- sr-users - kamailio.org

Use Kamailio and RTPEngine to play wav file as early media to announce a server maintenance
by jhayhoe＠liquidweb.com 15 Mar '23

15 Mar '23

I am looking to configure Kamailio and RTPEngine to play a wav file for any new calls during a scheduled asterisk server maintenance. I have set up the following within route{} to send the call to a early_media route based on a variable that I can set on the server using kamcmd. route { ... if ($sht(config=>maint)=="1") { if (is_method("INVITE") && !has_totag()) { xlog ("L_INFO","Ongoing maintenance. Call-ID:$ci | Method: $rm"); route(early_media); } } ... } route[early_media] { xlog("Maintenance: handling early_media\n"); rtpengine_manage("direction=external direction=external"); $var(body) = $(rb{re.subst,/(IP4.).*/\1##server IP redacted##/g}); set_reply_body($var(body), "application/sdp"); t_send_reply("183", "Session Progress"); sleep(2); unblock_dtmf("all"); play_media("file=/etc/kamailio/media/maintenance.wav"); sleep(21); rtpengine_delete(); t_reply("480","Temporarily Unavailable"); } This mostly works as I hear the recorded message in place of the ringing tone when a call is processed by kamailio and then a busy signal is heard as part of the 480 reply at the end. The sleep's are in place to match the duration of the recorded message as otherwise the 480 is sent immediately following the start of the play_media. Feel free to let me know if there is a better solution to wait the duration of the media file and then proceed with processing instead of using sleep(). Is there a way to instead send a CANCEL or a BYE instead of a 480 after the wav file is played? I tried using t_cancel_callid("$ci", "$cs", "0") as well as a dlg_bye("all") at the end of the early_media route and neither seemed to send any headers out.

3 3

stuck with webrtc on kamailio
by Arun K R 14 Mar '23

14 Mar '23

hello, I have issue of no voice on kamailio configured for webrtc. I have configured rtpengine with public and private interface. My kamilio has two interface of pubic and private ip. Somebody has suggested me that there is no ice trickling happens. I am using jssip webrtc client. Kindly suggest me or help me with this. I have send another email for same issue to this group and didnt get any reply. Kindly help me

1 0

Re: Your message to sr-users@lists.kamailio.org awaits moderator approval
by Arun K R 14 Mar '23

14 Mar '23

I need a help on issue with rtpengine configuration and I didn't get any repy for it. Kindly check about it and help me. If i won't get any help kindly let me know, so I can stop waiting for your reply On Mon, Mar 13, 2023 at 12:50 PM <sr-users-bounces(a)lists.kamailio.org> wrote: > Your mail to 'sr-users(a)lists.kamailio.org' with the subject > > webrtc audio issue > > Is being held until the list moderator can review it for approval. > > The message is being held because: > > The message is larger than the 150 KB maximum size > > Either the message will get posted to the list, or you will receive > notification of the moderator's decision. >

1 0

send messages from source ip in Linux anyIP list
by mohsen khashei 14 Mar '23

14 Mar '23

1 0

Dispatcher Incoming
by nutxase 14 Mar '23

14 Mar '23

Hi All I have setup kamailio using dispatcher to proxy registrations from the UAC to asterisk but when asterisk sends an incoming call it does not seem to keep the path header and therefore kamailio sends 404 is there anyway around this? Thanks Sent with [Proton Mail](https://proton.me/) secure email.

5 18

Use MySQL connector instead of mariadb connector
by suchendra adiga 14 Mar '23

14 Mar '23

Hi, I have been working on setting up the database with the Kamailio server. But facing a problem when my db user has configured with SSL required. But I learnt that there is no support to enable the SSL in mariadb connector. So I am thinking of using the MySQL connector instead. Is there anyway I could configure the kamailio to use MySQL instead of Mariadb? Thanks and regards, Suchendra

2 7

Advertise NAT IP address causing Multiple SIP INVITEs is that expected ?
by Muhammad Zaka 13 Mar '23

13 Mar '23

Hi We are using advertise in the listen interface for our NATed setup within AWS. Using following listen statement in kamailio: - version: kamailio 5.5.1 listen=udp:10.0.1.4:5060 advertise 3X.1XX.1XX.2XX:5060 dispatcher.list 1 sip:10.0.4.23:5060 8 10 weight=11 #FreeSwitch 1 We are seeing that kamailio (10.0.1.4) is sending two SIP INVITEs via dispatcher to FreeSwitch at 10.0.4.23 which seems strange or is it expected behaviour? (As seen below) SIP INVITE Request: - WAN Address --> Kamailio AWS NATed address --> FreeSwitch 1 1X.2X.2X.2X --> 3X.1XX.1XX.2XX 10.0.1.4 --> 10.0.4.23 10.0.1.4 --> 10.0.4.23 [cid:image001.png@01D955CE.9F0C2EE0] Many Thanks Regards Muhammad Zaka Muhammad Zaka UCaaS Solutions Engineer Tel: 03330147908 Email: Muhammad.Zaka(a)gamma.co.uk This is an email from Gamma Telecom Ltd, trading as “Gamma”. The contents of this email are confidential to the ordinary user of the email address to which it was addressed. This email is not intended to create any legal relationship. No one else may place any reliance upon it, or copy or forward all or any of it in any form (unless otherwise notified). If you receive this email in error, please accept our apologies, we would be obliged if you would telephone our postmaster on +44 (0) 808 178 9652 or email postmaster(a)gamma.co.uk Gamma Telecom Limited, a company incorporated in England and Wales, with limited liability, with registered number 04340834, and whose registered office is at The Scalpel, 18th Floor, 52 Lime Street, London EC3M 7AF and whose principal place of business is at Kings House, Kings Road West, Newbury, Berkshire, RG14 5BY.

