168.1.93 2021-06-29T13:07:07.848063378Z s=pjmedia 2021-06-29T13:07:07.848067785Z b=AS:117 2021-06-29T13:07:07.848071736Z t=0 0 2021-06-29T13:07:07.848075782Z a=X-nat:0 2021-06-29T13:07:07.848079588Z m=audio 4002 RTP/AVP 96 120 2021-06-29T13:07:07.848084210Z c=IN IP4 192.168.1.93 2021-06-29T13:07:07.848089110Z b=TIAS:96000 2021-06-29T13:07:07.848092756Z a=rtcp:4003 IN IP4 192.168.1.93 2021-06-29T13:07:07.848096578Z a=sendrecv 2021-06-29T13:07:07.848100173Z a=rtpmap:96 opus/48000/2 2021-06-29T13:07:07.848104153Z a=fmtp:96 useinbandfec=1 2021-06-29T13:07:07.848108660Z a=rtpmap:120 telephone-event/48000 2021-06-29T13:07:07.848112776Z a=fmtp:120 0-16 2021-06-29T13:07:07.848116856Z a=ssrc:816694098 cname:72cdb5f53fc2e8fb 2021-06-29T13:07:07.848120776Z ", "ICE": "force", "direction": [ "pub", "priv" ], "replace": [ "session-connection", "origin" ], "call-id": "be55bd1e-537d-123a-65a9-0242ac120028", "received-from": [ "IP4", "79.179.1.2" ], "from-tag": "cvcHy5UcS9pjj", "to-tag": "5f75877e-58c2-44f3-a5b3-43f8786c42a1", "command": "answer" } 2021-06-29T13:07:07.848128079Z [1624972027.846411] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Default sink codec is opus/48000/2 2021-06-29T13:07:07.848132255Z [1624972027.846418] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Output DTMF payload type is 120 2021-06-29T13:07:07.848136164Z [1624972027.846424] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for opus/48000/2 2021-06-29T13:07:07.848139819Z [1624972027.846429] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec opus/48000/2 2021-06-29T13:07:07.848143930Z [1624972027.846434] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for opus/48000/2 2021-06-29T13:07:07.848169208Z [1624972027.846441] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for telephone-event/48000 2021-06-29T13:07:07.848174300Z [1624972027.846445] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec telephone-event/48000 2021-06-29T13:07:07.848178752Z [1624972027.846449] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for telephone-event/48000 2021-06-29T13:07:07.848183460Z [1624972027.846586] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 192.168.1.93:4002 2021-06-29T13:07:07.848188805Z [1624972027.846593] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 192.168.1.93:4003 2021-06-29T13:07:07.848193111Z [1624972027.846597] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [ice] Shutting down ICE agent (nothing to do) 2021-06-29T13:07:07.848197042Z [1624972027.847228] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Replying to 'answer' from 172.18.0.20:47817 (elapsed time 0.000890 sec) 2021-06-29T13:07:07.848201623Z [1624972027.847243] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Response dump for 'answer' to 172.18.0.20:47817: { "sdp": "v=0 2021-06-29T13:07:07.848205754Z o=- 3833960833 3833960834 IN IP4 172.18.0.50 2021-06-29T13:07:07.848209696Z s=pjmedia 2021-06-29T13:07:07.848213446Z b=AS:117 2021-06-29T13:07:07.848217361Z t=0 0 2021-06-29T13:07:07.848220816Z a=X-nat:0 2021-06-29T13:07:07.848224363Z m=audio 30018 RTP/AVP 96 120 2021-06-29T13:07:07.848228432Z c=IN IP4 172.18.0.50 2021-06-29T13:07:07.848232603Z b=TIAS:96000 2021-06-29T13:07:07.848235992Z a=ssrc:816694098 cname:72cdb5f53fc2e8fb 2021-06-29T13:07:07.848240381Z a=rtpmap:96 opus/48000/2 2021-06-29T13:07:07.848243923Z a=fmtp:96 useinbandfec=1 2021-06-29T13:07:07.848247725Z a=rtpmap:120 telephone-event/48000 2021-06-29T13:07:07.848386453Z a=fmtp:120 0-16 2021-06-29T13:07:07.848395759Z a=sendrecv 2021-06-29T13:07:07.848399822Z a=rtcp:30019 2021-06-29T13:07:07.848403748Z a=ptime:20 2021-06-29T13:07:07.848407203Z a=ice-ufrag:4Q9H38CE 2021-06-29T13:07:07.848410827Z a=ice-pwd:OR8EfUn5dCLbhZqyiwsTUOyUdW 2021-06-29T13:07:07.848414680Z a=candidate:f75EKiqnFiVfQdoZ 1 UDP 2130706431 172.18.0.50 30018 typ host 2021-06-29T13:07:07.848419247Z a=candidate:f75EKiqnFiVfQdoZ 2 UDP 2130706430 172.18.0.50 30019 typ host 2021-06-29T13:07:07.848432370Z ", "result": "ok" } 2021-06-29T13:07:07.872251466Z [1624972027.871342] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] Created candidate pair f75EKiqnFiVfQdoZ:33d9f4832b3cc9a0:1 between 172.18.0.50 and 172.18.0.40:34404, type prflx 2021-06-29T13:07:07.872276197Z [1624972027.871381] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] Triggering check for f75EKiqnFiVfQdoZ:33d9f4832b3cc9a0:1 2021-06-29T13:07:07.872286600Z [1624972027.871396] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] ICE pair f75EKiqnFiVfQdoZ:33d9f4832b3cc9a0:1 has been nominated by peer 2021-06-29T13:07:07.872289762Z [1624972027.871401] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] ICE not completed yet and no usable candidates 2021-06-29T13:07:07.872292668Z [1624972027.871407] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Successful STUN binding request from 172.18.0.40:34404 2021-06-29T13:07:07.919668940Z [1624972027.918862] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] >>> in_ssrc changed for: 172.18.0.40:34404 new: 70f328c1 2021-06-29T13:07:07.919698433Z [1624972027.918897] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] >>> out_ssrc changed for 192.168.1.93:4002 new: 70f328c1 2021-06-29T13:07:07.919703321Z [1624972027.918911] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64423 TS 2880) 2021-06-29T13:07:07.939554198Z [1624972027.938733] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64424 TS 3840) 2021-06-29T13:07:07.959769679Z [1624972027.959017] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64425 TS 4800) 2021-06-29T13:07:07.979861917Z [1624972027.979067] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64426 TS 5760) 2021-06-29T13:07:07.999566603Z [1624972027.998777] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64427 TS 6720) 2021-06-29T13:07:08.019668998Z [1624972028.018802] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64428 TS 7680) 2021-06-29T13:07:08.040856248Z [1624972028.039663] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64429 TS 8640) 2021-06-29T13:07:08.061133612Z [1624972028.059847] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64430 TS 9600) Edward

1 1

Guidance building a WebRTC to Asterisk Intermediate Proxy
by Peter Manley 26 Jun '21

26 Jun '21

Hello, I'm working on building an RTP Proxy which will allow both regular SIP to SIP transactions and WebRTC to SIP transactions to an Asterisk Server. SIP UAC < ---UDP/SIP --- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > Asterisk Server AND WebRTC UAC < ---WSS--- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > Asterisk Server The Kamailio/RTPEngine Proxy is stateless (does not use the tm or registerer modules), and uses the Path module to replace the Route header with a Path header. It works grep for SIP/UDP to SIP/UDP transactions. HOWEVER... It only partially works for WSS to SIP/UDP transactions. The REGISTER/401(Unauthorized), REGISTER/200(OK) transaction works well. When the Asterisk server sends a SIP Options message, I get the following error: Jun 25 15:47:03 VRTPENGINE kamailio[4953]: INFO: <script>: => OPTIONS < sip:l4umfumh@192.0.2.97;transport=ws;alias=209.141.117.212~59751~6 -> <null> (IP:172.16.0.48:sp) Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> [core/msg_translator.c:2815]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> [core/msg_translator.c:2003]: build_req_buf_from_sip_req(): could not create Via header Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> [core/forward.c:550]: forward_request(): building failed Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> [core/receive.c:425]: receive_msg(): error while trying script It appears that the WebSocket corresponding to the WSS leg cannot be located when the request is initiated from the Asterisk Server. Any thoughts on how I can get this to work in the other direction (Asterisk to Kamailio)? I've been looking at the WebRTC to SIP Client and Server (https://github.com/havfo/WEBRTC-to-SIP), but that doesn't quite do what I want it to do (i.e. it appears to act as a registrar server and doesn't forward REGISTER requests on to another registrar). Thanks and regards, Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

2 1

using both ldap and local authenticate
by Thành Trần Duy 25 Jun '21

25 Jun '21

Hi all, I am using kamailio as PBX with some softphones and hard phones. I want user on a pc to login with LDAP and users using a hard phone login normally (username/password). Can I use 2 types of authentication at the same time? Many thanks, Tran Duy Thanh

2 2

Possible memory leak on 5.5.x (new)?
by George Diamantopoulos 25 Jun '21

25 Jun '21

Hello all, I'm still investigating the (most likely non-kamailio-related) memory leak of my previous message to the list, there have been no developments so far. I'll update if anything changes. This concerns a new finding which seems to affect kamailio 5.5.x. I have two kamailio instances receiving the same traffic via round-robin. I upgraded only one of them to 5.5.1 and left the other to 5.4.6 as I feared of any issues arising. I was lucky to do so, because with identical configuration, 5.5.x seems to run out of SHM very quickly. Here are links to graphs produced by our monitoring system: Old kamailio (no memory leak): https://pasteboard.co/K8fVBiD.png New kamailio (possible leak): https://pasteboard.co/K8fVS9N.png The configuration uses mtree, htable, vars and vns extensively. Has anyone come across anything similar? Let me know if I can provide any further information to help disect this. Thanks! BR, George

2 1

rr module: record_route_advertised_address()/record_route_preset() do ignore custom user AVP
by Valentin Christoph 25 Jun '21

25 Jun '21

Hi, Maybe a discussion for the DEV mailing list???? Want to overwrite the FQDN by a NATed IP address in the topmost Record-Route of an outgoing INVITE request. Cannot set the advertise parameter (this must tell the FQDN for all other situations), only in some special situations I want to overwrite the FQDN in Record-Route. My solution would be to implement in the Kamailio source code (rr module) that not only record_route() uses the get_custom_user() function, but the other both functions uses it, too. Any better proposals? All the best Christoph

2 3

Kamailio $Rp variable showing wrong port
by Shahid Hussain 25 Jun '21

25 Jun '21

Hi, I have configured the dispatcher default socket as below IP , the expectation here is that kamailio should send all traffic from this IP (port will be ephemeral) over TCP. modparam("dispatcher", "ds_default_socket", "tcp:172.27.45.219:5060") This system also listens on port 8080 which is meant for websocket communication (dispatcher communication is over non websocket). Kamailio is sending an OPTION message to dispatcher IP over TCP on 5060 and getting 200 OK response but Kamailio considering the response is on websocket port 8080 ,$Rp value is 8080. onreply_route { if (($Rp == 8080 ) && !(proto == WS)) { xlog("L_WARN", "SIP response received on $Rp protocol:proto RIP:$Ri SIP:$si method:$rm cid:$ci replystatus:$rs len:$ml\n"); xlog("L_WARN", "SIP message received: $mb\n"); drop; } ..... .... } Following is the output of above logs: Jun 24 15:10:08 localhost /usr/sbin/kamailio[6223]: WARNING: {2 10 OPTIONS 1ab6573a4640823b-6209(a)172.27.45.198} <script>: SIP response received on 8080 protocol:tcp RIP:172.27.45.219 SIP:172.27.44.125 method:OPTIONS cid:1ab6573a4640823b-6209@172.27.45.198 replystatus:200 len:416 Jun 24 15:10:08 localhost /usr/sbin/kamailio[6223]: WARNING: {2 10 OPTIONS 1ab6573a4640823b-6209(a)172.27.45.198} <script>: SIP message received: SIP/2.0 200 OK#015#012Via: SIP/2.0/TCP 172.27.45.198:5060;branch=z9hG4bK4526.92b79f34000000000000000000000000.0;received=172.27.45.219#015#012From: <sip:dispatcher@localhost>;tag=1ae6c95d92f9ea9ce927f715de5c48c8-eb141029#015#012To: <sip:172.27.44.125:5060;transport=tcp>;tag=a9d542e0#015#012Call-ID: 1ab6573a4640823b-6209(a)172.27.45.198#015#012CSeq: 10 OPTIONS#015#012Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, NOTIFY#015#012Content-Length: 0 Also I have attached a pcap. Dispatcher.list configuration: 1 sip:172.27.44.125:5060;transport=tcp 0 0 duid=sample-cas;maxload=1000 Thanks, Shahid

3 10

S-CSCF and I-CSCF
by Jayanth Acharya 24 Jun '21

24 Jun '21

Hi, Have found example configuration for S-CSCF, I-CSCF and P-CSCF in Kamailio repository, and a S-CSCF module for the registered user data, but no recent updates. Wondering if anyone can share insights into how widely it has been used, or interops done in say telco networks with it ? Does it handle third-party registrations, is it actually able to do a Cx interaction with HSS ? Finally, how have these components that make up the IMS-core from SIP signaling standpoint, been tested ? Are there any signaling only test harnesses ? cheers, Jay

2 1

Errors after upgrade to 5.5
by Daniel W. Graham 23 Jun '21

23 Jun '21

Receiving constant errors for all functions after upgrade to 5.5, any change that could have caused this? ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:968] took too long [1251 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:988] took too long [1309 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:1038] took too long [1321 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (5)] cfg [/usr/local/etc/kamailio/kamailio.cfg:452] took too long [1342 us] ERROR: <core> [db_query.c:68]: db_do_submit_query(): alert - query execution too long [1493 us] for [select `password`,`id` from `subscriber` where `username`='x'] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [www_authenticate (26)] cfg [/usr/local/etc/kamailio/kamailio.cfg:951] took too long [1667 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:968] took too long [1694 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:988] took too long [1817 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:1038] took too long [1835 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (5)] cfg [/usr/local/etc/kamailio/kamailio.cfg:452] took too long [1869 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [www_authenticate (26)] cfg [/usr/local/etc/kamailio/kamailio.cfg:951] took too long [1405 us] ……. -dan

2 4

AVPs in Switch/Case
by Joel Serrano 22 Jun '21

22 Jun '21

Hi guys, Quick question, are AVPs valid in "switch" as "case"'s? Example: $avp(var1) = "101"; $avp(var2) = "201"; switch($rU) { case "$avp(var1)": ... break; case "$avp(var2)": ... break; default: send_reply("403", "Forbidden"); exit; } For some reason I can't get the case to match, and I'm thinking it's not using the actual value of $avp(var1) and instead it's using literally "$avp(var1)" as a string. NOTE: I also tried without putting the AVPs in quotes, but Kamailio complains and doesn't start. Is this doable or do I need to use if..else if..else if..else for this? Thanks, Joel.

3 3

Web Sockets connection error: bad request, state=7, error=4
by Peter Manley 22 Jun '21

22 Jun '21

Hello, I'm using Kamailio 5.3.2 and am trying to establish a WebRTC connection for using SIP through Kamailio on port 8089. I am getting the following error in the Syslog: Jun 22 09:20:38 VRTPENGINE kamailio[43050]: ERROR: <core> [core/tcp_read.c:1535]: tcp_read_req(): bad request, state=7, error=4 buf:#012GET /ws HTTP/1.1#015#012Host: rtpengine.kombea.io:8089#015#012Connection: Upgrade#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/91.0.4472.106 Safari/537.36#015#012Upgrade: websocket#015#012Origin: https://vicidev.kombea.io#015#012Sec-WebSocket-Version: 13#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language: en-US,en;q=0.9#015#012Sec-WebSocket-Key: 1Pda50NxHxMenZaMW7cUkA==#015#012Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol: sip#015#012#015#012#012parsed:#012GET /ws HTTP/1.1#015#012Host: rtpengine.kombea.io:8089#015#012Connection: Upgrade#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/91.0.4472.106 Safari/537.36#015#012Upgrade: websocket#015#012Origin: https://vicidev.kombea.io#015#012Sec-WebSocket-Version: 13#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language: en-US,en;q=0.9#015#012Sec-WebSocket-Key: 1Pda50NxHxMenZaMW7cUkA==#015#012Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol: sip#015#012#015#012 Any idea on what would cause this? Thanks Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

2 2

execute callback every X minutes mid call
by Avremy Cohen 22 Jun '21

22 Jun '21

Hi I am trying to execute code every X min in active calls and I cant seem to find any way to do this. I basically have a function that checks if the user has balance and permission to carry on the call and the callback should trigger it every X mins. The call_control module doesnt work for me since A. It doesn't trigger a function it just ends the callback B. it uses an external application to calculate it any advice is appreciated Avi

3 3

Re: [SR-Users] Unable to establish TCP connection with Kamailio
by Peter Manley 17 Jun '21

