sr-users

sr-users@lists.kamailio.org

36 participants
37545 discussions

Kamilio Presence
by Jerry Kendall 20 Sep '21

20 Sep '21

Hey Bill, Just checking to see if you managed to get the BLFs to work on the Yealinks... I've Grandstream/Yealink/Polycom devices and can't get BLFs to work - the NOTIFY is not being sent. regards, Jerry __________________________________________________________ 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

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

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sr-users