1 0

(no subject)
by gui 10 Mar '23

10 Mar '23

Hello I have an issue with Rx in my VolTE call Flow. Please refer to picture and pcap in annex. I am using same rtp engine at origination and termination. I am sending rtpmanage for invite and 183 session progress at both origination and termination. My issue is that Rx is not sending for the proper media flow component to PCRF. At origination it is considering a media flow between A 50034 and RTPEngine 49186 (instead of considering RTP Engine Port 49152) At Termination it is considering a media flow between B 50054 and RTPEngine 49132 (instead of considering RTP EnginePort 49170) Any idea why?

3 4

use_dns_failover and dispatcher
by Lars Jansson 10 Mar '23

10 Mar '23

Hi, I had the intention to route calls to some SRV destinations via dispatcher and to other SRV destinations via DNS lookup and 'use_dns_failover=on'. Currently in a route block using dispatcher two out of three targets within the SRV destination are unavailable (by intention). During my tests to the dispatcher destination I have found that: A) If I have 'use_dns_failover=on' this setting "overrules" the dispatcher routing and the unavailable targets are also tried before DNS failover finds the available target. B) If I have 'use_dns_failover=off' the dispatcher routing logic works as expected and the call is always routed to the only available target directly, making use of the result of the dispatcher OPTIONS pinging. -- So my conclusion is now that you have to choose: Either use dispatcher routing, or use DNS failover based routing. - I'm fine with this, but I just would like to know if my conclusion is correct, or if there is actually a way to use dispacther for some SRV destinations and DNS failover for other SRV destinations. Regards, Lars

1 0

New 'reverse' transaction not using correct transport protocol
by Benoit Panizzon 10 Mar '23

10 Mar '23

Hi CPE registering with TCP (or TLS). 2nd leg is UDP. Within one transaction (INVITE to OK including all ACK and PRACK) transport is kept as desired for both legs. But when the B side disconnects (sending BYE, new reverse transaction), this is sent via UDP to the A CPE which initially talked TCP and thus ignored. record_route() is called on each leg. What could I be missing? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 6

Silo extra_hdrs duplication
by Dr. Barabás Péter 10 Mar '23

10 Mar '23

Hi, I’m using kamailio’s silo module to store offline messages in it. I set extra_hdrs modparam to e.g. : ‘Resent-from-silo’ so the clients get information that the message has arrived directly (immediatelly) from the sender or it was stored offline and comes from silo. So the problem is the following: * if receiver’s socket is broken (e.g. airplane mode on mobile) and kamailio ‘thinks’ it is registered, server tries to send the message * message will be timed out and based on config it will be stored in silo. * the next step is to check if receiver is registered or not (since the client can re-register during 30sec until the message has timed out) * If client is registered, kamailio tries to send the message immediately with m_dump(). * If the receiver’s connection is still broken, the message will be timed out and store in silo. * In every store, a new ‘resent’ extra_hdrs value is appended. When extra_hdrs length reaches 1024 bytes, the m_dump will fail and it blocks the dumping of messages to the given receiver. Question: can extra_hdrs value be removed before store? Or what can be the solution not to duplicate the extra_hdrs value in some bad network situation? Peter

3 2

Re: [sr-dev] KEMI python 'KSR' has no attribute 'sanity'
by Henning Westerholt 10 Mar '23

10 Mar '23

(adding sr-users) Hello, maybe because you're spelling it wrong? Compare e.g., to here: https://kamailio.org/docs/tutorials/5.6.x/kamailio-kemi-framework/modules/#… Cheers, Henning -----Original Message----- From: agural5(a)gmail.com <agural5(a)gmail.com> Sent: Freitag, 10. März 2023 08:32 To: sr-dev(a)lists.kamailio.org Subject: [sr-dev] KEMI python 'KSR' has no attribute 'sanity' Hello, I am trying to use KEMI KSR.sanity in python but got error: AttributeError: module 'KSR' has no attribute 'sanity' appreciate your help for solve this. Thanks! _______________________________________________ Kamailio (SER) - Development Mailing List To unsubscribe send an email to sr-dev-leave(a)lists.kamailio.org