17 Jun '21

Hello Ovidiu, Interestingly, that fixed the problem. I can now continue to make progress. Why does not forking and/or logging to stderr cause this to happen? Thank you! Regards, Peter Manley Ovidiu Sas osas at voipembedded.com<mailto:sr-users%40lists.kamailio.org?Subject=Re%3A%20%5BSR-Users%5D%20Unable%20to%20establish%20TCP%20connection%20with%20Kamailio&In-Reply-To=%3CCAND0LkvxyxE%3D5CPxZTe-Em43X6E5nep6z9n7a8f1zo3YbKksMw%40mail.gmail.com%3E> Thu Jun 17 23:05:14 CEST 2021 * Previous message (by thread): [SR-Users] Unable to establish TCP connection with Kamailio<https://lists.kamailio.org/pipermail/sr-users/2021-June/112759.html> * Messages sorted by: [ date ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/date.html#112760> [ thread ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/thread.html#112760> [ subject ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/subject.html#112760> [ author ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/author.html#112760> ________________________________ It seems that you have fork set to no, Let kamailio fork and collect logs via syslog (not via stderror). -ovidiu On Thu, Jun 17, 2021 at 16:26 Peter Manley <peter.manley at kombea.com<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> wrote: > Hello, > > > > I'm running Kamailio Version 5.3.2 on Ubuntu 20.04. Here my build > information: > > > > version: kamailio 5.3.2 (x86_64/linux) > > flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, > DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, > F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, > USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, > HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED > > ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, > BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > id: unknown > > compiled with gcc 9.2.1 > > > > I have the following configuration set in my kamalio.cfg file: > > > > ##!define WITH_MYSQL > > ##!define WITH_AUTH > > ##!define WITH_USRLOCDB > > ##!define WITH_TLS > > ##!define WITH_HOMER > > ##!define WITH_WEBSOCKETS > > ##!define WITH_ANTIFLOOD > > ##!define WITH_IPV6 > > ##!define WITH_BRIDGE_ON_FAIL > > ##!define WITH_LOCALHOST_WS > > ##!define WITH_LOCALHOST_SIP > > > > #!substdef "!MY_SIP_PORT!5060!g" > > #!substdef "!MY_SIPS_PORT!5061!g" > > #!substdef "!MY_WS_PORT!8088!g" > > #!substdef "!MY_WSS_PORT!8089!g" > > > > #!substdef "!MY_LOG_FACILITY!LOG_DAEMON!g" > > > > #!substdef "!MY_IP4_ADDR!172.16.0.57!g" > > #!substdef "!IP4_LOCALHOST!127.0.0.1!g" > > #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" > > #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" > > #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" > > #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" > > > > #!ifdef WITH_IPV6 > > #!substdef "!MY_IP6_ADDR![fe80::d6e4:2f5f:e5b7:7dff]!g" > > #!substdef "!IP6_LOCALHOST![::1]!g" > > #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" > > #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" > > #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" > > #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" > > #!endif > > > > #!substdef "!MY_DOMAIN!rtpengine.kombea.io!g" > > debug=10 // debug level (cmd line: -dddddddddd) > > children=4 > > #fork=yes > > force_rport=yes // Add recieved port to VIA > > #log_stderror=no // (cmd line: -E) > > > > // Uncomment these lines to enter debugging mode > > fork=no > > log_stderror=yes > > > > check_via=no // (cmd. line: -v) > > dns=no // (cmd. line: -r) > > rev_dns=no // (cmd. line: -R) > > > > log_facility=MY_LOG_FACILITY > > > > // Define the listening ports and addresses > > port=MY_SIP_PORT > > tls_port_no=MY_SIPS_PORT > > > > > > // Define TCP parameters > > tcp_children=4 > > tcp_no_connect=no > > tcp_rd_buf_size=16384 > > tcp_max_connections=2048 > > tcp_connection_lifetime=3605 > > disable_tcp=no > > #!ifdef WITH_TLS > > enable_tls=yes > > #!endif > > > > > > listen=MY_IP4_ADDR > > #!ifdef WITH_LOCALHOST_SIP > > listen=IP4_LOCALHOST > > #!endif > > #!ifdef WITH_IPV6 > > listen=MY_IP6_ADDR > > #!ifdef WITH_LOCALHOST_SIP > > listen=IP6_LOCALHOST > > #!endif > > #!endif > > > > #!ifdef WITH_WEBSOCKETS > > listen=MY_WS4_ADDR > > #!ifdef WITH_LOCALHOST_WS > > listen=LOCALHOST_WS4_ADDR > > #!endif > > #!ifdef WITH_IPV6 > > listen=MY_WS6_ADDR > > #!ifdef WITH_LOCALHOST_WS > > listen=LOCALHOST_WS6_ADDR > > #!endif > > #!endif > > #!ifdef WITH_TLS > > listen=MY_WSS4_ADDR > > #!ifdef WITH_LOCALHOST_WS > > listen=LOCALHOST_WSS4_ADDR > > #!endif > > #!ifdef WITH_IPV6 > > listen=MY_WSS6_ADDR > > #!ifdef WITH_LOCALHOST_WS > > listen=LOCALHOST_WSS6_ADDR > > #!endif > > #!endif > > #!endif > > #!endif > > > > force_rport=yes // Add received port to VIA > > > > loadmodule "pv.so" // Pseudo-Variable support > > loadmodule "xlog.so" // Formatted logging support > > loadmodule "sl.so" // Stateless request handling > > loadmodule "siputils.so" // SIP utility helpers > > loadmodule "sanity.so" // Sanity check > > loadmodule "maxfwd.so" // Support for Max-Forward header > > loadmodule "textops.so" // Text operations on SIP messages > > loadmodule "textopsx.so" // Extended Text operations on SIP messages > > loadmodule "sdpops.so" // SDP operations > > loadmodule "nathelper.so" // NAT traversal helpers > > loadmodule "rr.so" // Record Route handling > > loadmodule "outbound.so" // Outbound support > > loadmodule "path.so" // Path handling > > > > #!ifdef WITH_TLS > > loadmodule "tls.so" // TLS Support > > #!endif > > > > #!ifdef WITH_WEBSOCKETS > > loadmodule "xhttp.so" // XHTTP support > > loadmodule "websocket.so" // WebSocket support > > #!endif > > loadmodule "rtpengine.so" // RTPEngine control > > > > #!ifdef WITH_DEBUG > > loadmodule "debugger.so" > > #!endif > > > > When I start Kamailio, I see the following in the output to stderr: > > > > Listening on > > udp: 172.16.0.57:5060 > > tcp: 172.16.0.57:5060 > > Aliases: > > tcp: VRTPENGINE:5060 > > udp: VRTPENGINE:5060 > > > > WARNING: no fork mode > > 0(6299) INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt > as the io watch method (auto detected) > > > > Here's the problem: > > > > When I connect via UDP, everything works fine. I can connect and see all > SIP traffic. > > > > However, if I try to connect via TCP, I see the initial SYN packet in > attempting to establish a connection. > > > > Kamailio responds with an [RST,ACK] message and the TCP connection is > never established. No further TCP traffic is sent. > > > > Any ideas on what I'm doing wrong? > > <https://www.google.com/maps/search/3400%0D%0A+North+Ashton+Blvd,+Suite+470+…> > > > > *Peter H. Manley, Ph.D.* > > Chief Software Architect > > KomBea Corporation > > 3400 North Ashton Blvd, Suite 470 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > Lehi, Utah 84043 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > www.kombea.com<http://www.kombea.com> > <https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> > > > > Office:+1.801.610.5808 > > Mobile +1.801.243.7773 > > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > * sr-users at lists.kamailio.org<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -- VoIP Embedded, Inc. http://www.voipembedded.com<http://www.voipembedded.com/>

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Unable to establish TCP connection with Kamailio
by Peter Manley 17 Jun '21

17 Jun '21

Hello, I'm running Kamailio Version 5.3.2 on Ubuntu 20.04. Here my build information: version: kamailio 5.3.2 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 9.2.1 I have the following configuration set in my kamalio.cfg file: ##!define WITH_MYSQL ##!define WITH_AUTH ##!define WITH_USRLOCDB ##!define WITH_TLS ##!define WITH_HOMER ##!define WITH_WEBSOCKETS ##!define WITH_ANTIFLOOD ##!define WITH_IPV6 ##!define WITH_BRIDGE_ON_FAIL ##!define WITH_LOCALHOST_WS ##!define WITH_LOCALHOST_SIP #!substdef "!MY_SIP_PORT!5060!g" #!substdef "!MY_SIPS_PORT!5061!g" #!substdef "!MY_WS_PORT!8088!g" #!substdef "!MY_WSS_PORT!8089!g" #!substdef "!MY_LOG_FACILITY!LOG_DAEMON!g" #!substdef "!MY_IP4_ADDR!172.16.0.57!g" #!substdef "!IP4_LOCALHOST!127.0.0.1!g" #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" #!ifdef WITH_IPV6 #!substdef "!MY_IP6_ADDR![fe80::d6e4:2f5f:e5b7:7dff]!g" #!substdef "!IP6_LOCALHOST![::1]!g" #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" #!endif #!substdef "!MY_DOMAIN!rtpengine.kombea.io!g" debug=10 // debug level (cmd line: -dddddddddd) children=4 #fork=yes force_rport=yes // Add recieved port to VIA #log_stderror=no // (cmd line: -E) // Uncomment these lines to enter debugging mode fork=no log_stderror=yes check_via=no // (cmd. line: -v) dns=no // (cmd. line: -r) rev_dns=no // (cmd. line: -R) log_facility=MY_LOG_FACILITY // Define the listening ports and addresses port=MY_SIP_PORT tls_port_no=MY_SIPS_PORT // Define TCP parameters tcp_children=4 tcp_no_connect=no tcp_rd_buf_size=16384 tcp_max_connections=2048 tcp_connection_lifetime=3605 disable_tcp=no #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP4_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP4_LOCALHOST #!endif #!ifdef WITH_IPV6 listen=MY_IP6_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP6_LOCALHOST #!endif #!endif #!ifdef WITH_WEBSOCKETS listen=MY_WS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS6_ADDR #!endif #!endif #!ifdef WITH_TLS listen=MY_WSS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WSS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS6_ADDR #!endif #!endif #!endif #!endif force_rport=yes // Add received port to VIA loadmodule "pv.so" // Pseudo-Variable support loadmodule "xlog.so" // Formatted logging support loadmodule "sl.so" // Stateless request handling loadmodule "siputils.so" // SIP utility helpers loadmodule "sanity.so" // Sanity check loadmodule "maxfwd.so" // Support for Max-Forward header loadmodule "textops.so" // Text operations on SIP messages loadmodule "textopsx.so" // Extended Text operations on SIP messages loadmodule "sdpops.so" // SDP operations loadmodule "nathelper.so" // NAT traversal helpers loadmodule "rr.so" // Record Route handling loadmodule "outbound.so" // Outbound support loadmodule "path.so" // Path handling #!ifdef WITH_TLS loadmodule "tls.so" // TLS Support #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" // XHTTP support loadmodule "websocket.so" // WebSocket support #!endif loadmodule "rtpengine.so" // RTPEngine control #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif When I start Kamailio, I see the following in the output to stderr: Listening on udp: 172.16.0.57:5060 tcp: 172.16.0.57:5060 Aliases: tcp: VRTPENGINE:5060 udp: VRTPENGINE:5060 WARNING: no fork mode 0(6299) INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt as the io watch method (auto detected) Here's the problem: When I connect via UDP, everything works fine. I can connect and see all SIP traffic. However, if I try to connect via TCP, I see the initial SYN packet in attempting to establish a connection. Kamailio responds with an [RST,ACK] message and the TCP connection is never established. No further TCP traffic is sent. Any ideas on what I'm doing wrong? Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

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ATIS-1000085 STIR/SHAKEN DIV PASSporT
by Steven Wheeler 17 Jun '21

17 Jun '21

My Google searches aren't turning up any results, probably because this standard isn't finalized yet, but is anyone aware of a module which implements DIV PASSporTs for diverted calls? More details on the standard here: https://transnexus.com/blog/2020/shaken-div-std-letter-ballot/ Steven Wheeler

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Re: [SR-Users] Roadmap to the next major release - v5.5.0
by Henning Westerholt 16 Jun '21

16 Jun '21

Hello, If I remember correctly then in Debian Buster, the necessary library is not included and therefore the package cannot be build. You could use bullseye, the upcoming version, of compile by yourself. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: David Villasmil <david.villasmil.work(a)gmail.com> Sent: Wednesday, June 16, 2021 8:54 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Roadmap to the next major release - v5.5.0 I don't see the module in deb http://deb.kamailio.org/kamailio55 buster main Regards, David Villasmil email: david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337 On Wed, Jun 16, 2021 at 9:12 AM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, if you are referring to the module, its available in this release: https://kamailio.org/docs/modules/5.5.x/modules/secsipid.html Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of David Villasmil Sent: Wednesday, June 16, 2021 10:03 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; miconda(a)gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Roadmap to the next major release - v5.5.0 Hello Daniel, Is it possible at all to get secsipid in? Thanks David On Thu, 25 Feb 2021 at 15:21, Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>> wrote: Hello, it is the time to plan a bit the road to the next major release, to be versioned 5.5.0. The 5.4.0 was release by end of July 2020, therefore we are approaching the end of our usual development cycle, also during the last online devel meeting we considered the 2nd part of spring 2021 as a target time frame for 5.5. To narrow it down, I would propose to freeze the development at the end of March 2021, test during April and maybe beginning of May, then release v5.5.0. So far we have a lot of development to existing components, by the freeze date I expect 3-5 new modules to be in the repo as well. If anyone wants a different time line towards 5.5.0, let's discuss and choose the one that suits most of the developers. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337

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Roadmap to the next major release - v5.5.0
by Daniel-Constantin Mierla 16 Jun '21

16 Jun '21

Hello, it is the time to plan a bit the road to the next major release, to be versioned 5.5.0. The 5.4.0 was release by end of July 2020, therefore we are approaching the end of our usual development cycle, also during the last online devel meeting we considered the 2nd part of spring 2021 as a target time frame for 5.5. To narrow it down, I would propose to freeze the development at the end of March 2021, test during April and maybe beginning of May, then release v5.5.0. So far we have a lot of development to existing components, by the freeze date I expect 3-5 new modules to be in the repo as well. If anyone wants a different time line towards 5.5.0, let's discuss and choose the one that suits most of the developers. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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Using Kamailio with Asterisk/pjsip outbound_publish - early state is not notified
by Barry Flanagan 15 Jun '21

15 Jun '21

Hi Using Kamailio 5.4 with Asterisk 18 and using the pjsip outbound-publish to allow Kamailio to handle BLF for phones. BLF sort of works, in that when a line is on a call, the BLF reflects that. However the early/ringing state does not generate a NOTIFY to the phone, although Asterisk is sending the Publish and is receiving a 200OK reply from Kamailio. I would presume that since basic BLF is working that this early state PUBLISH should generate a NOTIFY but it doesn't Module params are as follows: # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:192.168.245.250:5060") modparam("presence", "local_log_level", 3) modparam("presence", "publ_cache", 0) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "sip_uri_match", 1) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) modparam("presence_dialoginfo", "force_single_dialog", 0) modparam("presence_dialoginfo", "force_dummy_dialog", 1) Any help appreciated! -Barry

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Kamailio v5.5.1 Released
by Daniel-Constantin Mierla 15 Jun '21

15 Jun '21

Hello, Kamailio SIP Server v5.5.1 stable release is out. This is a maintenance release of the latest stable branch, 5.5, that includes fixes since the release of v5.5.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.1. For more details about version 5.5.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/06/kamailio-v5-5-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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

1 0

Releasing Kamailio v5.5.1
by Daniel-Constantin Mierla 15 Jun '21

15 Jun '21

Hello, I am considering to release Kamailio v5.5.1 sometime next week, likely on Tuesday or Wednesday (June 15 or 16, 2021). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Is Kamailio GPL or GPLv2
by Yan Gao 15 Jun '21

15 Jun '21

Hello, I have a question about the license of Kamailio. It is said to be GPL in https://www.kamailio.org/w/: "Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second...." But GPLv2 in https://www.kamailio.org/w/info/ : "Kamailio is released under GNU Public License v2 (GPLv2). Beginning with 3.0.0, the application includes parts of code under BSD license that can be used such as individual components." Which one is the correct description? Thanks & Regards, Yan

2 2

vsf/vst question
by David Escartín 15 Jun '21

15 Jun '21

Hello all! as we knew vsf/vst were added to the RR header when using uac_replace for From/To headers. BUt we observed that some versions ago (we are using now 5.4) that stop happening with this setup modparam("uac", "restore_mode", "auto") modparam("uac", "restore_dlg", 1) modparam("rr", "enable_full_lr", 0) modparam("rr", "enable_double_rr", 1) modparam("rr", "append_fromtag", 1) and only ftag, did are added as param to RR However we have seen that when using multihomed kamailio instance and doing uac_replace, vsf/vst are included also with ftag and did params. Is that behaviour known or expected? thanks a lot and regards david -- <http://www.sonoc.io/> David Escartín NOC engineer *www.sonoc.io* <http://www.sonoc.io/> [image: LinkedIn] <https://www.linkedin.com/company/sonoc> [image: Twitter] <https://twitter.com/sonoc_> [image: Facebook] <https://www.facebook.com/SONOCio/> This e-mail is for the exclusive use of its recipients and may contain business secrets or other confidential or privileged information. Any unauthorised use, copying, publication or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please inform us immediately by replying to this e-mail and delete it, including any attachments or copies on your system. In accordance with the GDPR (EU) 2016/679 and the LOPDGDD 3/2018, we inform you that this e-mail address and/or any other personal data you have provided us with will be treated by SONOC with absolute confidentiality and with the only purpose of providing you with the requested services, due to your condition as a client, supplier or because you have requested information from us at any time. These data will only be kept for as long as required to comply with legal obligations. You can exercise your rights at any time by sending an e-mail to: *dataprotection(a)sonoc.io* <dataprotection(a)sonoc.io>*.* <dataprotection(a)sonoc.io>

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using path in Subscriber table
by Thành Trần Duy 14 Jun '21

14 Jun '21

Hi all, First of all, I would like to present the system model we are applying: - dSIPRouter : Sip relay for all users connected from the internet (installed with RTPEngine to handle media data). - kamailio 5.3.5: core system, registration server. - Asterisk 17: SBC. Sorry if our model seems a bit backwards but we have so many users (>12k configured users, > 3k concurrent registered users, > 500 concurrent users at cps 10), I don't know of any solution that can work with such a large load. With call outbound (from user to PSTN via SBC), everything is fine. But with internal or inbound calls, it wasn't successful. I showed that: - After dSIPRouter, the register message has Path parameter in SIP Header, - Kamailio store sip:user@domain in column contact at location table. - Using modparam registrar with use_path=1, path_mode=0, path_use_received=1, I stored this parameter in DB. - When calling internal or inbound, for the search destination of a phone, I am using sip_trace() function. But the result is always column contact. - With this uri, kamailio wasn't known and sent the invite to proxy. - Final error message: 404. I need one of these solutions: - get path column when result of sip_trace() like sip:user@domain (with phone in internal network, it like sip:user@ipaddress). - write path value to column contact when the register message has it. - sip_trace can get a path instead of contact if it has. Or you can have other solutions, pls help me. We can talk and test via my Webex PMR. Many thanks, Thanh Duy Tran

3 3

set empty custom global variable
by Sergey Safarov 14 Jun '21

14 Jun '21

In Kamailio config I use empty global string variable local.trace_troubleshooting_net = "" descr "trace all SIP messages from given network" When I check variable value I get blank string kamcmd cfg.get local trace_troubleshooting_net I can set new value [root@host]# kamcmd cfg.sets local trace_troubleshooting_net "2600::/48" [root@host]# kamcmd cfg.get local trace_troubleshooting_net 2600::/48 Now I want to set the empty custom global variable using a command kamcmd cfg.sets group trace_troubleshooting_net "" But get error error: 400 - error at parameter 2: expected string type but record doesn't match type How to properly set empty values?