2 1

webrtc from public internet have no voice
by mail4arunkr＠gmail.com 10 Mar '23

10 Mar '23

Hello, I have installed Kamailio 5.6 in debian 11 and RTPengine 11.3 also in the same server. I have configured kamailio to work as webrtc server and it forwards the registration to asterisk. Now when I am trying call from jssip webrtc client it reaches kamailio and route it through private interface to asterisk server. Asterisk then route it to the provider server. When i make a call asterisk server recives rtp from the provider and convert the rtp to srtp and sending back to kamailio. but there is no sound for webrtc from public internet . Also i am getting warning (SRTCP /RTP output wanted, but no crypto suite was negotiated) webrtc from Local network works fine with VPN For normal udp call without webrtc works fine. Kamailio having two interfaces, interface1 is private 10.13.1.140 and interface 2 is publi ip 100.x.x.x webrtc client sends calls to the public ip interface 100.x.x.x and kamaiio routes the call to the asterisk via private interface 10.13.1.140. asterisk server sends the call to remote server and gets the rtp back from that Asterisk server ---converts RTP to SRTP and forward to ----> kamailio 10.13.1.140 but there is nothing happens after that, please help me on this i am new to kamailio and rtpengine. The issue is only for webrtc from public internet. But when using webrtc from LAN works fine below are the logs Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: from domain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: 09496381412 has fromDomain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: Request going to srtp customied arun Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [control] Adding a proxy for control ng stats:127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'offer' from 127.0.0.1:53374 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'offer' from 127.0.0.1:53374: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=group:BUNDLE 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:9 IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:NM2h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:BF2WmB5sOZZWVIFppZ3YTiSH Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 9F:FA:18:B0:8D:17:A5:14:FE:22:BA:1E:73:1E:3E:02:6F:17:EC:B5:FC:77:BD:E1:0B:51:DF:7F:61:B0:2F:55 Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "223.190.206.123" ], "from-tag": "7n38ig1acl", "command": "offer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: NOTICE: [ui7jc53b9lfbt9lf4nu7]: [core] Creating new call Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to '' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for offerer 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30020 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30021 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30000] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [1] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'offer' from 127.0.0.1:53374 (elapsed time 0.022450 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'offer' to 127.0.0.1:53374: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30000 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30001 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:1 AEAD_AES_256_GCM inline:czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDJWi/0EDtIq16vYCOM Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:2 AEAD_AES_128_GCM inline:5abyoU5PKLg1lNfmLg1aBLIqyNAgmKNEEo8J1w Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:i4O28fJjCyvBVQmwjCb7dySy5K20yeg5n1ky+uhPNSpbqkv8vxkcIaerkO9Gbg Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:OOydiBrJ9nMu23yaBP7JuYNddpbXX8sN/V9Ab/psnTZIBRZKXenHWVPvJSM9Fw Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:5 AES_192_CM_HMAC_SHA1_80 inline:UtkHFdpsaZSO0nwvmEotIoGN+paLt3LHt7yMG4MfDv5rueBsb80 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:6 AES_192_CM_HMAC_SHA1_32 inline:hgFJ9vS9Roz1vAMhnrCyviKC/TwCPgrk5jeKbxHlIEKtqSvLdK4 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:07Q9Z/8DguM0HJgDHv5HF3YDWHdp23hbz+TepHa8 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:djpscuSL/05sVtFNocQpHmr1x+NTcLXKu8ePb0q+ Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:9 F8_128_HMAC_SHA1_80 inline:5FqC2VY+q3vqaju3JwroIJO/OlCM1i6TNRE/3Atj Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:10 F8_128_HMAC_SHA1_32 inline:kUfTs7OFD6ltRNpDi1dSG5ZjuQAFDeEQD5IvOwme Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:11 NULL_HMAC_SHA1_80 inline:JZ2ruS0914NL4mmS7jxg+ngz0IQaE6FBRlet8abd Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:12 NULL_HMAC_SHA1_32 inline:h21D+oGRkr1dDEoMm1+vEKmKsn+4GxNBwgIew2qA Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:8100c85eb1c58be3580e8acd7f097000 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:hcqW22Sc Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:QjHJKT4qNsfZm80zrcyh8SfI76 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30000 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 2 UDP 2130706430 10.13.1.127 30001 typ host Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 between 10.13.1.127 and 10.13.1.152:14450, type prflx Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Triggering check for GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30000 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30020 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30001 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30021 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing '7n38ig1acl' from 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' from '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30000 remote 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30001 remote 10.13.1.152:14451 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Replacing previously learned prflx ICE candidate with Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Learning new ICE candidate H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:H6719a439:1 between 10.13.1.127 and 103.25.164.57:14450, type host Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Recalculating all ICE pair priorities Mar 9 20:21:02 debian rtpengine[11823]: WARNING: [ui7jc53b9lfbt9lf4nu7]: [ice] Priority collision between candidate pairs GtpEjl9r4IzNpiXr:H6719a439:1 and GtpEjl9r4IzNpiXr:Ha0d0198:1 - ICE will likely fail Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Sending ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.003226 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as succeeded Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Best succeeded ICE pair with all components is GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE not completed yet, but can use pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS: Peer certificate accepted Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS handshake successful Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Start nominating ICE pairs Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Nominating ICE pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Sending nominating ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting nominated ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as valid Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE completed, using pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate Ha0d0198 in favour of Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate H6719a439 in favour of H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.002244 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New ingress SSRC for: 10.13.1.152:14450 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31783 TS 528 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 00, length 16, result 6806f5cc... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 01, length 20, result 044b0cce... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 02, length 14, result 4426c95d... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New egress (direct) SSRC for: 0.0.0.0:9 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: ERR: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] SRTP output wanted, but no crypto suite was negotiated Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Error when sending message. Error: Inappropriate ioctl for device Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31784 TS 1488 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31785 TS 2448 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31786 TS 3408 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected:

2 1

Re: Kamailio stops processing the calls - restart fixes it.
by Henning Westerholt 10 Mar '23

10 Mar '23

Hello, please keep the list in the mail flow. Have you tried to set the acc_function modparam to an empty string, as commented in the linked issue? modparam("uac_redirect","acc_function","") If you don't need the acc function from the uac_redirect. Regarding the other warnings, they should be fixed - but different compilers can report different warnings, so it's sometimes happening. If they are just warnings, they should not create an issue. If you are using Debian/Ubuntu you can just use the prebuild packages from deb.kamailio.org - you don't need to build by yourself. Cheers, Henning -----Original Message----- From: Muhammad Danish Moosa <danishmoosa(a)gmail.com> Sent: Donnerstag, 2. März 2023 14:25 To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Kamailio stops processing the calls - restart fixes it. Hi, I tried to install 5.6.4 (https://www.kamailio.org/pub/kamailio/5.6.4/src/) and get this error. Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: INFO: rr [rr_mod.c:188]: mod_init(): outbound module not available Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: ERROR: uac_redirect [../../modules/acc/acc_api.h:191]: acc_load_api(): cannot find bind_acc Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: ERROR: uac_redirect [uac_redirect.c:259]: redirect_init(): cannot bind to ACC API Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: ERROR: <core> [core/sr_module.c:975]: init_mod(): Error while initializing module uac_redirect (/usr/local/lib64/kamailio/modules/uac_redirect.so) It seems this issue was reported earlier and supposed to be fixed but apparently it's not. https://github.com/kamailio/kamailio/issues/3188 Besides that , I had seen warnings during compilation. What should be the most tested and supported version ? Example Warnings: dmq_funcs.c: In function ‘ki_dmq_send_message’: dmq_funcs.c:303:3: warning: missing braces around initializer [-Wmissing-braces] dmq_peer_t new_peer = {0}; ^ dmq_funcs.c:303:3: warning: (near initialization for ‘new_peer.peer_id’) [-Wmissing-braces] dmq_funcs.c: In function ‘ki_dmq_bcast_message’: dmq_funcs.c:373:3: warning: missing braces around initializer [-Wmissing-braces] dmq_peer_t new_peer = {0}; ^ dmq_funcs.c:373:3: warning: (near initialization for ‘new_peer.peer_id’) [-Wmissing-braces] CC (gcc) [M dmq.so] notification_peer.o CC (gcc) [M dmq.so] dmq.o dmq.c:61:1: warning: missing braces around initializer [-Wmissing-braces] sip_uri_t dmq_server_uri = {0}; On Wed, Mar 1, 2023 at 7:26 PM Henning Westerholt <hw(a)gilawa.com> wrote: > > Hello, > > better take the latest one, e.g. 5.6.4 released yesterday. Minor releases only contains bugfixes, documentation enhancements and similar. Only rarely regressions happen. But you should of course test it. > > Cheers, > > Henning > > -----Original Message----- > From: Muhammad Danish Moosa <danishmoosa(a)gmail.com> > Sent: Mittwoch, 1. März 2023 09:22 > To: Henning Westerholt <hw(a)gilawa.com> > Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] Kamailio stops processing the calls - restart fixes it. > > Thank you for your email. > > Is v5.6.1 (July 6, 2022) stable? > > obtained from > > https://www.kamailio.org/pub/kamailio/latest-stable-version-number > > > On Wed, Mar 1, 2023 at 5:50 PM Henning Westerholt <hw(a)gilawa.com> wrote: > > > > Hello, > > > > hard to say without more information, a backtrace etc... As I first step, I would suggest you to update the system to one of the supported releases, e.g. the latest 5.6.x or 5.5.x. > > > > Cheers, > > > > Henning > > > > -----Original Message----- > > From: Muhammad Danish Moosa <danishmoosa(a)gmail.com> > > Sent: Mittwoch, 1. März 2023 01:39 > > To: sr-users(a)lists.kamailio.org > > Subject: [SR-Users] Kamailio stops processing the calls - restart fixes it. > > > > Hi, > > > > I have a very simple proxy (stateful) where kamailio acts as a proxy between 2 endpoints. Everything works fine for weeks and suddenly I see kamailio stops responding. From pcap I can see kamailio is not proxying the session progress and bombarding invites to one endpoint without any reason. Even that invite was stripped on the Body part. > > > > Restarting kamailio fixes it immediately. Unfortunately I could not take bt full yet. > > > > Version is > > > > kamailio 5.5.3 (x86_64/linux) 473cef > > > > configuration is very simple , routing is based on tm.t_relay ( based on KEMI). > > > > Any help will be welcome. > > > > Danish > > > > > > > > -- > > Muhammad Danish Moosa > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions To > > unsubscribe send an email to sr-users-leave(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: > > > > -- > Muhammad Danish Moosa > > " The core of mans' spirit comes from new experiences. "___ > Christopher McCandless -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

4 21

WebRTC Error
by Social Boh 09 Mar '23

09 Mar '23

Hello, I have this error when trying to register this softphone: https://github.com/InnovateAsterisk/Browser-Phone 61096#015#012Host: sip1.kamailio.xyz:8443#015#012Connection: Upgrade#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Upgrade: websocket#015#012Origin: https://campus.voztovoice.org#015#012Sec-WebSocket-Version: 13#015#012User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/110.0.0.0 Safari/537.36#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language: es-419,es;q=0.6#015#012sec-gpc: 1#015#012Sec-WebSocket-Key: uj5cXle8zzBFUzv4kWB2pg==#015#012Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol: sip#015#012#015#012> Mar 9 08:10:55 sip1 /sbin/kamailio[194807]: WARNING: <script>: HTTP request received on 8443 Mar 9 08:10:55 sip1 /sbin/kamailio[194807]: DEBUG: xhttp [xhttp_mod.c:426]: xhttp_send_reply(): sending out response: 403 Forbidden Any help is really apreciated -- --- I'm SoCIaL, MayBe

1 1

Location / Registrar, TCP: Use existing TCP connection
by Benoit Panizzon 09 Mar '23

09 Mar '23

Hi List CPE behind Firewall, registering to Kamailio via TCP (or TLS). When a call is sent to the CPE, kamailio attemts to open a new TCP connection and is blocked by Firewall. Is there an option to tell kamailio to use the existing registered TCP connection? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