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Kamailio 5.5.0 | CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 26
by Laurent Schweizer 11 Jun '21

11 Jun '21

Dear all, I setup a Kamailio 5.5.0 (from GIT) on a ubuntu Ubuntu 18.04.5 LTS and I have some error with the libssl/ libmysql, seems not happening every time. Regarding the config file, it's a config running correctly on a Kamailio 5.3.x on a centos 7 I have attached the returned error and the result of the core file. Any idea ? BR Laurent version: kamailio 5.5.0 (x86_64/linux) 66370c flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 66370c compiled on 07:23:15 Jun 11 2021 with gcc 7.5.0 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: loading modules under config path: /usr/local/lib64/kamailio/modules/ Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: Listening on Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: udp: 80.132.9.112:5060 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: udp: 80.132.9.112:6000 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tcp: 80.132.9.112:5060 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tcp: 80.132.9.112:6000 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tls: 80.132.9.112:5061 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: Aliases: Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tls: zrh02-srv03:5061 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tcp: zrh02-srv03:6000 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tcp: zrh02-srv03:5060 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: udp: zrh02-srv03:6000 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: udp: zrh02-srv03:5060 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: *: sipbkp.demo.ch:* Jun 11 21:35:45 zrh02-srv03 /usr/local/sbin/kamailio[7135]: WARNING: tls [tls_init.c:793]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 13107200 and 6553600 bytes Jun 11 21:35:45 zrh02-srv03 /usr/local/sbin/kamailio[7135]: NOTICE: tls [tls_domain.c:1136]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... Jun 11 21:35:46 zrh02-srv03 kernel: [304361.950608] traps: kamailio[7147] general protection ip:7f5d209a344b sp:7ffe6ae5adb8 error:0 in libcrypto.so.1.1[7f5d207b5000+29b000] Jun 11 21:35:46 zrh02-srv03 systemd[1]: Started Kamailio SIP Server. Jun 11 21:35:54 zrh02-srv03 /usr/local/sbin/kamailio[7161]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 26 Jun 11 21:35:54 zrh02-srv03 /usr/local/sbin/kamailio[7135]: ALERT: <core> [main.c:788]: handle_sigs(): child process 7147 exited by a signal 11 Jun 11 21:35:54 zrh02-srv03 /usr/local/sbin/kamailio[7135]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated Jun 11 21:35:54 zrh02-srv03 systemd[1]: kamailio.service: Main process exited, code=exited, status=1/FAILURE Jun 11 21:35:54 zrh02-srv03 systemd[1]: kamailio.service: Failed with result 'exit-code'. This GDB was configured as "x86_64-linux-gnu". Type "show configuration" for configuration details. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/. Find the GDB manual and other documentation resources online at: http://www.gnu.org/software/gdb/documentation/. For help, type "help". Type "apropos word" to search for commands related to "word"... Reading symbols from /usr/local/sbin/kamailio...done. [New LWP 7147] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -m 5000 -M 128 -P /run/kamailio/kamailio.pid -f /confi'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f5d209a344b in X509_LOOKUP_by_subject () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 (gdb) bt #0 0x00007f5d209a344b in X509_LOOKUP_by_subject () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #1 0x00007f5d209a3a5b in X509_STORE_CTX_get_by_subject () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #2 0x00007f5d209a418d in X509_STORE_CTX_get1_issuer () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #3 0x00007f5d209a9207 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #4 0x00007f5d209a9f96 in X509_verify_cert () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #5 0x00007f5d20caf337 in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #6 0x00007f5d20cd037b in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #7 0x00007f5d20cd2995 in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #8 0x00007f5d20ccc48c in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #9 0x00007f5d20cb84e4 in SSL_do_handshake () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #10 0x00007f5d215cbdb1 in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 #11 0x00007f5d2159f4de in mysql_real_connect () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 #12 0x00007f5d21b410b1 in db_mysql_new_connection (id=0x7f5d21ee2b30) at km_my_con.c:157 #13 0x00007f5d211456dc in db_do_init2 (url=0x7f5d1e9dd4e0 <ul_db_url>, new_connection=0x7f5d21b3f6a4 <db_mysql_new_connection>, pooling=DB_POOLING_PERMITTED) at db.c:317 #14 0x00007f5d211443f9 in db_do_init (url=0x7f5d1e9dd4e0 <ul_db_url>, new_connection=0x7f5d21b3f6a4 <db_mysql_new_connection>) at db.c:270 #15 0x00007f5d21b3459e in db_mysql_init (_url=0x7f5d1e9dd4e0 <ul_db_url>) at km_dbase.c:202 #16 0x00007f5d1e7a1987 in child_init (_rank=-1) at usrloc_mod.c:471 #17 0x000055ddd6c45602 in init_mod_child (m=0x7f5d21e03818, rank=-1) at core/sr_module.c:864 #18 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e03bb8, rank=-1) at core/sr_module.c:860 #19 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e04498, rank=-1) at core/sr_module.c:860 #20 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e056a8, rank=-1) at core/sr_module.c:860 #21 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e06390, rank=-1) at core/sr_module.c:860 #22 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e06b38, rank=-1) at core/sr_module.c:860 #23 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e06fb8, rank=-1) at core/sr_module.c:860 #24 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e075f0, rank=-1) at core/sr_module.c:860 #25 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e090c0, rank=-1) at core/sr_module.c:860 #26 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e09500, rank=-1) at core/sr_module.c:860 #27 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0a400, rank=-1) at core/sr_module.c:860 #28 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0aa70, rank=-1) at core/sr_module.c:860 #29 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0b0b0, rank=-1) at core/sr_module.c:860 #30 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0b5d8, rank=-1) at core/sr_module.c:860 #31 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0bdd0, rank=-1) at core/sr_module.c:860 #32 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0c968, rank=-1) at core/sr_module.c:860 #33 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0f7f8, rank=-1) at core/sr_module.c:860 #34 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e105d8, rank=-1) at core/sr_module.c:860 #35 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e109b0, rank=-1) at core/sr_module.c:860 #36 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e10ea0, rank=-1) at core/sr_module.c:860 #37 0x000055ddd6c45ef9 in init_child (rank=-1) at core/sr_module.c:909 #38 0x000055ddd6b9eebe in fork_process (child_id=-1, desc=0x55ddd6ee7847 "slow timer", make_sock=1) at core/pt.c:337 #39 0x000055ddd6abaa62 in main_loop () at main.c:1814 #40 0x000055ddd6ac6095 in main (argc=9, argv=0x7ffe6ae5d298) at main.c:3053

1 0

Kamailio SBC+ Freeswitch as media server
by John Tuxies 09 Jun '21

09 Jun '21

in a Debian system i am running Kamailio 5.5 installed from repo and with the basic setup i have 2 users 101 & 102 to be able to call each other. So far so good. i added Freeswitch in the same system and tried to follow this guide http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc even though it is old and there are many changes. Freeswitch is installed from deb repo as well. after following the guide i tried to adapt to newer versions of kamailio and freeswitch, but i am getting errors. The problem is that the users now are able to register to Kamailio, but unable to call each other. I can see users being registered in Kamailio: *root@kamfs: ~ $ kamctl online101102root@kamfs: ~ $* while when 101 calls 102 i am getting: *2021-06-08 10:35:19.015670 [NOTICE] switch_channel.c:1118 New Channel sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> [f3b8f5a0-652a-4640-b2d3-9b7cc21d7f91]2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Running State Change CS_NEW (Cur 1 Tot 5)2021-06-08 10:35:19.015670 [INFO] sofia.c:10362 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> receiving invite from 192.168.1.15:5060 <http://192.168.1.15:5060> version: 1.10.6 -release-18-1ff9d0a60e 64bit call-id: ZDFiZWFkZmVmYjgxZDEwOTgyYTYxZjkxNzkwMzlmYmQ.2021-06-08 10:35:19.015670 [DEBUG] sofia.c:10456 verifying acl "domains" for ip/port 192.168.1.15:0 <http://192.168.1.15:0>.2021-06-08 10:35:19.015670 [WARNING] sofia.c:10569 IP 192.168.1.15 Rejected by acl "domains"2021-06-08 10:35:19.015670 [NOTICE] sofia.c:2445 Hangup sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> [CS_NEW] [CALL_REJECTED]2021-06-08 10:35:19.015670 [DEBUG] sofia.c:1549 Channel is already hungup.2021-06-08 10:35:19.015670 [DEBUG] sofia.c:1549 Channel is already hungup.2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:604 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State NEW2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Running State Change CS_HANGUP (Cur 1 Tot 5)2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:848 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Callstate Change DOWN -> HANGUP2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State HANGUP2021-06-08 10:35:19.015670 [DEBUG] mod_sofia.c:453 Channel sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> hanging up, cause: CALL_REJECTED2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:60 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> Standard HANGUP, cause: CALL_REJECTED2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State HANGUP going to sleep2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State Change CS_HANGUP -> CS_REPORTING2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Running State Change CS_REPORTING (Cur 1 Tot 5)2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:936 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State REPORTING2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:174 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> Standard REPORTING, cause: CALL_REJECTED2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:936 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State REPORTING going to sleep2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:611 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State Change CS_REPORTING -> CS_DESTROY2021-06-08 10:35:19.015670 [DEBUG] switch_core_session.c:1736 Session 5 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Locked, Waiting on external entities2021-06-08 10:35:19.015670 [NOTICE] switch_core_session.c:1754 Session 5 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Ended2021-06-08 10:35:19.015670 [NOTICE] switch_core_session.c:1758 Close Channel sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> [CS_DESTROY]2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Running State Change CS_DESTROY (Cur 0 Tot 5)2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State DESTROY2021-06-08 10:35:19.015670 [DEBUG] mod_sofia.c:364 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> SOFIA DESTROY2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:181 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> Standard DESTROY2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State DESTROY going to sleepfreeswitch@kamfs>* i am attaching the kamailio.cfg file and the whole /etc/freeswitch folder, if someone could help me to solve this issue, please. Kind regards, John

3 2

Any way to force socket used to reach databases ?
by David VILLAUME 08 Jun '21

08 Jun '21

Hello, I’m trying to initiate a redis connection in ndb_redis for topos_redis on a keepalived slave host, but host tries to use the VIP instead of the second socket I defined. Is there a way to influence the ‘database source socket’ so the server can connect with a secondary address even if it does not own the keepalived VIP (tried mhomed options without improvment) ? Regards, David

2 1

List of Header Names
by Evan Verma 07 Jun '21

07 Jun '21

hello! new to working with sip/kamailio. I am currently trying to get a list of all of the header names on the invite the SBC receives. I am having trouble finding a way to do this. Does anyone know of a good solution?

5 4

Problem NOTIFY in Presence module
by hamed r 07 Jun '21

07 Jun '21

hello dears ,i am using presence module and NOTIFY message are working properly. but when i put the database on another server , NOTIFY message will not be sent to subscribers. what is problem?

2 2

List of Header Names
by Evan Verma 05 Jun '21

05 Jun '21

hello! new to working with sip/kamailio. I am currently trying to get a list of all of the header names on the invite the SBC receives. I am having trouble finding a way to do this. Does anyone know of a good solution? -- Evan

2 1

STIR/SHAKEN
by David Villasmil 04 Jun '21

04 Jun '21

Hello guys, So I'm testing secsipid with made up certificates, basically: [image: Untitled.png] The verification _always_ comes back OK, event if I create the Identity with different From numbers, should that fail? Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Info: ClueCon Weekly about Kamailio 5.5 series - Jun 4, 2021, 10:00am CT
by Daniel-Constantin Mierla 04 Jun '21

04 Jun '21

Hello, short announcement to let everyone know that next Friday, Jun 4, 2021, at 10:00am CT (17:00 Berlin), I will join ClueCon Weekly to discuss about latest Kamailio 5.5 series. If you have questions about or just want to be part of the discussions, consider to join the live video conference session, more details at: * https://www.cluecon.com/cclive-6-4/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

About Kamailio ass authentication service
by Nael Silega 04 Jun '21

04 Jun '21

hi, i have a question, can i use Kamailio as a authentication service? i´m traing to set up STIR/SHAKEN framework and im looking for a server that work as authentication service

3 2

Questions on http_async_query() in event_route[xhttp:request] and async best practices for Kamailio
by James Aylett 04 Jun '21

04 Jun '21

Hi, I am using Kamailio v5.2 as a WebRTC proxy for SIP clients, as part of my setup I call a REST API before calling ws_handle_handshake() in event_route[xhttp:request] using http_client_query() to authenticate and retrieve some user details (routing etc) for the user connected via WebSockets. This worked fine until increased load on the system meant API responses were slow, this caused a knock-on effect to existing connected users. Specifically, we saw mass WebSocket disconnects for existing connected users - believed to be due to using too aggressive proxy timeout settings on our reverse proxy in front of Kamailio. My understanding is that event_route[xhttp:request] uses the shared SIP TCP worker threads, so potentially could slow API requests over the network block all the handlers? Could this potentially impact the keepalive processes used by the WebSocket module to check existing WebSocket connections? We were using ping keepalives, again with pretty aggressive timers... To resolve this issue, I am looking at whether I can call API requests asynchronously so that TCP workers are not blocked, my first thought was to use http_async_query() in event_route[xhttp:request] and call ws_handle_handshake() in the HTTP_REPLY route when the API call had completed, but I get "ERROR: websocket [ws_handshake.c:143]: ws_handle_handshake(): retrieving connection". I presume then this approach is a deadend, looking at newer kamailio versions 5.5, there doesn't seem to be any way to do this, correct? Instead of using http_async_query() in event_route[xhttp:request], I presume I could set a short timeout in http_client_query(), but my concern is new WebSocket connections could still impact and block existing ones in this case... My other thought was to move my API call out from the event_route[xhttp:request] into my route handler for REGISTER requests, thinking I could offload new register requests to async workers and move the API call there so as to not block existing connections. Does this seem like a reasonable approach to the problem? Given my current script does not use any async workers, how would one go about optimising the number of plain SIP TCP workers to async workers, are there any guides for this? In general, for any potentially blocking "APIs" or calls over the network, is it best practice to offload them into async workers. For example, we run RTPEngine on separate hosts to Kamailio so should I be wrapping rtpengine_offer/answer calls into async workers in case of a slow network? Apologies for the long text, but I would really appreciate any help to understand these problems. Thank you

2 1

Need help analyze crash
by Andrew Chen 03 Jun '21

03 Jun '21

Hi all, Today I upgraded Kamailio to 5.4.5 and a random crash was just reported in our crash monitoring script. Below is the output of the crash: #3 0x00007fa149ebe58d in ?? () from /usr/lib/x86_64-linux-gnu/kamailio/modules/tls.so #4 0x00007fa14989e277 in ERR_clear_error () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #5 0x00007fa149933d81 in X509_STORE_set_default_paths () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #6 0x00007fa10310d374 in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 #7 0x00007fa10310d9f3 in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 #8 0x00007fa1030e0427 in mysql_real_connect () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 #9 0x00007fa10368428c in db_mysql_new_connection () from /usr/lib/x86_64-linux-gnu/kamailio/modules/db_mysql.so #10 0x00007fa1501f96dc in db_do_init2 () from /usr/lib/x86_64-linux-gnu/kamailio/libsrdb1.so.1 #11 0x00007fa1501f83f9 in db_do_init () from /usr/lib/x86_64-linux-gnu/kamailio/libsrdb1.so.1 #12 0x00007fa103689710 in db_mysql_init () from /usr/lib/x86_64-linux-gnu/kamailio/modules/db_mysql.so #13 0x00007fa15062d35f in carrierroute_db_open () from /usr/lib/x86_64-linux-gnu/kamailio/modules/carrierroute.so #14 0x00007fa1506719c6 in ?? () from /usr/lib/x86_64-linux-gnu/kamailio/modules/carrierroute.so Is this crash due to libcrypto or mysql client? Currently I have the following mysql client installed on the kamailio instance: ii lib*mysql*client20:amd64 5.7.34-0ubuntu0.18.04.1 amd64 MySQL database client library ii *mysql*-client 5.7.34-0ubuntu0.18.04.1 all MySQL database client (metapackage depending on the latest version) ii *mysql*-client-5.7 5.7.34-0ubuntu0.18.04.1 amd64 MySQL database client binaries ii *mysql*-client-core-5.7 5.7.34-0ubuntu0.18.04.1 amd64 MySQL database core client binaries ii *mysql*-common 5.8+1.0.4 all MySQL database common files, e.g. /etc/*mysql*/my.cnf root@ashintgtpsg51:/var/lib/cores # apt-cache madison mysql-client mysql-client | 5.7.34-0ubuntu0.18.04.1 | http://us-east-1.ec2.archive.ubuntu.com/ubuntu bionic-updates/main amd64 Packages mysql-client | 5.7.34-0ubuntu0.18.04.1 | http://security.ubuntu.com/ubuntu bionic-security/main amd64 Packages mysql-client | 5.7.21-1ubuntu1 | http://us-east-1.ec2.archive.ubuntu.com/ubuntu bionic/main amd64 Packages root@ashintgtpsg51:/var/lib/cores # Any suggestions would be greatly appreciated. Thanks in advance. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 10

handle_ruri_alias() and multiple aliases
by Denys Pozniak 02 Jun '21

02 Jun '21

Hello! I need help understanding how the *handle_ruri_alias()* function works. *Call-flow:* Upstream operator -> Kamailio Proxy -> Edpoint. The upstream operator in the initial SIP INVITE in the Contact field sends us an alias parameter, in turn, our Kamailio Proxy adds its own too. The Contact after the Proxy looks something like this: *Contact: <sip:10.0.0.115:5060;alias=10.0.0.115~5060~1;alias=3.3.3.3~5060~1>* When the Endpoint sends us a SIP BYE, then the *handle_ruri_alias()* absorbs not the last alias, but the first one, and this leads to the incorrect formation of a $du. The RURI after the Proxy looks something like this: *Request-Line: BYE sip:10.0.0.115:5060;alias=3.3.3.3~5060~1 SIP/2.0* The first thing that comes to mind is a modification of the original Contact field with storing it's within a dialogue ... Is it possible to elegantly solve this problem? -- BR, Denys Pozniak

4 7

kamailio 5.4.5 crash after dialog [dlg_hash.c:1182]: next_state_dlg(): bogus event 2 in state 5 for dlg
by Trevor Hemsley 02 Jun '21

02 Jun '21

Hi We've seen a few occurrences of this crash since we implemented kamailio 5.4.5 for our inbound call traffic. Preceding the crash we always get a message in the logs about "dialog [dlg_hash.c:1182]: next_state_dlg(): bogus event 2 in state 5 for dlg" and whenever we see that message, we get the crash. Jun 1 10:12:14 thissystem /usr/sbin/kamailio[20001]: CRITICAL: {2 1 INVITE 6787142-3831531134-1330894187(a)some.telco.domain} dialog [dlg_hash.c:1182]: next_state_dlg(): bogus event 2 in state 5 for dlg 0x7fae153d3cf0 [3973:6059] with clid '82608924-3831089984-1833452161(a)some.telco.domain' and tags '3831089984-684203260' '' The crash happens on line 879 of https://github.com/kamailio/kamailio/blob/master/src/modules/dialog/dlg_db_… SET_STR_VALUE(values+7, cell->bind_addr[DLG_CALLER_LEG]->sock_str); Looking at gdb, I see that cell->bind_addr[DLG_CALLER_LEG] == 0 so trying to use it results in the segfault. I have thought about patching that to check for bind_addr being set and bypassing that line if not but I do not think it's the right fix though it probably would avoid the current crash (and do odd things later instead!). I did check that everything else in that function that uses DLG_CALLER_LEG is set correctly and it appears that it is - or at least if not correctly then they are at least set! The "bogus event 2 in state 5" message is coming out from line 1143 in src/modules/ims_dialog/dlg_hash.c case DLG_EVENT_RPL1xx: switch (dlg->state) { case DLG_STATE_UNCONFIRMED: case DLG_STATE_EARLY: dlg->state = DLG_STATE_EARLY; break; default: log_next_state_dlg(event, dlg); } break; (I didn't really understand why it would set dlg->state = DLG_STATE_EARLY if it's already DLG_STATE_EARLY but...) I have a couple of core dumps of this from the last day or two - usually seems to crash at least once a week on our pair of kamailio inbound servers. If there's any more info I can provide to help fix this properly rather than my proposed bodge then feel free to ask. Trevor Hemsley Disclaimer The information contained in this communication from the sender is confidential. It is intended solely for use by the recipient and others authorized to receive it. If you are not the recipient, you are hereby notified that any disclosure, copying, distribution or taking action in relation of the contents of this information is strictly prohibited and may be unlawful. This email has been scanned for viruses and malware, and may have been automatically archived by Mimecast Ltd, an innovator in Software as a Service (SaaS) for business. Providing a safer and more useful place for your human generated data. Specializing in; Security, archiving and compliance. To find out more visit the Mimecast website.

2 2

UAC module register originating from private interface on server
by Jonathan Hunter 02 Jun '21

02 Jun '21

Hi, I am working with the UAC module and trying to register with providers, and can see that kamailio keeps sending the REGISTER from the private IP address socket on the server (eth0). Is it possible to set the default socket or send socket for the REGISTER message that gets sent out via kamailio? I am running version 4.4.7. I notice SIP OPTIONS are being sent out from the public IP address (eth1) on the server (as I have set socket in dispatcher) and was hoping for same behaviour with uac module. Please can anyone advise? Many thanks Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

2 4

STIR/SHAKEN tests
by David Villasmil 02 Jun '21

02 Jun '21

Hello guys, I want to test secsipid, but i don't yet have the certificate. So i thought i'd create a cert like: openssl req -new -newkey rsa:4096 -nodes -keyout snakeoil.key -out snakeoil.csr openssl x509 -req -sha256 -days 365 -in snakeoil.csr -signkey snakeoil.key -out snakeoil.pem Then i'm simply doing: $var(rc) = secsipid_add_identity("$fU", "$rU", "A", "", " https://somedomain.com/stir/$rd/cert.pem <https://kamailio.org/stir/$rd/cert.pem>", "/etc/kamailio/snakeoil.pem"); if ( $var(rc) ) { xlog("L_ERR", "[STIR/SHAKEN][$ci] Shaken authentication added (SIP Identity Header created)\n"); } else { xlog("L_ERR", "[STIR/SHAKEN][$ci] Failed\n"); } But no matter what i do it silently fails: INVITE d54c2919-39b6-123a-95a7-0e29a5289b8d} <script>: [STIR/SHAKEN][d54c2919-39b6-123a-95a7-0e29a5289b8d] Failed I have debug on 6, but i don't get more info regarding the error. Any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 11

Possible memory leak?
by George Diamantopoulos 02 Jun '21

02 Jun '21

Hello all, I've been experiencing memory exhaustion (all 8GiB of it) on hosts running kamailio for quite some time now while on the 5.4.x train. I can't really say about previous versions, because the setup was quite different back when I was using those. After about 2 days of operation, RAM is exhausted and kamailio is killed by the kernel OOM which frees up memory again. Restarting kamailio via systemd will also free up most memory. I can't seem to find what is using it up, so I'm guessing there might be a memory leak, but I'm not 100% sure so I thought I'd ask the list for opinions. I'm including the output of kamcmd pkg and shm stats and kamctl stats here. My take on those is that there's nothing to show where all this RAM is going. Kamailio version is 5.4.5+bpo10 on debian buster and is started with '-P /run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 2048 -M 16' (51 processes total). Thanks! kamctl stats { "jsonrpc": "2.0", "result": [ "core:bad_URIs_rcvd = 6470677", "core:bad_msg_hdr = 25370", "core:drop_replies = 0", "core:drop_requests = 342064", "core:err_replies = 0", "core:err_requests = 0", "core:fwd_replies = 24367", "core:fwd_requests = 375703", "core:rcv_replies = 45021597", "core:rcv_replies_18x = 834856", "core:rcv_replies_1xx = 1505425", "core:rcv_replies_1xx_bye = 17", "core:rcv_replies_1xx_cancel = 7", "core:rcv_replies_1xx_invite = 1497313", "core:rcv_replies_1xx_message = 0", "core:rcv_replies_1xx_prack = 0", "core:rcv_replies_1xx_refer = 0", "core:rcv_replies_1xx_reg = 0", "core:rcv_replies_1xx_update = 0", "core:rcv_replies_2xx = 41764997", "core:rcv_replies_2xx_bye = 328464", "core:rcv_replies_2xx_cancel = 223572", "core:rcv_replies_2xx_invite = 371129", "core:rcv_replies_2xx_message = 190", "core:rcv_replies_2xx_prack = 0", "core:rcv_replies_2xx_refer = 3205", "core:rcv_replies_2xx_reg = 12783948", "core:rcv_replies_2xx_update = 0", "core:rcv_replies_3xx = 1052", "core:rcv_replies_3xx_bye = 0", "core:rcv_replies_3xx_cancel = 0", "core:rcv_replies_3xx_invite = 1052", "core:rcv_replies_3xx_message = 0", "core:rcv_replies_3xx_prack = 0", "core:rcv_replies_3xx_refer = 0", "core:rcv_replies_3xx_reg = 0", "core:rcv_replies_3xx_update = 0", "core:rcv_replies_401 = 17120", "core:rcv_replies_404 = 525917", "core:rcv_replies_407 = 553", "core:rcv_replies_480 = 23138", "core:rcv_replies_486 = 74201", "core:rcv_replies_4xx = 1715492", "core:rcv_replies_4xx_bye = 2146", "core:rcv_replies_4xx_cancel = 594", "core:rcv_replies_4xx_invite = 317850", "core:rcv_replies_4xx_message = 208", "core:rcv_replies_4xx_prack = 0", "core:rcv_replies_4xx_refer = 1", "core:rcv_replies_4xx_reg = 2382", "core:rcv_replies_4xx_update = 0", "core:rcv_replies_5xx = 29462", "core:rcv_replies_5xx_bye = 15", "core:rcv_replies_5xx_cancel = 10", "core:rcv_replies_5xx_invite = 3501", "core:rcv_replies_5xx_message = 0", "core:rcv_replies_5xx_prack = 0", "core:rcv_replies_5xx_refer = 0", "core:rcv_replies_5xx_reg = 25157", "core:rcv_replies_5xx_update = 0", "core:rcv_replies_6xx = 5169", "core:rcv_replies_6xx_bye = 0", "core:rcv_replies_6xx_cancel = 0", "core:rcv_replies_6xx_invite = 5169", "core:rcv_replies_6xx_message = 0", "core:rcv_replies_6xx_prack = 0", "core:rcv_replies_6xx_refer = 0", "core:rcv_replies_6xx_reg = 0", "core:rcv_replies_6xx_update = 0", "core:rcv_requests = 67620661", "core:rcv_requests_ack = 707431", "core:rcv_requests_bye = 336874", "core:rcv_requests_cancel = 226878", "core:rcv_requests_info = 70", "core:rcv_requests_invite = 1038320", "core:rcv_requests_message = 796", "core:rcv_requests_notify = 6027113", "core:rcv_requests_options = 29079952", "core:rcv_requests_prack = 0", "core:rcv_requests_publish = 40344", "core:rcv_requests_refer = 3217", "core:rcv_requests_register = 20919429", "core:rcv_requests_subscribe = 5296608", "core:rcv_requests_update = 0", "core:unsupported_methods = 0", "dialog:active_dialogs = 458", "dialog:early_dialogs = 123", "dialog:expired_dialogs = 105", "dialog:failed_dialogs = 321224", "dialog:processed_dialogs = 645092", "dns:failed_dns_request = 534", "dns:slow_dns_request = 0", "mysql:driver_errors = 18", "shmem:fragments = 8836", "shmem:free_size = 1955034032", "shmem:max_used_size = 250791392", "shmem:real_used_size = 192449616", "shmem:total_size = 2147483648", "shmem:used_size = 105997528", "siptrace:traced_replies = 0", "siptrace:traced_requests = 0", "sl:1xx_replies = 0", "sl:200_replies = 14330540", "sl:202_replies = 0", "sl:2xx_replies = 0", "sl:300_replies = 0", "sl:301_replies = 0", "sl:302_replies = 0", "sl:3xx_replies = 0", "sl:400_replies = 3", "sl:401_replies = 7979168", "sl:403_replies = 14845", "sl:404_replies = 1159237", "sl:407_replies = 2445205", "sl:408_replies = 0", "sl:483_replies = 79336", "sl:4xx_replies = 28", "sl:500_replies = 0", "sl:5xx_replies = 0", "sl:6xx_replies = 0", "sl:failures = 1", "sl:received_ACKs = 342064", "sl:sent_err_replies = 0", "sl:sent_replies = 26008362", "sl:xxx_replies = 0", "sst:expired_sst = 0", "tcp:con_reset = 0", "tcp:con_timeout = 0", "tcp:connect_failed = 0", "tcp:connect_success = 0", "tcp:current_opened_connections = 0", "tcp:current_write_queue_size = 0", "tcp:established = 24562", "tcp:local_reject = 0", "tcp:passive_open = 24562", "tcp:send_timeout = 0", "tcp:sendq_full = 0", "tmx:2xx_transactions = 41122607", "tmx:3xx_transactions = 1049", "tmx:4xx_transactions = 1904640", "tmx:5xx_transactions = 8025", "tmx:6xx_transactions = 5165", "tmx:UAC_transactions = 2602132", "tmx:UAS_transactions = 43036902", "tmx:active_transactions = 145", "tmx:inuse_transactions = 888", "tmx:rpl_absorbed = 1541413", "tmx:rpl_generated = 1111996", "tmx:rpl_received = 44996889", "tmx:rpl_relayed = 43455476", "tmx:rpl_sent = 44567472" ], "id": 15546 } kamcmd mod.stats all pkg Module: core { init_io_wait(469): 8040 init_modules(911): 16 get_hdr_field(116): 464 parse_headers(325): 128 rpc_hash_add(151): 2048 cnt_hash_add(394): 2048 register_select_table(458): 48 sr_wtimer_add(373): 80 add_callback(59): 312 sr_wtimer_init(351): 136 cfg_declare(50): 3168 fix_sock_str(500): 480 fix_hostname(1466): 96 grp_hash_add(241): 728 init_dst_set(89): 32296 cfg_script_fixup(303): 8 cfg_script_fixup(247): 288 cfg_script_fixup(241): 120 new_db_id(318): 304 fix_expr(554): 192 parse_pvname_list(1599): 480 mk_switch_jmp_table(73): 448 fix_param(1140): 672 pv_parse_index(776): 1232 subst_parser(291): 144 subst_parser(274): 64 subst_parser(301): 8 route_new_list(199): 1152 init_io_wait(516): 7256 db_do_init2(299): 280 parse_select(204): 2496 fix_param(1079): 704 mk_match_cond_table(100): 944 mk_name_lst(3754): 72 mk_listen_id(3771): 96 yyparse(2853): 19728 fix_param(1045): 11792 fixup_regexp_null(213): 368 tr_new(1730): 12352 fix_match(490): 384 fixup_pvar_all(279): 14512 mk_rval_expr2(2703): 845408 mk_elem(90): 528 fix_match_rve(3059): 1464 pv_cache_add(349): 63872 mk_rval_expr1(2643): 522656 mk_case_stm(3879): 3432 mk_rval_expr_v(2562): 2513784 mk_action(118): 493240 parse_params2(593): 6496 set_mod_param_regex(145): 1248 set_mod_param_regex(134): 2520 tr_table_add(1959): 360 register_module(253): 39928 register_module(236): 4784 load_module(437): 2624 new_cfg_script_var(134): 88 dupl_string_name(74): 56 dupl_string(49): 184 new_cfg_script_var(126): 104 pv_parse_format(1173): 30744 cfg_new_group(79): 864 new_sock_info(304): 96 new_sock_info(299): 2400 yyparse(659): 352 yyparse(1879): 216 add_alias(103): 256 add_alias(99): 320 yyparse(2388): 408 yyparse(678): 368 sr_push_yy_state(1695): 208 sr_push_yy_state(1634): 584 addstr(1448): 572072 rpc_hash_add(105): 28128 str_hash_alloc(59): 512 pv_table_add(236): 19448 pv_init_buffer(2139): 327680 pv_init_buffer(2129): 320 init_nonsip_hooks(43): 24 rval_get_str(1281): 54920 init_rlist(146): 24 fix_socket_list(1675): 56 route_add(124): 5312 str_hash_alloc(59): 768 mk_rval_expr_v(2575): 3304 pp_define(1827): 1344 init_counters(125): 128 cnt_hash_add(341): 27856 str_hash_alloc(59): 1280 Total: 5708952 } Module: permissions { get_pathname(243): 64 Total: 64 } Module: auth { generate_random_secret(240): 32 generate_random_secret(239): 32 Total: 64 } Module: dialog { dlg_bridge_init_hdrs(66): 80 Total: 80 } Module: acc { parse_acc_extra(116): 6384 Total: 6384 } Module: xlog { mod_init(212): 8200 xdbg_fixup_helper(498): 7736 xlog_fixup_helper(533): 36280 Total: 52216 } Module: dmq { make_socket_str_from_uri(143): 24 Total: 24 } Module: rr { register_rrcb(61): 96 Total: 96 } Module: jsonrpcs { jsonrpc_dgram_mod_init(182): 40 jsonrpc_init_fifo_server(123): 128 jsonrpc_fifo_mod_init(525): 40 Total: 208 } Module: sqlops { sqlops_tr_buffer_init(46): 2048 sql_init_con(83): 480 Total: 2528 } Module: db_mysql { db_mysql_new_connection(78): 3560 db_mysql_new_connection(68): 96 Total: 3656 } Module: phonenum { sr_phonenum_add_item(99): 16 sr_phonenum_add_item(93): 608 pv_parse_phonenum_name(131): 224 Total: 848 } Module: pv { tr_parse_string(2717): 96 tr_parse_uri(2806): 64 tr_parse_string(2639): 336 tr_parse_string(2628): 424 tr_parse_paramlist(2978): 288 tr_parse_paramlist(2906): 192 tr_parse_string(2569): 120 pv_parse_xavp_name(541): 2664 pv_parse_xavp_name(515): 3720 tr_parse_string(2484): 864 tr_parse_string(2473): 904 add_var(58): 4384 tr_parse_string(2519): 2144 tr_parse_string(2509): 5184 add_var(65): 1544 Total: 22928 } Module: textops { tr_txt_parse_re(212): 32 hname_fixup(3528): 1960 fixup_method(3598): 1648 Total: 3640 } Module: tmx { pv_parse_t_var_name(390): 176 Total: 176 } Module: rtpengine { build_rtpp_socks(1801): 24 child_init(1987): 240 Total: 264 } Module: htable { fixup_ht_key(352): 352 pv_parse_ht_name(158): 2240 Total: 2592 } Module: textopsx { fixup_hname_param(593): 320 Total: 320 } Module: avpops { avpops_parse_pvar(57): 64 Total: 64 } Module: ndb_redis { redisc_add_server(348): 216648 Total: 216648 } Module: sl { sl_register_callback(514): 64 Total: 64 } Module: siptrace { mod_init(341): 384 Total: 384 } Module: tm { Total: 0 } Module: kex { Total: 0 } Module: topos { Total: 0 } Module: mtree { Total: 0 } Module: cfgutils { Total: 0 } Module: dispatcher { Total: 0 } Module: outbound { Total: 0 } kamcmd mod.stats all shm Module: core { dns_cache_mk_rd_entry(1110): 1288 counters_prefork_init(212): 65280 cfg_clone_str(130): 136 cfg_shmize(217): 824 main_loop(1341): 8 init_pt(107): 8 init_pt(106): 8 init_pt(105): 7552 build_req_buf_from_sip_req(2218): 791408 xavp_new_value(116): 186472 sip_msg_shm_clone(496): 4101904 msg_lump_cloner(986): 755320 create_avp(175): 145064 register_timer(995): 216 cfg_register_ctx(47): 64 init_tcp(4962): 8192 init_tcp(4956): 32768 init_tcp(4947): 8 init_tcp(4940): 8 init_tcp(4933): 8 init_tcp(4927): 8 init_tcp(4915): 8 init_avps(90): 8 init_avps(89): 8 init_dst_blacklist(438): 16384 init_dst_blacklist(430): 8 timer_alloc(498): 96 init_dns_cache(361): 8 init_dns_cache(352): 16384 init_dns_cache(344): 16 init_dns_cache(336): 8 init_timer(267): 8 init_timer(266): 16384 init_timer(265): 8 init_timer(264): 8 init_timer(253): 8 init_timer(221): 8 init_timer(210): 278544 init_timer(209): 8 init_timer(197): 8 cfg_child_cb_new(829): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 ksr_shutdown_phase_init(120): 8 qm_shm_lock_init(1202): 8 Total: 6424576 } Module: permissions { domain_name_table_insert(978): 96 addr_hash_table_insert(458): 2776 subnet_table_insert(670): 336 init_addresses(526): 8 new_domain_name_table(865): 2048 init_addresses(512): 8 new_subnet_table(635): 57456 init_addresses(498): 8 new_addr_hash_table(418): 2048 init_trusted(220): 8 new_hash_table(98): 2048 Total: 66840 } Module: auth { Total: 0 } Module: dialog { dlg_update_cseq(636): 456 dlg_get_iuid_shm_clone(308): 4688 set_dlg_profile(580): 139680 dlg_set_leg_info(553): 64608 dlg_set_leg_info(544): 16296 build_new_dlg(469): 356616 dlg_set_leg_info(536): 29176 new_dlg_var(127): 624976 new_dlg_var(108): 1675696 init_dlg_table(297): 131088 init_dlg_table(289): 8 init_dlg_table(282): 8 init_dlg_table(277): 8 init_dlg_timer(57): 8 init_dlg_timer(48): 32 new_dlg_profile(188): 2272 new_dlg_var(117): 642216 dlg_set_leg_info(562): 18208 init_dlg_callback(55): 32 register_dlgcb(124): 189728 Total: 3895800 } Module: acc { Total: 0 } Module: xlog { Total: 0 } Module: dmq { alloc_job_queue(255): 64 shm_str_dup(802): 280 build_dmq_node(164): 3616 add_peer(68): 168 mod_init(236): 8 mod_init(229): 48 init_dmq_node_list(70): 24 init_peer_list(33): 24 Total: 4232 } Module: rr { Total: 0 } Module: jsonrpcs { Total: 0 } Module: sqlops { Total: 0 } Module: db_mysql { Total: 0 } Module: phonenum { Total: 0 } Module: pv { add_shvar(177): 8 add_shvar(170): 64 lock_set_alloc(66): 80 Total: 152 } Module: textops { Total: 0 } Module: tmx { tmx_check_pretran(271): 6280 tmx_check_pretran(250): 3320 tmx_init_pretran_table(90): 512 Total: 10112 } Module: rtpengine { rtpengine_hash_table_init(111): 14336 rtpengine_hash_table_init(96): 2048 rtpengine_hash_table_init(85): 1024 rtpengine_hash_table_init(76): 2048 rtpengine_hash_table_init(67): 2048 rtpengine_hash_table_init(58): 32 add_rtpengine_socks(978): 96 add_rtpengine_socks(959): 256 get_rtpp_set(873): 8 get_rtpp_set(864): 56 mod_init(1606): 8 mod_init(1599): 24 mod_init(1586): 8 mod_init(1579): 8 mod_init(1565): 16 rtpp_function_call(2657): 1367600 shm_str_dup(802): 2205800 Total: 3595416 } Module: htable { ht_init_tables(394): 2574336 ht_cell_new(186): 68875496 ht_add_table(315): 256 ht_add_table(289): 8424 Total: 71458512 } Module: textopsx { Total: 0 } Module: avpops { Total: 0 } Module: ndb_redis { Total: 0 } Module: sl { init_sl_stats_child(125): 9792 sl_startup(90): 8 init_sl_stats(110): 8 Total: 9808 } Module: siptrace { mod_init(320): 8 Total: 8 } Module: tm { cancel_reason_pack(1224): 104 update_totag_set(321): 656 build_local_reparse(328): 28384 update_totag_set(320): 864 _reply_light(500): 21480 init_tm_stats_child(56): 13056 build_cell(334): 8250680 relay_reply(2064): 439848 build_uac_req(1597): 118672 insert_tmcb(135): 49272 init_tmcb_lists(72): 16 init_tmcb_lists(70): 16 init_hash_table(482): 2097152 tm_rpc_response_list_init(77): 16 Total: 11020216 } Module: kex { pkg_proc_stats_init(79): 2448 Total: 2448 } Module: topos { lock_set_alloc(66): 2064 Total: 2064 } Module: mtree { mt_add_to_tree(229): 2656 mod_init(247): 8 mt_add_to_tree(255): 9559984 mt_add_to_tree(291): 811224 mt_add_to_tree(301): 269848 mt_init_tree(149): 152 mt_init_tree(136): 256 mt_init_tree(125): 176 mt_init_tree(114): 2376 mt_table_spec(840): 8 Total: 10646688 } Module: cfgutils { mod_init(898): 32 mod_init(892): 8 mod_init(884): 8 Total: 48 } Module: dispatcher { mod_init(484): 80 reindex_dests(723): 7680 ds_avl_insert(3621): 11984 ds_set_attrs(301): 960 pack_dest(411): 1456 ds_init_data(262): 16 ds_init_data(254): 16 ds_parse_reply_codes(1112): 808 mod_init(339): 8 mod_init(337): 8 ds_ping_active_init(138): 8 Total: 23024 } Module: outbound { mod_init(99): 24 Total: 24 }