Call recovery for HA with sems
by Patrick Karton 09 Mar '23

09 Mar '23

1 0

TLS -> UDP ReInvite
by Social Boh 08 Mar '23

08 Mar '23

hello, I'm configuring Kamailio to permite calls like: UDP -> UDP TLS -> TLS UDP -> TLS TLS -> UDP Until now all is ok; the only problem is with Microsip SoftPhone. When make a call from Microsip TLS to another phone UDP, when the callee answer, Microsip send a new INVITE and Kamailio don't delete all references to SRTP so the call happens but without audio. I know I have to process the reINVITE where I do has_totag stuff but I don't know how. Regards -- --- I'm SoCIaL, MayBe

1 0

Re: sr-users Digest, Vol 213, Issue 52
by Mohamed SADIK 08 Mar '23

08 Mar '23

Hello, Thank you for your reply! I've used your suggestion to accomplish some amazing things. However, I'm having an issue when calling the *"sl.stats"* or any other procedures. I've noticed that the number of "200" responses gets incremented by 2, which is not the case when using the `*kamcmd*` tool. I suspect that this may be due to jsonrpc using *http/https*, which is causing Kamailio to treat it as a "200" response code. Could you advise on how to avoid this issue? Is there any further configuration that needs to be done? Best regards. On Sat, Feb 25, 2023 at 12:00 PM <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via email, send a message with subject or > body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..."Today's Topics: > > 1. Kamailio RPC configuration (sadik.oualla.mohamed(a)gmail.com) > 2. Re: Kamailio RPC configuration (Alex Balashov) > > > > ---------- Forwarded message ---------- > From: sadik.oualla.mohamed(a)gmail.com > To: sr-users(a)lists.kamailio.org > Cc: > Bcc: > Date: Fri, 24 Feb 2023 17:07:33 -0000 > Subject: [SR-Users] Kamailio RPC configuration > Dear Kamailio community, > > I am a newbie in Kamailio and I am currently working on a project that > involves communicating with Kamailio using Python. I have been exploring > the Kamailio documentation and I have come across RPC, JSONRPCS..., which I > believe can help me achieve my goal. > > However, I am not sure about the steps to follow in the Kamailio > configuration file to create my own script to communicate with Kamailio > using Python (the script is on an other machine on the same LAN). I would > be grateful if someone could guide me through the process. > > I have already set up Kamailio (builed it from the source code 5.5.5) and > Python on my machines and I have basic knowledge of Kamailio configuration. > I just need some guidance on how to set up the RPC module in Kamailio and > how to create a Python script to communicate with it. I found that I can > use RPyC library of python. > > I would appreciate any help or suggestions from the Kamailio community. > Thank you in advance for your time and assistance. > > Best regards, > Wild Coder. > > > > ---------- Forwarded message ---------- > From: Alex Balashov <abalashov(a)evaristesys.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 24 Feb 2023 13:22:16 -0500 > Subject: [SR-Users] Re: Kamailio RPC configuration > Hi, > > The easiest way to communicate with the JSONRPCS module is probably to use > HTTP, in the form of the `xhttp` module: > > https://kamailio.org/docs/modules/5.6.x/modules/xhttp.html > > You will need... > > 1) To enable the HTTP transport in the `jsonrpcs` module; > > 2) A TCP or TLS listener (listen= config directive) to accept the traffic; > > 3) An xhttp `event_route[xhttp:request]` route, as detailed in that > module's docs: > > > https://kamailio.org/docs/modules/5.6.x/modules/xhttp.html#xhttp.f.xhttp_re… > > 4) Probably some measure of security / authentication for #3. > > The example from jsonrpc_dispatch() is perfectly sufficient to get you > started: > > > https://kamailio.org/docs/modules/5.6.x/modules/jsonrpcs.html#jsonrpcs.f.js… > > You should then be able to do things like: > > # curl \ > -X POST \ > -d '{"jsonrpc": "2.0", "id": "abc123", "method": "ul.dump"}' \ > http://${KAMAILIO_IP}:5060/RPC <http://${kamailio_ip}:5060/RPC> > > -- Alex > > > On Feb 24, 2023, at 12:07 PM, sadik.oualla.mohamed(a)gmail.com wrote: > > > > Dear Kamailio community, > > > > I am a newbie in Kamailio and I am currently working on a project that > involves communicating with Kamailio using Python. I have been exploring > the Kamailio documentation and I have come across RPC, JSONRPCS..., which I > believe can help me achieve my goal. > > > > However, I am not sure about the steps to follow in the Kamailio > configuration file to create my own script to communicate with Kamailio > using Python (the script is on an other machine on the same LAN). I would > be grateful if someone could guide me through the process. > > > > I have already set up Kamailio (builed it from the source code 5.5.5) > and Python on my machines and I have basic knowledge of Kamailio > configuration. I just need some guidance on how to set up the RPC module in > Kamailio and how to create a Python script to communicate with it. I found > that I can use RPyC library of python. > > > > I would appreciate any help or suggestions from the Kamailio community. > Thank you in advance for your time and assistance. > > > > Best regards, > > Wild Coder. > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions > > To unsubscribe send an email to sr-users-leave(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: > > -- > Alex Balashov > Principal Consultant > Evariste Systems LLC > Web: https://evaristesys.com > Tel: +1-706-510-6800 e > > _______________________________________________ > sr-users mailing list -- sr-users(a)lists.kamailio.org > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >

1 0

Using jsonrpc_dispatch in xhttp:request - is it possible to restrict the RPC that can be invoked ?
by Chaigneau, Nicolas 07 Mar '23