2 1

Kamailio v5.4.5 Released
by Daniel-Constantin Mierla 02 Jun '21

02 Jun '21

Hello, Kamailio SIP Server v5.4.6 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.6. Note that 5.4 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.5, with v5.5.0 being release out of it. For more details about version 5.4.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/06/kamailio-v5-4-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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 Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

Releasing Kamailio v5.4.6
by Daniel-Constantin Mierla 02 Jun '21

02 Jun '21

Hello, I am considering to release Kamailio v5.4.6 sometime next week, likely on Wednesday or Thursday (June 2 or 3, 2021). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

IMS IPSec does use UDP socket?
by H Yavari 01 Jun '21

01 Jun '21

Hi to all, I am interested in the implementation of IPsec module. When I run Kamailio, it is creating processes based on the below formulate: UDP children * IPSEC_MAX_CONN * interfaces (v4 and v6) + (TCP+UD) processes for each interface. For example: children = 16tcp_children=16 IPSEC_MAX_CONN = 25and two interface : IPv6 and V4 Kamailio process count = (16 * 25 * 2 ) + 16 + 16 = 832 But why IPsec module is listening to UDP not TCP? this is an internal connection? Thank you. Regards,Hossein

2 2

SDP body, add maxptime attribut
by Victor Velo 01 Jun '21

01 Jun '21

Hello, I would like to add the maxptime attribut to the sdp body. How and where should I do insertion? This kamailio is also using RTPEngine, so is it better to do the insertion using RTPEngine? And the source of the SDP is actually FreeSwitch :), maybe there to tweak? Of course, I don't know either way of doing it so please advise. Thanks, Victor

2 2

New MariaDB server "limiting" Kamailio CPS
by Stephen Lokker 01 Jun '21

01 Jun '21

Good afternoon All, Im hoping for some advise or direction with the below. We are running a Kamailio SBC used for high call volumes and recently decided to upgrade the Database servers as the existing servers were struggling. Both the old and new databases are using MariaDB(version details all below), the problem we face is that since migrating to the new database there appears to a limitation to the amount of DB connections that can be processed or accepted. The SBC functions normally until the CPS starts to exceed around 100cps at which point the SBC stops responding to invites(in bursts which resembles flood prevention). We have checked both the Kamailio logs and the MySQL logs on the DB, both log files have no indication of errors that would cause the problem experienced(the logs attached are seen when the SBC is cut back to use the old DB, not sure they are related to the fault), the connections/processes in mysql do not drastically increase or get stuck either. The fault points to a db connection limit of some kind but all users have no limitations set. If anyone has experienced a similar problem or has any advise it would be greatly appreciated. Some info on the Kamialio and DB versions New DB root@voice-jhb-db01:~# mysql --version mysql Ver 15.1 Distrib 10.3.27-MariaDB, for debian-linux-gnu (x86_64) using readline 5.2 Old DB [root@isadb01 ~]# mysql --version mysql Ver 15.1 Distrib 10.0.36-MariaDB, for Linux (x86_64) using readline 5.1 Kamailio SBC [root@sbc01 ~]# kamailio -v version: kamailio 5.1.4 (x86_64/linux) Kind regards, Stephen Voice Engineer Tel: 2404 Fax: 3468 Email:

3 3

How To setup Provider SIP trunk with registration
by zak ast 31 May '21

31 May '21

Hi, I know I can you uacreg to register a SIP trunk as a SIP client with username/password. But how can I set up a SIP trunk as a provider and get the other end (SIP client) to register with username/password. Regards, Zak.

2 1

Architecture - your opinion is welcome and your experience too
by Daian Conrad 28 May '21

28 May '21

Hi all, I am studying and improving my understanding of how kamailio works, I have used asterisk for a few years(and starting learning freeswitch too), and I use an application to manage extensions, trunks and other media services. I'm building a new scenario where kamailio is facing the internet and the asterisk(s) are internally on private networks, would like the opinion of you who are more experienced with more complex environments than me. I cannot change the application that uses asterisk for now, due to the effort to develop event monitoring, cdr and other features that I use today. I imagined and working on making kamailio functional by saving the registration of extensions with location on DB, and with the UAC module making the registration of extensions on asterisk(s), replacing the registration address with the address of kamailio, its functional at this point, but data replication with extensions turns things some hard to mantain manually. I think about using the DNS domain for each asterisk and make this forwarded, each asterisk response for a fqdn and its extensions, like ( pbx1.example.com is forwarded to asterisk1, pbx2.example.com to asterisk2) and so on. Read about the dispatcher, rr, htable,carrierroute module to identify the domain and forward based on that , none made me sure to be chosen for the role, however, everyone has the resources to do it. I am very wrong to follow this path, which option in your opinion is "less" painful for a beginner apprentice like me? I think about simple proxy based on domain requests(all messages and dialogs) to asterisk where is responsible for the domain(realm) but, loss the ability to use great security features of kamailio, is what I understand now with the knowledge I have. Sorry and forgive my english, i'm not very good at writing. -- ================ **Daian Conrad** E-mail: daian.conrad(a)gmail.com OpenS Team (DaCoD) Linux user: #248912

6 11

No Audio For Outbound Calls
by Kashish Raheja 28 May '21

28 May '21

Hi All, I have set up Kamailio in the following manner: Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk) ---> Asterisk Server (on Cloud having public IP) I am successfully able to route the call to Asterisk server on Cloud when I make a call to the number provided by the carrier and there is audio also on both sides. However, when I am making an outbound call from Asterisk server to the number through Kamailio, there is no audio when I pick up the call. I have tried to capture the traces but not able to understand the exact problem here. Note: I am running the RTP proxy on Kamailio server. Any help on why this might be happening? Thanks. Regards Kashish +919413745250

4 15

kamailio: split initialized call to two (three or four or five) numbers
by Фёдор Измайлов 28 May '21

28 May '21

Hello, I have question regarding of call forking on kamailio. Maybe somebody can help me. For example I have table like +--------------+---------------+--------------+---------------+--------------+ | First number | Second number | Third number | Fourth number | Fifth number | +--------------+---------------+--------------+---------------+--------------+ | A number | B number | C number | --- | --- | +--------------+---------------+--------------+---------------+--------------+ Table 1: Extra_numbers or separated by comma +--------------------+-----------------------------+ | First_number | Additional_numbers | +--------------------+-----------------------------+ | Primary number | 2nd,3rd number | +--------------+-----------------------------------+ Table 2: Extra_numbers +--------------------+---------------------------------+ | First_number | Additional_numbers | +--------------------+---------------------------------+ | Primary number | 2nd number,3rd number,4 number | +--------------------+---------------------------------+ Table 3: Extra_numbers If kamailio get the call (INVITE) to some number/extension (for example, "1st number"), then in routing I request (select `Additional_numbers` from `Extra_numbers` where First_number='1 number';) from the mysql database and there I check, if "Additional_numbers" exists in the database. Then I add "Additional_numbers" to the routing, and generate INVITE to "Additional_numbers" and 1 number. (always) set r-uri ($ru) to first destination (2 number) set r-uri ($ru) to second destination (3 number) set r-uri ($ru) to Third destination (1 number) set r-uri ($ru) to n destination How could this be done? Who can help with routing? -- Best regards, Fjodor Izmajlov

3 2

Setting the prefix in Json formatted logs
by Abdirahman Osman 27 May '21

27 May '21

Hello everyone, Tried to use the log-engine to push json formatted logs, here is a what kamailio is sending, my question is, How can I set the kamailio prefix? (232) { "kamailio.level": "DEBUG", "kamailio.module": "tm", "kamailio.file": "timer.c", "kamailio.line": 634, "kamailio.function": "wait_handler", "kamailio.callid": "", "kamailio.logprefix": "", "message": "finished transaction: 0x7f232236b1c0 (p:0x7f232216ae58/n:0x7f232216ae58)" } Thank you! Abdirahman

2 2

kamailio: split initialized call to two (three or four or five) numbers
by Фёдор Измайлов 27 May '21

27 May '21

Hello, I have question regarding of call forking on kamailio. Maybe somebody can help me. For example I have table like +--------------+---------------+--------------+---------------+--------------+ | First number | Second number | Third number | Fourth number | Fifth number | +--------------+---------------+--------------+---------------+--------------+ | A number | B number | C number | --- | --- | +--------------+---------------+--------------+---------------+--------------+ Table 1: Extra_numbers or separated by comma +--------------------+-----------------------------+ | First_number | Additional_numbers | +--------------------+-----------------------------+ | Primary number | 2nd,3rd number | +--------------+-----------------------------------+ Table 2: Extra_numbers +--------------------+---------------------------------+ | First_number | Additional_numbers | +--------------------+---------------------------------+ | Primary number | 2nd number,3rd number,4 number | +--------------------+---------------------------------+ Table 3: Extra_numbers If kamailio get the call (INVITE) to some number/extension (for example, "1st number"), then in routing I request (select `Additional_numbers` from `Extra_numbers` where First_number='1 number';) from the mysql database and there I check, if "Additional_numbers" exists in the database. Then I add "Additional_numbers" to the routing, and generate INVITE to "Additional_numbers" and 1 number. (always) set r-uri ($ru) to first destination (2 number) set r-uri ($ru) to second destination (3 number) set r-uri ($ru) to Third destination (1 number) set r-uri ($ru) to n destination How could this be done? Who can help with routing? -- Best regards, Fjodor Izmajlov

1 0

Body formats for filter_body()
by Duarte Rocha 27 May '21

27 May '21

Hello all, A couple years ago the function filter_body() from textops would only work with "application/sdp" if the string on the body was "Content-Type: application/sdp\r\n". It would fail if there wasn't a space after ":" and it would fail if there was present "\r\n\r\n" instead of "\r\n". Is this the actual behaviour of the function or has it been changed to also identify SDP in the formats mentioned above? Best Regards,

2 1

Collecting stats from Kamailio.cfg using Kamailio_exporter Prometheus
by Hisham Ahmad Jan 26 May '21

26 May '21

Hi, I'm using Kamailio version 5.3.7 with DISPATCHER module and I would like to integrate my kamailio with a kamailio_exporter for monitoring using Prometheus server. Is it possible to collect data such as the number of total calls (inbound, outbound) and plot time-graphs using those stats? If so, what modules/functions would I need apart from the Dispatcher module to be able to have efficient results? Any degree of help will be appreciated. *Hisham Jan* *VoIP Engineer* *0322-6398775*

3 3

not able to display number in SCA module
by Aatif Shaikh 26 May '21

26 May '21

Hello All, I am using sca module to handle SLA presence and I am using sca.update_appearance RPC command to update the display name. kamcmd sca.update_appearance sip:215@voice.example.com 3 held KELVIN after executing the above command I am getting the KELVIN name on my Polycom phone display but I want to also update the display number so is there any way to update the display number for SLA presence using the SCA module

2 1

segfault kamailio 4.4.6 Registrar
by Jonathan Hunter 26 May '21

26 May '21

Hi Guys, We are running kamailio 4.4.6 on debian 8 and not had any issues, however we did suffer a segfault and a core was generated; May 25 09:17:06 Kam-HA-1 /usr/sbin/kamailio[9560]: ALERT: <core> [main.c:740]: handle_sigs(): child process 9575 exited by a signal 11 May 25 09:17:06 Kam-HA-1 /usr/sbin/kamailio[9560]: ALERT: <core> [main.c:743]: handle_sigs(): core was generated May 25 09:17:06 Kam-HA-1 /usr/sbin/kamailio[9560]: INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD Running gdb on the core file generated returns this output; Reading symbols from /usr/sbin/kamailio...Reading symbols from /usr/lib/debug/.build-id/bc/fbd7e44c057e9e7e0680c2a5e8f6a47290c5ba.debug...done. done. [New LWP 9575] warning: Error reading shared library list entry at 0x28250c3348 warning: Error reading shared library list entry at 0x79c0850018bf73e8 Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x0000000000589cce in lval_pvar_assign (lv=<optimized out>, rv=<optimized out>, msg=<optimized out>, h=<optimized out>) at lvalue.c:353 353 lvalue.c: No such file or directory. (gdb) bt full #0 0x0000000000589cce in lval_pvar_assign (lv=<optimized out>, rv=<optimized out>, msg=<optimized out>, h=<optimized out>) at lvalue.c:353 r_avp = 0x6 destroy_pval = -874501672 pvar = 0x0 pval = {rs = {s = 0x0, len = 0}, ri = 0, flags = 0} avp_val = {n = -874501704, s = {s = 0x7fb6cbe029b8 "\272/}", len = 5807505}, re = 0x7fb6cbe029b8} v = 6 #1 lval_assign (h=0x7fb6cbe029d8, msg=0x344, lv=0x0, rve=0x0) at lvalue.c:400 rv = 0x7fb6cbe029b8 ret = -874501672 __FUNCTION__ = "lval_assign" #2 0x0000000000000000 in ?? () No symbol table info available. Is this of any help? Could someone advise what next steps might be? Should we be looking to upgrade or is there a way to get more detail out of the core dump? Many Thanks Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

3 2

Kamailio 5.3.8 and RTPengine 9.4.1.1+0~mr9.4.1.1
by Mihai Cezar 23 May '21

23 May '21

Hi All, I've been using Kamailio as a proxy for asterisk (with public ips), everything was fine until i've changed the setup to private ips for asterisk. What I did so far: - Kamailio and rtpengine are on a box with 1 public IP. - Asterisk containers on private ips each asterisk has 2 trunks ( 1trunk voip provider 1trunk the Proxy). What works: - Proxy registrations and sip signaling works. What it does not work: - On the asterisk I got only on rtp connexion, my guess is that has something to do with the SDP, asterisk put the private ip in the SDP header. RTPengine is configured using this guide https://github.com/sipwise/rtpengine Can you have a look at the kamailio config, i think i am missing something or do something wrong. Thank you! command to start was: rtpengine --table=42 --listen-ng=127.0.0.1:2223 --interface=public/110.20.20.1 --tos=184 --pidfile=/run/rtpengine.pid --no-fallback and the Kamailio cfg is this: #!KAMAILIO #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define WITH_DEBUG #!define WITH_NAT ## This is the main configuration file for our proxy! ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 memdbg=5 memlog=5 log_stderror=yes disable_core_dump=no sip_warning=yes #!else debug=-0 log_stderror=no sip_warning=no disable_core_dump=yes #!endif log_facility=LOG_LOCAL0 fork=yes children=8 auto_aliases=no port=5060 disable_tcp=yes server_id = 1 server_signature=yes server_header="Server: proxy" dns_try_ipv6=no dns_try_naptr=no dns_retr_time=1 dns_retr_no=1 rev_dns=no ####### Modules Section ######## #loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "path.so" loadmodule "nathelper.so" loadmodule "rtpengine.so" ####### Modules Settings # ----------------- setting module-specific parameters --------------- modparam("xlog", "buf_size", 4096) modparam("xlog", "prefix", "xlog: ") modparam("xlog", "log_facility", "LOG_DAEMON") modparam("xlog", "log_colors", "L_ERR=cr") modparam("xlog", "log_colors", "L_ERR=cr;L_WARN=px") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 0) #!ifdef WITH_NAT modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@proxy") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") modparam("rtpengine", "rtpengine_retr", 2) modparam("rtpengine", "hash_table_size", 256) modparam("rtpengine", "hash_table_tout", 600) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "append_branches", 0) modparam("registrar", "max_expires", 3600) modparam("registrar", "gruu_enabled", 1) # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/plugins/dispatcher.list") modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_from", "sip:ping@proxy") modparam("dispatcher", "ds_probing_mode", 1) modparam("path", "use_received", 1) ####### Routing Logic ########### # main request routing logic request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); if(ds_is_from_list()) { setflag(FLT_FS); } # CANCEL processing if (is_method("CANCEL")) { rtpengine_delete(); if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); if (isflagset(FLT_FS)) { route(FROM_WORLD); exit; } ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting sl_send_reply("100","Trying"); } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); } route[FROM_WORLD] { record_route(); route(RELAY); exit; } route[RELAY] { if (!t_relay()) { sl_reply_error(); } if (!add_path()) { sl_send_reply("503", "Internal Path Error"); } if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (has_body("application/sdp")) { rtpengine_manage(); } append_hf("Supported: path\r\n"); exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("L_WARN","Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { rtpengine_delete(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); # exit; } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); } if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); }; route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not found"); exit; } # Dispatch requests route[DISPATCH] { # to add more servers in k8s if (!ds_select_dst("1", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT if (nat_uac_test("3")) { if (is_method("REGISTER")) { xlog("L_WARN", "natdetect fix_nated_register ip: $si\n"); fix_nated_register(); } else { xlog("L_WARN", "natdetect set_contact_alias ip: $si\n"); fix_nated_contact(); } force_rport(); } if (has_body("application/sdp") && nat_uac_test("8")) { xlog("L_WARN", "sdp fix ip: $si\n"); fix_nated_sdp("10"); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { xlog("L_WARN", "nat_uac_test 1..."); rtpengine_manage("replace-origin replace-session-connection direction=public ICE=force"); } else { xlog("L_WARN", "nat_uac_test addrtp..."); rtpengine_manage("codec-strip-all codec-transcode-PCMA codec-transcode-PCMU codec-transcode-G729 replace-origin replace-session-connection direction=public ICE=force "); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_WARN", "new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_WARN", "incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } if (has_body("application/sdp")) { rtpengine_manage(); } } onreply_route[REPLY_ROUTE] { if(status=~"2[0-9][0-9]") { rtpengine_answer("replace-session-connection replace-origin"); } if (has_body("application/sdp")) { rtpengine_offer(); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_WARN", "failure route\n"); route(NATMANAGE); if (t_is_canceled()) exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xlog("Retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

3 3

Can Kamailio help with this?
by Antony Stone 22 May '21

22 May '21

Hi. I've been aware of Kamailio, and on this list, for several years, but so far I have not implemented Kamailio for any purpose. I wonder if it could be a suitable tool for the following scenario. I currently use Asterisk to register as a client, with username and password, to other SIP PBXs in order to receive calls. As far as the other PBX is concerned, Asterisk looks like a SIP telephone on a particular extension. However, Asterisk's SIP client capabilities are limited to handling phone calls only, and it specifically cannot receive presence information about the state of other extensions from the remote PBX. So, could Kamailio be used in the path between Asterisk and the remote PBX so that the registration on the PBX comes from Kamailio (with any incoming calls being passed to Asterisk), but with Kamailio also receiving presence information from the remote PBX and making this available to some script or application (which I would expect to have to write)? The main point is that I don't want to have two things registering to the remote PBX, one for calls and one for presence, so I'm looking for a way to register with something which understands presence, and can also pass calls on to Asterisk. Can anyone suggest whether Kamailio could do this, and if so, point me at some resources to help me get started; or alternatively suggest ideas on some other tool which might be more appropriate than Kamailio? Thanks for any ideas, Antony. -- Schrödinger's rule of data integrity: the condition of any backup is unknown until a restore is attempted. Please reply to the list; please *don't* CC me.

6 11

core kamailio 5.4
by David Escartín 21 May '21

21 May '21

Dear all we have seen a core in a kamailio instance today, i attach you the bt in dropbox https://www.dropbox.com/s/3dg72cbuz7nvx5b/core_kamailio_21mayo?dl=0 https://www.dropbox.com/s/iyko32fwfxj8aam/core_kamailio_21mayo_2?dl=0 seems there might be a relation async module when delaying a request after communicating with the rtpengine ?¿ could you please take a look? thanks a lot and regards david -- <http://www.sonoc.io/> David Escartín NOC engineer *www.sonoc.io* <http://www.sonoc.io/> [image: LinkedIn] <https://www.linkedin.com/company/sonoc> [image: Twitter] <https://twitter.com/sonoc_> [image: Facebook] <https://www.facebook.com/SONOCio/> This e-mail is for the exclusive use of its recipients and may contain business secrets or other confidential or privileged information. Any unauthorised use, copying, publication or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please inform us immediately by replying to this e-mail and delete it, including any attachments or copies on your system. In accordance with the GDPR (EU) 2016/679 and the LOPDGDD 3/2018, we inform you that this e-mail address and/or any other personal data you have provided us with will be treated by SONOC with absolute confidentiality and with the only purpose of providing you with the requested services, due to your condition as a client, supplier or because you have requested information from us at any time. These data will only be kept for as long as required to comply with legal obligations. You can exercise your rights at any time by sending an e-mail to: *dataprotection(a)sonoc.io* <dataprotection(a)sonoc.io>*.* <dataprotection(a)sonoc.io>

2 2

Kamailio Stop Processing Call
by M Arqum CH 21 May '21

21 May '21

Hello Members, I have kamailio setup backed with mysql db, using db_cluster module for spliting read and write.. Lisiting on three different virtual ips are bind with same/single udp port *1* for processing calls. One static IP is bind with udp port*2* to for processing/response to option messages send from keepalived external script after every few seconds and for DMQ messages from other kamailio node. General configuation details: children=16 fork=yes listen=udp:VIP:SIP_PORT1 listen=udp:VIP2:SIP_PORT1 listen=udp:VIP3:SIP_PORT1 listen=udp:StaticIP3:SIP_PORT2 listen=tcp:SIP:RPC_PORT SH_MEM=5012 PRIV_MEM=264 System Specification : 24 GB RAM, 12 CPUs Sometimes kamailio radomly stop processing the requests or give no response on new requests recevied on virtual ips and as well it did't show any error in syslog. To monitor the kamailio memory I used promethues ( https://github.com/florentchauveau/kamailio_exporter) The moment kamailio stop processing the requests it also stop responding to kamailio exporter and scrap request get timeout. i.e error log on console debian prometheus-kamailio-exporter[83440]: 2021/05/20 12:15:27 [error] cannot read header: read unix @->/tmp/kamailio__ctl: i/o timeout After getting first timeout occurrence of kamailio-exporter scrap request, kamailio processed sip messages for almost 30 seconds then completly stop replying to any message. But it keep replying to option message arrive on different port from keepalived script. Stats : Active dialog: 75 Shared memory used: 150 MB Kamailio Installed version : kamailio -v version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 compiled on 12:51:24 Sep 21 2020 with gcc 6.3.0 Is anyone else face similar issue or idea why kamailio is showing such behaviour.? Please guide me on what I should do to find and fix the issue. Thanks -- Regards M Arqum

2 2

Video transcode for doorbell
by Luke Kenny 21 May '21

21 May '21

I have a SIP video doorbell device that is only capable of sending H264 and MJPEG video. The site is using 3CX and its associated mobile app, which only supports VP8 and VP9. Without properly reading the documentation, I went ahead and set up kamailio and rtpengine to sit in between the doorbell and 3CX. I has assumed rtpengine was capable of transcoding video as well as audio, however upon proper inspection it appears that isn't the case. Is there a kamailio module that is capable of video transcoding, even if it isn't real time, or at a reduced framerate? As the application is a doorbell, it doesn't need to be perfect. Many thanks, Luke Kenny

1 0

Eternal config - Kamailio and rtpengine
by Mihai Cezar 20 May '21

20 May '21

Hi! I use lxc containers with asterisk wich their are connecting to kamailio, so far so good they work when i put the containers with public ips, but when i switched to private ip I have one-way audio. I tried to with rtpengine but no luck: rtpengine --table=42 --listen-ng=127.0.0.1:2223 --interface=56.12.19.100 --tos=184 --pidfile=/run/rtpengine.pid --no-fallback --log-level=7 --log-facility=local3 the setup looks like this: VM-KAMAILIO-(public ip 56.12.19.100 with static routes for the private IP) --> LxC(n) (with private class 192.168.0.n) The problem i think is when LxC asterisk is sending back the 200 OK from Invite it puts his private ip in the sdp filed: v=0 o=- 2171376102 3 IN IP4 192.168.0.2 s=Asterisk c=IN IP4 192.168.0.2 t=0 0 m=audio 11336 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv i've loaded the modules, but after this i am stuck.... loadmodule "nathelper.so" loadmodule "rtpengine.so" modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@proxy") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") modparam("rtpengine", "rtpengine_retr", 2) modparam("rtpengine", "hash_table_size", 256) modparam("rtpengine", "hash_table_tout", 600)

2 3

Text transformations s.select and newline char
by Martin Nyström 20 May '21

20 May '21

Hello, I am simply trying to parse a SDP body, read it line by line and save the line into a variable for further processing. It seems when using a newline \n as the delimiter it breaks down with errors. How should I treat this issue and properly managed the newline as delimiter? May 17 12:00:00 ip-10-120-120-12 kamailio[12878]: 0(12878) ERROR: pv [pv_trans.c:2326]: tr_parse_string(): invalid select transformation: select,$var(i),$var(delimeter)}!! May 17 12:00:00 ip-10-120-120-12 kamailio[12878]: 0(12878) ERROR: <core> [core/pvapi.c:1717]: tr_lookup(): error parsing [{s.select,$var(i),$var(delimeter)}] Block extracted from .cfg sdp_get("$avp(sdp)"); $avp(lines) = $(avp(sdp){line.count}); $var(delimeter) = "\n"; $var(i) = 0; while($var(i) < $avp(lines)) { # This works xlog("L_INFO", "==> Position $var(i): $(avp(sdp){s.select,$var(i),\n}) \n"); # This fails $var(l1) = $(avp(sdp){s.select,$var(i),$var(delimeter)}); # This fails $var(l2) = $(avp(sdp){s.select,$var(i),\n}); # This fails $var(l3) = $(avp(sdp){s.select,$var(i),\\n}); # Supposed to do something funny with the $var(line) here $var(i) = $var(i) + 1; } /M

3 3

5.4 dependency package question
by Andrew Chen 19 May '21

19 May '21

Hi guys, I'm trying to build a deb file of kamailio 5.4 and when doing dpkg-buidpackage it gave a bunch of dependency warnings. I was able to take care a good number of them until I ran into this: *dpkg-checkbuilddeps: **error*: Unmet build dependencies: gcj-jdk libval-dev libxmlrpc-c3-dev None of these packages are in the "apt-cache search" output. Below is a list of modules currently loaded in 5.2.5 and want to use them again: INCLUDE_MODULES="exec rtpengine topoh carrierroute statsd dialplan kex tm tmx sl rr pv maxfwd usrloc textops sdpops siputils xlog sanity ctl acc dispatcher cfgutils permissions domain presence presence_xml nathelper dialog tls htable pike db_mysql userblacklist sqlops websocket xhttp siptrace tcpops jsonrpcs rtimer" Thanks -- Andy Chen Sr. Telephony Lead Engineer 415 516 5535 (M) 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.*

4 18

Permissions sometimes does not find group
by Romain Untereiner 19 May '21

19 May '21

Hello all, I am having some trouble with Permissions module of Kamailio, regarding the allow_source_address_group() function. In 99.9% of the cases the IP of our partner is allowed as expected, but randomly it can fail (returning -1 for the same IP address that was allowed for many other calls in the meantime). I can’t understand why it fails and why only « sometimes » - I confirm this append on the same Kamailio box, with same IP and Port of the partner. In my custom logs: « X.X.X.X:YYYY is allowed through group Z » —> as expected and sometimes (like 1% of the cases): « X.X.X.X:YYYY is not allowed - allow_source_address_group returned -1» It is maybe to note that in the address table, this range is defined as /24 and not /32. Until now it never happened on /32 defined address but I don’t know if it can be correlated. Would someone have a clue on this? Versions: - kamailio 5.5.0 (x86_64/linux) d4c1a1 - debian 9 stretch - postgresql 11.7 Many thanks! Romain Untereiner Co founder at Cantoo www.cantoo.co

2 8

Kamailio as Load Balancer
by Melek Oktay 19 May '21

19 May '21

Hi, I have been investigating distributing incoming/outgoing SIP traffic to behind Kamailio SIP servers in order to scale and other advantages too!. SRV type distribution is not a good option, it is already tested. So, I am trying to deploy Kamailio as a *LoadBalancer* in Tier1 , so it distributes traffic backbone Kamilio servers in Tier 2. There is a good presentation as follows, actually I want to implement it. https://www.kamailio.org/events/2013-KamailioWorld/23-Daniel-Constantin.Mie… However, I could not find sample configurations and sample study ! Would you show me sample configurations that Kamailio uses as a Load Balancer if you have it. Current Kamailio responsibilities in my system are: 1. Registar Server 2. Presence Server 3. Sip proxy server 4. WebSocket Maybe it will be very good to separate *Registar *Server & *Presence *Server, but could not find sample configurations too. Best Regards Melek P.S. If we could collect such configurations, I would share them for community on Github.

2 4

(no subject)
by Melek Oktay 18 May '21

18 May '21

Hi, I have been investigating distributing incoming/outgoing SIP traffic to behind Kamailio SIP servers in order to scale and other advantages too!. SRV type distribution is not a good option, it is already tested. So, I am trying to deploy Kamailio as a *LoadBalancer* in Tier1 , so it distributes traffic backbone Kamilio servers in Tier 2. There is a good presentation as follows, actually I want to implement it. https://www.kamailio.org/events/2013-KamailioWorld/23-Daniel-Constantin.Mie… However, I could not find sample configurations and sample study ! Would you show me sample configurations that Kamailio uses as a Load Balancer if you have it. Current Kamailio responsibilities in my system are: 1. Registar Server 2. Presence Server 3. Sip proxy server 4. WebSocket Maybe it will be very good to separate *Registar *Server & *Presence *Server, but could not find sample configurations too. Best Regards Melek P.S. If we could collect such configurations, I would share them for community on Github.