07 Mar '23

Hello, I'm using jsonrpc_dispatch in a "xhttp:request" section (with xhttp and jsonrpcs modules), e.g.: event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; } xhttp_reply("404", "Not Found", "", ""); exit; } This works, but I would like to restrict the RPC functions that can be invoked this way. Any advice on how I could do that ? The xhttp module provides "url_skip" and "url_match" but this is not useful because in the URL I only have something like "RPC". And the method is in the JSON body. I could parse the JSON body before calling jsonrpc_dispatch, but it's a bit complicated (I don't know if it's feasible in Kamailio script ?), and seems redundant because jsonrpc_dispatch already does that... Maybe somehow this could be checked from the "void *" second argument passed to the RPC function ? Thanks. Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 4

Info: Kamailio World 2023 - Announcing First Group Of Speakers
by Daniel-Constantin Mierla 06 Mar '23

06 Mar '23

Hello, it's now like 3 months till the start of the next Kamailio World Conference (taking place in Berlin, Germany, during June 5-8, 2023) and the first group of speakers has just been published, the details are available at: - https://www.kamailioworld.com/k2023/speakers/ Not all proposals were reviewed and Call for Speakers is still open, therefore if you want to share your interesting experiences or knowledge in the RTC space, submit it via: - https://www.kamailioworld.com/k2023/call-for-speakers/ All together, expect another edition of Kamailio World conference with great content covering topics such as RTC scalability, security, cloud-based deployments and elasticity, NG112/911, IoT, WebRTC, VoLTE and 4G/5G. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Remove silo extra_hdrs before store
by Dr. Barabás Péter 06 Mar '23

06 Mar '23

Hi, I’m using kamailio’s silo module to store offline messages in it. I set extra_hdrs modparam to e.g. : ‘Resent-from-silo’ so the clients get information that the message has arrived directly (immediatelly) from the sender or it was stored offline and comes from silo. So the problem is the following: * if receiver’s socket is broken (e.g. airplane mode on mobile) and kamailio ‘thinks’ it is registered, server tries to send the message * message will be timed out and based on config it will be stored in silo. * the next step is to check if receiver is registered or not (since the client can re-register during 30sec until the message has timed out) * If client is registered, kamailio tries to send the message immediately with m_dump(). * If the receiver’s connection is still broken, the message will be timed out and store in silo. * In every store, a new ‘resent’ extra_hdrs value is appended. When extra_hdrs length reaches 1024 bytes, the m_dump will fail and it blocks the dumping of messages to the given receiver. Question: can extra_hdrs value be removed before store? Or what can be the solution not to duplicate the extra_hdrs value in some bad network situation? Peter

1 0

How to access rtpengine rtcp-metrics?
by Benoit Panizzon 06 Mar '23

06 Mar '23

Hi all I'm trying to incorporate rtcp data from rtpengine into our CDRS. So trying with the first interesting value, the average mos. modparam("rtpengine", "mos_average_pv", "$avp(mos_average)") If I understood right, to get this variable set, I need to call rtpengine_manage() on the message that terminates the call. So on an established call, that is either BYE or the 200 OK to the BYE. Right? if ($rm == "BYE") { rtpengine_manage(); xlog("L_INFO", "$cfg(route): $rm: MOSS: $avp(mos_average)\n"); } onreply_route[MANAGE_REPLY] { [...] rtpengine_manage(); xlog("L_INFO", "$cfg(route): $rm reply MOS: $avp(mos_average)\n"); } Messages pass those blocks, $avp(mos_average) is 'null' no mater what. What am I missing? In the syslog output of rtpengine I see there is rtcp data. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 2

tcp_set_connection_lifetime => in respect to 'idle' connection?
by Benoit Panizzon 06 Mar '23

06 Mar '23

Hi List Digging further into my tcp/tls issues... Upon successful authentication I call: tcp_keepalive_enable("60", "5", "5"); tcp_set_connection_lifetime("120"); I tought, this would do to keep the connection alive. I noticed, despite keepalive packets being exchanged, kamailio is closing the connection after a bit more than 2 minutes. I am right at assuming, that tcp_set_connection_lifetime is not related to how long the connection is idle, but is counted from the SYN, no matter if keepalives were successfully exchanged? Does calling tcp_set_connection_lifetime() again restart the lifetime like for 2 more minutes like in this example? Or would I have to keep increase the value of the lifetime on each successful re-register to keep it open? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

Kamailio v5.6.3 Released
by Daniel-Constantin Mierla 05 Mar '23

05 Mar '23

Hello, Kamailio SIP Server v5.6.3 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.3. For more details about version 5.6.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/01/kamailio-v5-6-3-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 World Conference - June 5-7, 2023 - www.kamailioworld.com

5 10

Kamailio v5.5.6 Released
by Daniel-Constantin Mierla 02 Mar '23

02 Mar '23

Hello, Kamailio SIP Server v5.5.6 stable release is out. This is a maintenance release of the stable branch 5.5 that includes fixes since the release of v5.5.5. 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 v5.5.6. Note that 5.5 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.6, with v5.6.4 being release out of it. For more details about version 5.5.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/03/kamailio-v5-5-6-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 World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Releasing Kamailio v5.5.6
by Daniel-Constantin Mierla 02 Mar '23

02 Mar '23

Hello, I am considering to release Kamailio v5.5.6 soon, branch on branch 5.5, likely on Thursday or Wednesday next week (Mar 2/3, 2023). 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 World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 1

Websocket closing
by David Villasmil 01 Mar '23

01 Mar '23

Hello guys, We're seeing corner cases where the following happens: On proxy-ws - IP1:PORT1 connects via websocket from Client1 - Registration happens on an upstream kamailio - for any reason, the TCP socket closes or times out. - IP1:PORT1 (same IP:PORT combination) connects via websocket from Client2 - Registration happens on an upstream kamailio Now a call comes in to Client1. Because the first connection was never cleaned up, it is sent to the proxy-ws and the proxy will send it to the IP1:PORT1 where Client2 is connected. Short story, proxy-ws doesn't check the IP1:PORT1 where it is sending the INVITE is the actual client it is supposed to be sending... It seems that when a socket is closed, the mapping IP:PORT to Address (i.e.: sip:d4f27e34@994f31243be9.invalid;transport=ws) doesn't seem to be cleared... is this by design? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 6

Kamailio stops processing the calls - restart fixes it.
by Muhammad Danish Moosa 01 Mar '23

01 Mar '23

Hi, I have a very simple proxy (stateful) where kamailio acts as a proxy between 2 endpoints. Everything works fine for weeks and suddenly I see kamailio stops responding. From pcap I can see kamailio is not proxying the session progress and bombarding invites to one endpoint without any reason. Even that invite was stripped on the Body part. Restarting kamailio fixes it immediately. Unfortunately I could not take bt full yet. Version is kamailio 5.5.3 (x86_64/linux) 473cef configuration is very simple , routing is based on tm.t_relay ( based on KEMI). Any help will be welcome. Danish -- Muhammad Danish Moosa

3 5

Kamailio v5.6.4 Released
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, Kamailio SIP Server v5.6.4 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.4. For more details about version 5.6.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/02/kamailio-v5-6-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. An additional note to say that call for speakers and registration for Kamailio World Conference 2023 are now open, more details at: * https://www.kamailioworld.com Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Kamailio MS Teams TLS Issue
by iliusha.md＠gmail.com 27 Feb '23

27 Feb '23

Hello, We have one Kamailio Instance connected with MS Teams (based on this instruction: https://skalatan.de/en/blog/kamailio-sbc-teams), which worked fine for a while until recently we noticed that calls from teams are not working anymore. When I looked through the logs I found that Microsoft cannot establish a TLS connection to our server because of the cipher: TLS accept:error:1408A0C1:SSL routines:ssl3_get_client_hello:no shared cipher (sni: sbc.example.com - domain is obfuscated). Certificate is valid, the configuration is below: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt server_name = sbc1-teams.example.net ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt [client:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt We use a certificate from Sectigo, but I've tried with Let's Encrypt - and it's the same. Any idea what could be the reason?

5 7

Releasing v5.6.4
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, I am considering to release Kamailio v5.6.4 (out of branch 5.6) next week (likely on Monday or Tuesday, Feb 27/28, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 1

Kamailio RPC configuration
by sadik.oualla.mohamed＠gmail.com 24 Feb '23

24 Feb '23

Dear Kamailio community, I am a newbie in Kamailio and I am currently working on a project that involves communicating with Kamailio using Python. I have been exploring the Kamailio documentation and I have come across RPC, JSONRPCS..., which I believe can help me achieve my goal. However, I am not sure about the steps to follow in the Kamailio configuration file to create my own script to communicate with Kamailio using Python (the script is on an other machine on the same LAN). I would be grateful if someone could guide me through the process. I have already set up Kamailio (builed it from the source code 5.5.5) and Python on my machines and I have basic knowledge of Kamailio configuration. I just need some guidance on how to set up the RPC module in Kamailio and how to create a Python script to communicate with it. I found that I can use RPyC library of python. I would appreciate any help or suggestions from the Kamailio community. Thank you in advance for your time and assistance. Best regards, Wild Coder.

2 1

OT: MS Teams source IP address ranges
by Jon Bonilla (Manwe) 24 Feb '23

24 Feb '23

Hi Sorry for the OT but I think here's the place where I an find a lot of Ms teams integrations I've been working on MS teams direct routing integration for PekePBX. It works. I guess I've done it as everybody else, using Henning's guide as base and extending it for multitenant setup (thanks Henning!) What I've realized is that the source IP address of calls coming from MS are not always matching dispatcher hosts. Sometimes they come from another source IP and failover to the dispatcher hosts when they receive no response. That makes some of the calls to have an additional latency Searching in the MS doc I see that they document these nets as source of their signaling: 52.112.0.0/14 52.120.0.0/14 But I've seen IP addresses outside of this range as source. In this blog https://erwinbierens.com/microsoft-teams-direct-routing-ip-addresses/ The ranges are listed as 52.112.0.0/16 52.113.0.0/16 52.114.0.0/16 52.115.0.0/16 52.120.0.0/16 52.121.0.0/16 52.122.0.0/16 52.123.0.0/16 which looks better but scares me out. Having no auth is it secure to bind so many ranges to MS? Do you use anything else than certificate verification for these calls? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

4 6

Dispatcher failover when SRV failover is on?
by Jon Bonilla (Manwe) 23 Feb '23

23 Feb '23

Hi Again, fighting with MS Teams but this time it's a kamailio question. I have MSteams hosts in the dispatcher for OPTIONS and for destination. Acording to MS you need to try sip.pstnhub.foo then sip2.pstnhub.foo and sip3.pstnhub.foo 2 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 1 2 | sip:sip2.pstnhub.microsoft.com;transport=tls | 0 | 2 2 | sip:sip3.pstnhub.microsoft.com;transport=tls | 0 | 3 I was going to implement dispatcher failover as usual ds_select_domain (alg 8) with failure routes but checking the SRV of the first domain I see it already includes 4 hosts: host -t srv _sips._tcp.sip.pstnhub.microsoft.com _sips._tcp.sip.pstnhub.microsoft.com has SRV record 10 0 5061 sip.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 20 0 5061 sip2.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 30 0 5061 sip3.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 40 0 5061 sip4.pstnhub.microsoft.com. So... If I have dns srv failover enabled in kamailio... what would be the point of the dispatcher failover? In case I only want failover in case 408 or 5XX wouldn't the tm module do it for me? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