1 0

Question about Kamailio IMS in box
by Ibrahim soumana djibo 17 May '21

17 May '21

Hello, I am new to kamailio. And I Am following the work of Franz Edler. I would like to test it in a local environment without internet connection on ubuntu. What I don't understand is, to download the packages I necessarily need the internet. Do I have to use two network interfaces for the same virtual machine in order to access the internet and make the DNS configuration locally with the second interface? Cordially, *SOUMANA** DJIBO **Ibrahim* *Ingénieur Réseaux & Télécoms* *5 Avenue Emile Baudot, 91300 Massy* +33 07 61 80 20 35 [image: Résultat de recherche d'images pour "logo paris saclay"] https://www.universite-paris-saclay.fr/

2 1

Getting MOS feedback from rtpengine
by Barry Flanagan 17 May '21

17 May '21

Hi Using Kamailio 5.4.4 on Debian Buster. I am trying to get feedback of the MOS scores at call end. However the variables are always <null>. I have the modparams set as follows: # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") modparam("rtpengine", "rtpengine_disable_tout", -1) modparam("rtpengine", "rtpengine_tout_ms", 2000) modparam("rtpengine", "rtpengine_allow_op", 1) modparam("rtpengine", "mos_min_pv", "$avp(s:mos_min)") modparam("rtpengine", "mos_max_pv", "$avp(s:mos_max)") modparam("rtpengine", "mos_average_pv", "$avp(s:mos_avg)") modparam("rtpengine", "mos_average_packetloss_pv", "$avp(s:mos_avg_packetloss)") modparam("rtpengine", "mos_average_jitter_pv", "$avp(s:mos_avg_jitter)") modparam("rtpengine", "mos_average_roundtrip_pv", "$avp(s:mos_avg_roundtrip)") I am trying to log the values from within the dialog:end event_route: event_route[dialog:end] { rtpengine_delete(); xlog("L_INFO","mos_avg=$avp(mos_avg), packetloss_avg=$avp(mos_avg_packetloss), jitter_avg=$avp(mos_avg_jitter), latency_avg=$avp(mos_avg_roundtrip), sip_last_reply=$rs - $ci\n"); return; } Am I missing something? I have tried rtpengine_query and rtpengine_manage in place of the _delete and also tried placing this in the BYE handling section of WITHIN_DIALOG but no joy. Any assistance much appreciated! -Barry -- -Barry

5 10

Hunting Kamailio.cfg for Voice termination System
by Edo 16 May '21

16 May '21

Hi, I am looking to get a complete kamailio.cfg file that I can use to set up a voice termination system. This system will only connect with peers making multiple simultaneous calls, registering strictly via ip, accounting calls to sql db, capable of multiple rate tables for the different peers, call control, CDR tool and billing capable and call routing to multiple media servers. I am willing to try to build this myself but at this time I do not even understand why I need to use "record_route" or "t_precheck_trans" -- :)). I am a lover of information and cannot find a simple enough text to help in my kamailio journey -- unlike asterisk. If you are interested please respond to me directly at edo.ekunwe(a)gmail.com and I am willing to pay for this information or guidance(if you have the patience and time to mentor me). Thanks -- ----- Nogie Ekunwe

3 4

Help to overwrite the $od, or another solution
by Victor Velo 15 May '21

15 May '21

Hello, I'm facing to some challenge in my learning process and using Kamalio as MS Teams SBC, the request I receive from MS Teams seems to be correct but I need to overwrite the Original Domain, which is the FQDN of Kamailio itself, to another domain name: INVITE sip:+15142144069@dev-sbc.openfabrik.com:5061;user=phone;transport=tls SIP/2.0Record-Route: <sip:155.138.140.233;r2=on;lr=on;ftag=3e2ca0c5d0da4335b121f205a6f20541>Record-Route: <sip:155.138.140.233:5061;transport=tls;r2=on;lr=on;ftag=3e2ca0c5d0da4335b121f205a6f20541>FROM: Victor Velo<sip:+14507001777@sip.pstnhub.microsoft.com:5061;user=phone>;tag=3e2ca0c5d0da4335b121f205a6f20541TO: <sip:+15142144069@dev-sbc.openfabrik.com:5061;user=phone> rewrite to: INVITE sip:+15142144069@dev-sbc.openfabrik.com:5061;user=phone;transport=tls SIP/2.0Record-Route: <sip:155.138.140.233;r2=on;lr=on;ftag=3e2ca0c5d0da4335b121f205a6f20541>Record-Route: <sip:155.138.140.233:5061;transport=tls;r2=on;lr=on;ftag=3e2ca0c5d0da4335b121f205a6f20541>FROM: Victor Velo<sip:+14507001777@sip.pstnhub.microsoft.com:5061;user=phone>;tag=3e2ca0c5d0da4335b121f205a6f20541TO: <sip:+15142144069@dev.myinternaldomain.openfabrik.com:5061;user=phone> What would the best why to achieve this rewriting? Thanks, Victor

2 2

loose_route() and FQDN
by Igor Olhovskiy 14 May '21

14 May '21

Hello, I saw there are some topics on this already and carefully walked through all of them, but can't solve following issue. For a reason I do need that in Record-Route header sent to endpoint would present FQDN. If it matters, it's UDP/WSS conversion done on Kamailio. So, scheme is quite simple Enpoint A ->UDP-> Kamailio ->WSS-> Endpoint B (NAT) Main issue here, that if in Record-Route headers it's FQDN, but not IP addresses, on a new transactions with a dialog (ACK on 200, PRACK, BYE), Kamailio*loose_route()* function resolves address of destination not current dialog, but actual R-URI (or itself, if R-URI is something from WebRTC world) that is not correct due to NAT. If in RR headers IP addresses are present - all is working as expected. As an example (RR with FQDN) B answers 200 SIP/2.0 200 OK Record-Route: <sip:KAMAILIO_FQDN:8089;transport=ws;r2=on;lr=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss> Record-Route: <sip:KAMAILIO_FQDN;r2=on;lr=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss> Via: SIP/2.0/UDP <A_IP_ADDRESS>:5060;received=A IP ADDRESS;rport=5060;branch=z9hG4bKPj67fb6d86-97d7-4231-995b-e54b0f62881e To: <sip:88290@<KAMAILIO_FQDN>>;tag=hvra7mj3q0 From: <sip:+XXXX7688881@<KAMAILIO_FQDN>>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb CSeq: 18326 INVITE Contact: <sip:88290@toleivu2gdbh.invalid;transport=wss> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound Content-Type: application/sdp Content-Length: 817 ACK looks like ACK sip:88290@toleivu2gdbh.invalid;transport=wss SIP/2.0 Via: SIP/2.0/UDP A_IP_ADDRESS:5060;rport;branch=z9hG4bKPj8d05548a-91ef-4332-8617-32f8eeebf8f2 From: <sip:88881@A_IP_ADDRESS>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d To: <sip:88290@KAMAILIO_FQDN>;tag=hvra7mj3q0 Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb CSeq: 18326 ACK Route: <sip:FQDN;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss> Route: <sip:FQDN:8089;transport=ws;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss> Max-Forwards: 70 User-Agent: Asterisk PBX 13.33.0 Content-Length: 0 And Kamailio on *loose_route()* loops ACK to itself. (with result of function == 1) In a case if in Record-Route/Route headers just IP address of Kamailio present, all works as expected, but it's not really behavior I want to achieve. I've tried to play with *record_route_preset("...")* specifying only WSS part (as suggested in https://skalatan.de/de/blog/kamailio-sbc-teams) with FQDN, but no luck. Also wanted to try approach using record_route_preset() function in branch route, but it was working only with first branch, not affecting others (but I assume having different RR headers across branches is not a good idea) -- Regards, Igor

4 18

Setting $conid per branch
by Marrold 14 May '21

14 May '21

Hi all, I'm currently adding a feature to our Kamailio configuration to fork calls based on user agent. To do so I'm getting the registered endpoints with reg_fetch_contacts() iterating through and matching on them, then using seturi() / append_branch() and setting the dst-uri and flags as required. However, calls are intermittently going to the wrong TCP connection, despite the logs showing the correct destination IP and port in the ONSEND route. Looking in the location table I can see each registration has a connection_id, and the debug log indicates it's finding the connection by ID: DEBUG: <core> [core/tcp_main.c:1651]: _tcpconn_find(): found connection by id: 3 Is there a way to set the conid per branch? Is it necessary? We're using kamailio 5.3.3 on Debian 10. Thanks Matthew

2 8

Possible to get the value of $DLG_lifetime before the BYE hits loose_route()
by ryan embgrets 13 May '21

13 May '21

Greetings, I would like to know if it is possible to get the value of $DLG_lifetime before the BYE hits loose_route(). opensips provides a similar function which lets you get dialog life. https://opensips.org/html/docs/modules/3.1.x/dialog.html#func_load_dialog_c… But i could not get the same from kamailio or maybe kamailio has a better way for this? Thanks for any pointer. Ryan

1 0

Bad config - you can not call 'handle_publish' function (db_url not set)
by Marrold 13 May '21

13 May '21

Hi, We're trying to upgrade from 5.3.X to 5.4.5 however kamailio will not start and produces the following error, despite the db_url being set in the config and printed in the logs: Bad config - you can not call 'handle_publish' function (db_url not set) Looking at the code this error only seems plausible if the *pres_library_mode* is set to 1 here <https://github.com/kamailio/kamailio/blob/cc2a9518207e36b7b18aef5c8714f0c21…> but we don't see the corresponding "switch to library mode" message in the debug logs which leaves me a bit stuck - I've included the grepped logs at the bottom of the message. Does anyone have any ideas what is causing this issue? I've seen a previous issue relating to interactions between presence and presence_xml which we're also using, but it looks like this was fixed in an older version. Any suggestions appreciated. # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 60) modparam("presence", "local_log_facility", "LOG_LOCAL3") modparam("presence", "max_expires", 14430) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) root@kamailio:/etc/kamailio# grep -i "presence" /var/log/kamailio/kamailio.log May 10 12:44:14 kamailio /sbin/kamailio[23729]: ERROR: presence [presence.c:656]: fixup_presence(): Bad config - you can not call 'handle_publish' function (db_url not set) May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.y:1810]: yyparse(): loading module presence.so May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:525]: load_module(): trying to load </lib64/kamailio/modules/presence.so> May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/kemi.c:2927]: sr_kemi_modules_add(): adding module: presence May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:287]: register_module(): register PV from: presence May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1796]: pp_define(): defining id: MOD_presence May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.y:1810]: yyparse(): loading module presence_xml.so May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:525]: load_module(): trying to load </lib64/kamailio/modules/presence_xml.so> May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/kemi.c:2927]: sr_kemi_modules_add(): adding module: presence_xml May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1796]: pp_define(): defining id: MOD_presence_xml May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.y:1810]: yyparse(): loading module presence_mwi.so May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:525]: load_module(): trying to load </lib64/kamailio/modules/presence_mwi.so> May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1796]: pp_define(): defining id: MOD_presence_mwi May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.y:1810]: yyparse(): loading module presence_dialoginfo.so May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:525]: load_module(): trying to load </lib64/kamailio/modules/presence_dialoginfo.so> May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1796]: pp_define(): defining id: MOD_presence_dialoginfo May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <db_url> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <db_url> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <db_update_period> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <db_update_period> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <clean_period> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <clean_period> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <local_log_facility> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <local_log_facility> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <max_expires> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <max_expires> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence_xml' matches module 'presence_xml' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <db_url> in module presence_xml [/lib64/kamailio/modules/presence_xml.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <db_url> in module presence_xml [/lib64/kamailio/modules/presence_xml.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence_xml' matches module 'presence_xml' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <force_active> in module presence_xml [/lib64/kamailio/modules/presence_xml.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <force_active> in module presence_xml [/lib64/kamailio/modules/presence_xml.so] May 10 12:44:43 kamailio kamailio: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <handle_publish> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:43 kamailio kamailio: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <handle_subscribe> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:43 kamailio kamailio: DEBUG: <core> [core/route.c:129]: route_add(): mapping routing block (0x560e73f9a440)[PRESENCE] to 11 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:873]: init_mod(): presence May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [presence.c:293]: mod_init(): db_url=mysql://kamailio:PASSWORD@localhost/kamailio (len=50 addr=0x7f3a33d4e288) May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [presence.c:337]: mod_init(): server_address parameter not set in configuration file May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [subscribe.c:2529]: restore_db_subs(): loading information from database 0 records May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:873]: init_mod(): presence_xml May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [presence_xml.c:170]: mod_init(): db_url=mysql://kamailio:PASSWORD@localhost/kamailio (len=50 addr=0x7f3a33d4ef00) May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <bind_presence> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [presence/1] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [presence.winfo/2] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: presence - len= 8 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [add_events.c:75]: xml_add_events(): added 'presence' event to presence module May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [presence.winfo/2] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [presence/1] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:188]: add_event(): Found wipeer event [presence] for event [presence.winfo] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: presence.winfo - len= 14 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [add_events.c:94]: xml_add_events(): added 'presence.winfo' event to presence module May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [xcap-diff/0] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [xcap-diff.winfo/0] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: xcap-diff - len= 9 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [add_events.c:132]: xml_add_events(): added 'xcap-diff' event to presence module May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:873]: init_mod(): presence_mwi May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <bind_presence> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [message-summary/6] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [message-summary.winfo/0] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: message-summary - len= 15 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:873]: init_mod(): presence_dialoginfo May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <bind_presence> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [dialog/5] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [dialog.winfo/0] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: dialog - len= 6 May 10 12:44:47 kamailio /sbin/kamailio[23830]: ERROR: presence [presence.c:656]: fixup_presence(): Bad config - you can not call 'handle_publish' function (db_url not set) May 10 12:44:47 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [presence_xml.c:295]: destroy(): start May 10 12:44:47 kamailio /sbin/kamailio[23830]: DEBUG: presence [subscribe.c:2433]: timer_db_update(): db_update timer May 10 12:44:47 kamailio /sbin/kamailio[23830]: DEBUG: presence [subscribe.c:2137]: update_db_subs_timer(): update_db_subs_timer: start

3 6

secsipid
by David Villasmil 12 May '21

12 May '21

Hello guys, I'm trying to use secsipid as per https://kamailio.org/docs/modules/5.5.x/modules/secsipid.html I just installed go on a new vm: the env is as follows: # env GOPATH=/root/go then: ~# go get github.com/asipto/secsipidx ~# Then: ~# cd $GOPATH/src/github.com/asipto/secsipidx/csecsipid/ bash: cd: /root/go/src/github.com/asipto/secsipidx/csecsipid/: No such file or directory it looks like it's not downloaded where the documentation says. It is however in: go |-- bin `-- pkg |-- mod | |-- cache | | `-- download | | |-- github.com | | | |-- asipto | | | | `-- secsipidx | | | | `-- @v | | | `-- google | | | `-- uuid | | | `-- @v | | `-- sumdb | | `-- sum.golang.org | | |-- lookup | | | `-- github.com | | | |-- asipto | | | `-- google | | `-- tile | | `-- 8 | | |-- 0 | | | |-- x010 | | | |-- x011 | | | `-- x016 | | | `-- 560.p | | |-- 1 | | | `-- 064.p | | `-- 2 | | `-- 000.p | `-- github.com | |-- asipto | | `-- secsipidx(a)v1.0.0 | | |-- csecsipid | | `-- secsipid | `-- google | `-- uuid(a)v1.2.0 `-- sumdb `-- sum.golang.org 38 directories But going into ~/go/pkg/mod/github.com/asipto/secsipidx(a)v1.0.0/csecsipid we have: ~/go/pkg/mod/github.com/asipto/secsipidx(a)v1.0.0/csecsipid# ls -l total 16 -r--r--r-- 1 root root 231 May 11 13:14 Makefile -r--r--r-- 1 root root 1778 May 11 13:14 csecsipid.go -r--r--r-- 1 root root 1988 May 11 13:14 libsecsipid.h -r--r--r-- 1 root root 120 May 11 13:14 secsipid.h The Makefile contains the proper liba: # Makefile to build libraries # all: libso libso: go build -o libsecsipid.so -buildmode=c-shared csecsipid.go liba: go build -o libsecsipid.a -buildmode=c-archive csecsipid.go clean: rm -f libsecsipid.so rm -f linsecsipid.a But "make liba" returns: # make liba go build -o libsecsipid.a -buildmode=c-archive csecsipid.go csecsipid.go:6:2: no required module provides package github.com/asipto/secsipidx/secsipid: go.mod file not found in current directory or any parent directory; see 'go help modules' make: *** [Makefile:10: liba] Error 1 Obviously not a "go" guy here. What am I doing wrong? Thanks for your help! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 9

bogus event with terminated dialogs with late 200ok responses
by David Escartín 12 May '21

12 May '21

Hello all using kamailio version 5.4.4, we are seeing some errors like CRITICAL: dialog [dlg_hash.c:1205]: next_state_dlg(): bogus event 3 in state 5 (with BYE) for dlg 0x7f6ca2304b48 [15135:39671] with clid '00463385_20210512_002_HPRA_1.1.4.40.sp3_7532_0_54533900281' and tags 'Out-55089-0000000CB27947F9' '170cB11c48vmj' which seem to be generated when we receive a BYE from the remote UA but we were expecting an ACK also, For example INVITE ----------> <---------------183 <---------------200 BYE -------------> (no ACK) <--------------200 <--------------200 .... so the b2bua (freeswitch) keeps retransmitting the 200ok, and on each 200ok retransmission we are getting this type of message. These seem to be like race conditions since 200/BYE flow almost at the same time, and as I understand, the UA should send the ACK. But my doubt is if these errors can be harmful for the kamailio or they can be ignored and address the issue to the remote UA on each case thanks a lot and regards david -- <http://www.sonoc.io/> David Escartín NOC engineer *www.sonoc.io* <http://www.sonoc.io/> [image: LinkedIn] <https://www.linkedin.com/company/sonoc> [image: Twitter] <https://twitter.com/sonoc_> [image: Facebook] <https://www.facebook.com/SONOCio/> This e-mail is for the exclusive use of its recipients and may contain business secrets or other confidential or privileged information. Any unauthorised use, copying, publication or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please inform us immediately by replying to this e-mail and delete it, including any attachments or copies on your system. In accordance with the GDPR (EU) 2016/679 and the LOPDGDD 3/2018, we inform you that this e-mail address and/or any other personal data you have provided us with will be treated by SONOC with absolute confidentiality and with the only purpose of providing you with the requested services, due to your condition as a client, supplier or because you have requested information from us at any time. These data will only be kept for as long as required to comply with legal obligations. You can exercise your rights at any time by sending an e-mail to: *dataprotection(a)sonoc.io* <dataprotection(a)sonoc.io>*.* <dataprotection(a)sonoc.io>

2 1

Kamailio as front proxy for multiple sip servers
by Eliphas Levy Theodoro 10 May '21

10 May '21

Hello! I am trying to config one kamailio as reverse proxy for a bunch of internal (no internet address) separate asterisk sip instances (per domain). The kamailio server would be the only with the valid IP address, so would use rtpengine to force to be in the media path. Like this scenario: https://opensips.org/pipermail/users/2020-August/043610.html I have used as starting point this very basic config: https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-reques… Basically just added rtpproxy support, and voilà, inter-SIP is working, media always passing into the proxy. The problem: I would have WebRTC phones connecting too. I tried setting WSS up in kamailio, and asterisk (pjsip) wouldn't know how to send the message to the proxy: on register it has trasnport=wss in the contact: header, looks like it is confusing the asterisk. So, I resort for the wisdom of the list :) What would be the good-best-path to take here, hack the header, or put the webphones registering directly on the asterisks (with a nginx reverse proxy maybe)? Someone must have already made a blog with such setup, but I could not google-unearth it at least until now. Regards, -- Eliphas

5 7

MS Teams SBC
by Victor Velo 09 May '21

09 May '21

Hello, I'm trying to test a Henning's tutorial for Teams SBC described here: https://skalatan.de/en/blog/kamailio-sbc-teams. A minor change at the MS Teams setup is just about the login. Now it requires MFA.The license I used is : Microsoft 365 developer subscriptions (E5).The domanin dv-sbc.openabrik.com is added + 1 user PS> Connect-MicrosoftTeams (MFA)PS> New-CsOnlinePSTNGateway -Fqdn dev-sbc.openfabrik.com -SipSignalingPort 5061 -MaxConcurrentSessions 10 -ForwardCallHistory $true -Enabled $true Everything seems to be OK at MS Teams side. I'm using the Kamailio coming with Kazoo. Version 5.2.5[root@dev-sbc kamailio]# kamailio -versionversion: kamailio 5.2.5 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: unknown compiled on 01:14:59 Nov 19 2019 with gcc 7.3.1 When Kamailio is probing MS Teams, the log shows the following errors (for the 3 GW): hG4bKc188.6a606040000000000000000000000000.0#015#012To: <sip:sip2.pstnhub.microsoft.com;transport=tls>#015#012From: <sip:dev-sbc.openfabrik.com>;tag=d9a52ca041d9e6253dd2a7d5399056e3-2f56#015#012CSeq: 10 OPTIONS#015#012Call-ID: 4ed99b8a73a96dcd-10234(a)155.138.140.233#015#012Max-Forwards: 70#015#012Content-Length: 0May 9 01:26:17 dev-sbc kamailio[10234]: ERROR: <core> [core/tcp_main.c:618]: _wbufq_add(): (517 bytes): write queue full or timeout (0, total 1100374016, last write 41683154 s ago)May 9 01:26:17 dev-sbc kamailio[10234]: ERROR: <core> [core/tcp_main.c:2673]: tcpconn_1st_send(): 52.114.76.76:5061: EAGAIN and write queue full or failed for 0x7fce434db860May 9 01:26:17 dev-sbc kamailio[10234]: ERROR: tm [../../core/forward.h:293]: msg_send_buffer(): tcp_send failedMay 9 01:26:17 dev-sbc kamailio[10234]: ERROR: tm [uac.c:648]: send_prepared_request_impl(): Attempt to send to precreated request failedMay 9 01:26:17 dev-sbc kamailio[10234]: INFO: <script>: Sent out tm request to Teams: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0#015#012Via: SIP/2.0/TLS 155.138.140.233:5061;branc config: tls.cfg [server:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/openfabrik.com/privkey.pemcertificate = /etc/letsencrypt/live/openfabrik.com/fullchain.pemca_list = /etc/letsencrypt/live/openfabrik.com/bc2025.pemserver_name = dev-sbc.openfabrik.com [client:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/openfabrik.com/privkey.pemcertificate = /etc/letsencrypt/live/openfabrik.com/fullchain.pemca_list = /etc/letsencrypt/live/openfabrik.com/bc2025.pem ==== event_route[tm:local-request]{ setflag(FLAG_LOCAL_REQUEST); if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:dev-sbc.openfabrik.com:5061;transport=tls>\r\n"); xlog("L_INFO", "Sent out tm request to Teams: $mb\n"); } xlog("L_DEBUG", "$ci|local|start $pr request $rm $ou\n"); xlog("L_DEBUG", "$ci|local|source $si:$sp -> $dd:$dp\n"); xlog("L_DEBUG", "$ci|local|from $fu\n"); xlog("L_DEBUG", "$ci|local|to $tu\n"); #!ifdef SIP_TRACE_ROLE route(SIP_TRACE); #!endif #!ifdef PRESENCE_ROLE route(PRESENCE_LOCAL_REQUEST); #!endif } ====# setid(integer) destination(sip uri) flags (integer, optional), priority(int,opt), attrs (str,optional)kamctl dispatcher add 2 'sip:sip.pstnhub.microsoft.com;transport=tls' 0 3 'socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com'kamctl dispatcher add 3 'sip:sip2.pstnhub.microsoft.com;transport=tls' 0 2 'socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com'kamctl dispatcher add 4 'sip:sip3.pstnhub.microsoft.com;transport=tls' 0 1 'socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com' [root@dev-sbc kamailio]# kamctl dispatcher showdispatcher gatewayssqlite_query: /usr/bin/sqlite3 /etc/kazoo/kamailio/db/kazoo.db select * FROM dispatcher ORDER BY setid; 1|1|sip:155.138.140.233:11000|9|0|zone=local;profile=sipinterface_1;duid=243;node=freeswitch@dev-sbc.openfabrik.com|added by nodes role2|2|sip:sip.pstnhub.microsoft.com;transport=tls|0|3|socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com|3|3|sip:sip2.pstnhub.microsoft.com;transport=tls|0|2|socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com|4|4|sip:sip3.pstnhub.microsoft.com;transport=tls|0|1|socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com| Is it a bug on the Kamailio 5.2.5 release or Do I miss something? Thanks, Victor

2 3

Unable to parse the Cancel request
by Ilie Soltanici 05 May '21

05 May '21

Hello, We are having an issue here where Kamailio is complaining about duplicate To header, while in the SIP Packet there is just one. See below the log message: DEBUG: tm [t_msgbuilder.c:391]: build_local_reparse(): duplicate To header ERROR: tm [t_msgbuilder.c:531]: build_local_reparse(): cannot build CANCEL request ERROR: tm [t_cancel.c:297]: cancel_branch(): attempt to build a CANCEL failed ERROR: tm [t_fwd.c:1434]: e2e_cancel(): cancel error And this is the sip message: CANCEL sip:123456789@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:19373 ;branch=z9hG4bK-524287-1---7774073fad1a3264;rport Max-Forwards: 70 To: <sip:123456789@1.2.3.4:5060> From: "101"<sip:101@1.2.3.4:5060>;tag=6f373b66 Call-ID: hC2O6zx8ZaUJ1di046 CSeq: 2 CANCEL Proxy-Authorization: Digest username="101",realm="1.2.3.4",nonce="YJKbfWCSmlFTHxdJfKmySydF9L7FzfaqcgyOo4A=",uri=" sip:123456789@1.2.3.4:5060 ",response="2bf2ba8ef2374d6aa8c39d650613077d",algorithm=MD5 User-Agent: TEST Content-Length: 0 version: kamailio 5.5.0 (x86_64/linux) d4c1a1 The route itself, is like in the standard documentation: # CANCEL processing if KSR.is_CANCEL() : KSR.info(RANDOM_LOG + 'KSR.is_CANCEL') if KSR.tm.t_check_trans()>0 : self.ksr_route_relay(msg) return 1 def ksr_route_relay(self, msg): if KSR.is_method_in("IBSU") : if KSR.tm.t_is_set("branch_route")<0 : KSR.tm.t_on_branch("ksr_branch_manage") if KSR.is_method_in("ISU") : if KSR.tm.t_is_set("onreply_route")<0 : KSR.tm.t_on_reply("ksr_onreply_manage") if KSR.is_INVITE() : if KSR.tm.t_is_set("failure_route")<0 : KSR.tm.t_on_failure("ksr_failure_manage") if KSR.tm.t_relay()<0 : KSR.sl.sl_reply_error() return -255 Thanks

2 7

start_recording and stop_recording inside event_route[xhttp:request]
by Володимир Іванець 05 May '21

05 May '21

Hello! I'm testing call recording with Rtpengine. It works fine when the "record-call=on" flag is added to the *rtpengine_offer* or *start_recording* is used in the *request_route*. But I was wondering if the call recording can be managed by a separate application. So I add the following lines to the *event_route[xhttp:request]* and triggered it with an HTTP request after the call was established. The call-id value was taken from the Rtpengine log and sent with the request. * if ($hu =~ "^/CALL_RECORD_START/") {* * $var(call_id) = *"call-id=" + *$(hu{s.select,2,/});* xlog("L_DBG", "$var(call_id)"); * start_recording($var(call_id));* exit; * }* In this case, Kamailio will log errors and will not send the Rtpengine command to start call recording. *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: DEBUG: <script>: call-id=3de74068278ede4417086eb82bdf5a06@127.0.0.1:5080.0* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: ERROR: rtpengine [rtpengine_funcs.c:294]: get_callid(): call-id not found* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: ERROR: rtpengine [rtpengine.c:2383]: rtpp_function_call(): can't get Call-Id field* Have anyone tried to call functions *start_recording* and *stop_recording* inside *event_route[xhttp:request]*? Thanks a lot!

3 12

Kamailio v5.5.0 Released - new major version is out
by Daniel-Constantin Mierla 05 May '21

05 May '21

Kamailio v5.5.0 is out – it comes with 6 new modules and a considerable set of improvements touching again more than 80 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-5-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.5.0 brings more flexibility and optimizations across many components, enhancements to distributed message queue (dmq module), load balancer, http client, STIR/SHAKEN support, IMS/VoLTE, rtpengine and tls as well as to make available more functions to KEMI interface ... just to list only a few here. Enjoy Kamailio v5.5.0! Thank you for flying Kamailio! Stay safe and healthy! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

dialog OPTIONS
by David Villasmil 05 May '21

05 May '21

Hello guys, I'm using the dialog module, params are: # ----- dialog params ----- modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", 20) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) modparam("dialog", "ka_failed_limit", 2) modparam("dialog", "send_bye", 1) modparam("dialog", "timer_procs", 1) When receiving an INVITE, i'm doing dlg_set_property("ka-src"); dlg_set_property("ka-dst"); dlg_set_property("timeout-noreset"); I have a couple questions: The flow is the following: clientA>-\ kamailio<->freeswitch clientB<-/ Kamailio receives an INVITE from ClientA, which I forward to freeswitch, which in turn forwards it back to Kamailio. In reality I would like to monitor only the client legs, but I'm having problems, so I'm enabling OPTIONS on all legs. On the ClientA legs, I see OPTIONS going from kamailio to both freeswitch and the client. But on the ClientB legs i don't see any OPTIONS at all. Can someone point me in the right direction to enable it on all legs? Something else i'm seeing is even though i set "ka_failed_limit" to 2, kamailio is sending several OPTIONS before sending BYEs to both legs (clientA, since there are no OPTIONS on clientB) Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

The plan for releasing Kamailio v5.5.0
by Daniel-Constantin Mierla 05 May '21

05 May '21

Hello, after almost 4 weeks since development was frozen, it is time to plan releasing the next major stable version, respectively 5.5.0, therefore I propose to do it on Wednesday, May 5, 2021. The branch 5.5 was already created about one week ago, several doc resources were published (core/variables/transformations/rpcs/stats cookbooks, alphabetic indexes, ...). If anyone is testing an upgrade from branch 5.4 to 5.5 and encounters changes that should be done in Kamailio config, add notes about at: * https://www.kamailio.org/wiki/install/upgrade/5.4.x-to-5.5.0 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

possible bug on {param.value,name}
by David Villasmil 04 May '21

04 May '21

Hello guys, I'm possible using this wrong, but here it is; I need to get a previously set parameter to rr, named "wsconnid": $var(wsconnid) = $(hdr(Route){param.value,wsconnid}) xlog("L_ERR", "Connection ID parameter: $var(wsconnid)"); And i'm getting: May 3 19:24:45 ip-10-231-32-37 /usr/sbin/kamailio[10015]: ERROR: <script>: WS Connection ID 13> May 3 19:24:49 ip-10-231-32-37 /usr/sbin/kamailio[10014]: ERROR: <script>: Got an OPTIONS We now need to get the connection id from rr This is an example: Route: <sip:10.231.32.37;r2=on;lr;wsconnid=13> The function is returning "13>" instead of "13". Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 8

geting $conid on outbound INVITE
by David Villasmil 04 May '21

04 May '21

Hello guys, I don't have a lot of experience with webrtc (websocket). I'm trying to get $conid when i receive an INVITE via UDP which is GOING to a ws, how do i do that? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Re: [SR-Users] Kamailio 5.4.4 and Siptrace
by pwer spire 04 May '21

04 May '21

Hi, Any one have some idea about this? Or how it should be configured in order to still have the original response received in kamailio instead of the mapping one? Thanks in advance, Regards, Tiago

3 7

Can't make tcp_conid_alive work with $var
by David Villasmil 03 May '21

03 May '21

Hello guys, I'm trying to do the following: On INVITE, i add a param to RR "wsconnid" which is the connection id of the WS socket. Afterwards, i get an OPTIONS including said param. Next I want to retrieve the param and check if the connection is alive. NOTE: using simply "$(hdr(Route){param.value,wsconnid})" returns an extra ">" from the end of the string, and i haven't been able to remove that any other way but with "striptail", it works fine. *** $var(conid) = $(hdr(Route){param.value,wsconnid}); $var(conid) = $(var(conid){s.striptail,1}); xlog("L_ERR", "Connection ID parameter: [$var(conid)]"); if (tcp_conid_alive("$var(conid)")) { xlog("L_ERR", "WS Connection ID $var(conid) is alive"); } else { xlog("L_ERR", "WS Connection ID $var(conid) is NOT alive"); } *** With that, "$var(conid)" contains the connection id properly: *** ERROR: <script>: Connection ID parameter: [2] *** but tcp_conid_alive complaints: *** ERROR: <core> [core/sr_module.c:1638]: get_int_fparam(): Could not convert PV to int ERROR: tcpops [tcpops_mod.c:374]: w_tcp_conid_state(): invalid parameter 'conid' (must be a number) *** Using a hardcoded number such as "1" generates no complaints. Kamailio version: version: kamailio 5.3.7 (x86_64/linux) Not sure what's going on here, any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

named account with multiple numbers
by beer Ll 03 May '21

03 May '21

Hi I have a configuration with sip accounts which authenticate with the real name of the customers ex: customer1(a)sip.myvoip.dev customer2(a)sip.myvoip.dev ... INBOUND call of multiple numbers are connected with alias_db to the accounts ex: +331234567890 | sip.myvoip.dev | customer1 | sip.myvoip.dev | +331234567891 | sip.myvoip.dev | customer1 | sip.myvoip.dev | +341234567893 | sip.myvoip.dev | customer2 | sip.myvoip.dev | +341234567894 | sip.myvoip.dev | customer2 | sip.myvoip.dev | ... and all works perfectly. OUTBOUND call after authentication with the username kamailio change the caller id and the ip of my ISP-SIP-TRUNK to relay the call with the correct CID for the ISP using the variable $fn (display name in the From header) [SIP-GW-ISP] { #out with ISP route(CID); $ru = "sip:" + $rU + "@" + "IP_ADDRESS_MY-ISP"; route(RELAY); exit; } ROUTE[CID] { $var(caller) = $(fn{re.subst,/"//g}); uac_replace_from("$var(caller)","sip:$var(caller)@$fd"); } Is possible do it with a different approach for example using Remote-Party-ID , PPI P-Preferred-Identity ? Is possible to check if the customer uses the correct outbound number present in the alias_db table ? ex for customer1 only +331234567890 or +331234567891 numbers many thanks Luca

1 0

connect multiple asterisk to kamailio
by Vahid sadrnia 03 May '21

03 May '21

hi everybody how can i connect multiple asterisk to kamailio and user just connect to specific asterisk(not load balancing),in fact each customer have their own media gateway(asterisk) and use kamailio for topolgy hiding. by regards and thanks.

2 1

Set $uac_req(body) to raw hex value
by Nick 03 May '21

03 May '21

Hi all, It seems in the UAC module the $uac_req(body) PV is evaluated as a string, How can I set the value of the request body of a generated request, to a raw hex value without it being interpreted as a string? Cheers, Nick

2 3

new in Kamailio 5.5.x document updated
by Henning Westerholt 03 May '21

03 May '21

Hello, the "new in Kamailio 5.5.x" document has been updated. Again, a really impressive set of changes that lead to this release: Over 1350 commits have been done in the last development period from over 70 individual contributors. You find the document at the usual place in our wiki: https://www.kamailio.org/wiki/features/new-in-5.5.x Many thanks to everybody who contributed to Kamailio. Help in updating the documentation (e.g. core cookbook, pseudo-variable and transformation section) in the wiki is really welcome, as some content is probably still missing there. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

2 1

potential PRACK related topos bug [solved]
by Juha Heinanen 03 May '21

03 May '21

Thanks to Daniel who patched topos/topos_redis code, topos now works also in situations where requests spiral through the same proxy twice, for example: A - P1 - P2 - P1 - B. -- Juha

2 1

cannot xlog messages into /var/log/syslog
by Abdulaziz Alghosh 02 May '21

02 May '21

Hi Everyone, i am a beginner with Kamaili Server and I am trying to get along with Kamailio config file by logging some messages by xlog Module. xlog module logs two already-written messages which had been delivered with the default configuration file with <script> as a prefix. Unfortunately, I am not able to log anything further neither with xlog() nor xdbg() functions. may someone please advise? Best regards. Abdul

2 1

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2004

2003

2002

sr-users