3 2

What does l_username in UAC do?
by Дилян Палаузов 23 Feb '23

23 Feb '23

Hello, the uac module-registrations have the fields l_uuid - user for Contact: in REGISTER (before @). Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local unique id used to build and match contact addresses.”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local unique user id, e.g.,: 12345678” l_username - Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local username”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local user name, e.g.,: test ”. For me there is no difference between “local unique id” (as for l_uuid) and “local username“ (here). l_domain - used for Contact: in REGISTER (after @), described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “local domain” When is l_username used, what is it purpose? Does it solely serve as possible filter criterion for the RPC commands: uac.reg_info l_username abc uac.reg_enable l_username abc uac.reg_disable l_username abc, and uac.reg_unregister l_username abc Do I see correctly, that via UAC-RPC an unREGISTER can be triggered, but there is no way to trigger a REGISTER over UAC-RPC? Thanks for the clarifications Дилян

2 2

change default 477 response code per case
by Patrick Karton 22 Feb '23

22 Feb '23

Hello , i would like to know if there is a way to change the default 477 response code generated by kamailio in some cases. for example if kamalio generates 477 because its not able to reach a peer i would like to change it to another code before to forward it. Thanks.

2 1

Call History
by Anthony Wittig 21 Feb '23

21 Feb '23

Hello, I'd like to start tracking a call history/log as soon as I receive a call. I'd like the customer to be able to see that there's an incoming call to one of their numbers in real time. I'm thinking of using the RABBITMQ Module to manually emit an event from my main route when I receive an INVITE. Is that a bad idea? What if the broker isn't located on the same host (latency/availability)? I don't want to negatively affect the call processing. I was looking at the Dialog Module and it has three event routes: * event_route[dialog:start] * event_route[dialog:end] * event_route[dialog:failed] It's really close to what I want, but I'd like to capture the dialog before it starts (Unconfirmed or Early). Maybe another option is to scan the Dialog table with an external process and emit events from there. Has anyone had success in creating a realtime call history? Is there a better way to go about doing so? Thanks! -- Anthony Wittig Mango Voice Developer

7 7

Info: Registration to Kamailio World 2023 is open!
by Daniel-Constantin Mierla 21 Feb '23

21 Feb '23

Hello, a quick note to the community forums to announce that the registration for the next Kamailio World Conference (June 5-7, 2022, in Berlin, Germany) is now open! The event returns to an in-person conference format, at the same wonderful location in the city center of Berlin! More details are available at: * https://www.kamailioworld.com/k2023/registration/ Call for presentations is also open, there is already a group of very interesting submissions, we aim to publish details about them in the near future. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

2 2

topos breaking route set on hairpin calls?
by Benoît Panizzon 20 Feb '23

20 Feb '23

Hi Having this situation while elaborating topos in lab. * Kamailio Registration Instance with topos enabled where customers are registered. * Kamailio Core Instance with Dialogue enabled and handling interconnections. Calls in transit from customer to IC work flawlessly, topos does what it should, hides and restores headers needed on the core. But I have run into an issue with customer calling another customer on same registrar instance. To generate a CDR with diaglogue and for decision where to route the calls to (registrar is kept simple and only handles registrations and locations), those calls are also routed via core. This works as long as PRACK is disabled. When PRACK is enabled and CPE B is requesting 100rel, the PRACK send from CPE A has no route set (it was removed by topos). Therefore the PRACK only makes it to the core who then does not know where to route that PRACK. It looks like topos on the REG only restores ONE Route pointing to the core (maybe because the 2nd one pointed to itself?) header when restoring them. I have attempted to spam more record_route() here and there in the config of the core and the registrar to try to force creation of route headers, but failed. Is this a known limitation of using topos? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 4

Re: OT: MS Teams source IP address ranges
by Henning Westerholt 20 Feb '23

20 Feb '23

Hi Jon, you could of course try to add some heuristic, in the end the used IPs are a smaller sub-set. But to just quote from the official docs, it seems to be pretty clear: "You need to open ports for all these IP address ranges in your firewall to allow incoming and outgoing traffic to and from the addresses for signaling." Cheers, Henning ________________________________ From: Jon Bonilla (Manwe) Sent: Monday, February 20, 2023 15:49 To: Henning Westerholt; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] OT: MS Teams source IP address ranges El Mon, 20 Feb 2023 13:06:30 +0000 Henning Westerholt <hw(a)gilawa.com> escribió: > Hello, > > correct me if I am wrong, but the second group of addresses ("/16") is > included in the first group ("/14"), right? > > Regarding the large IP scope, this is the way Microsoft designed it, > unfortunately. There is not that much what you can do about. > > Right. Didn't check the masks. they all were 16 in my mind :) So, if I add those ranges in the permissions table my only way of avoiding spoofed calls is certificate validation? I'm already doing that and also checking the company domain and ms account email address. Just wanted to be sure that nothing "essential" was missing. -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

sendig 4 byte udp nat pings
by Bernd Krueger-Knauber 18 Feb '23

18 Feb '23

Hi, maybe I'm stupid, but I can not find out how to send 4 byte UDP NAT packets to the sip server. loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 15) modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") I'm a absolute beginner with kamailio and the page about the nathelper Module is not really helpful. At the moment no pings are sendet. But I receive 4 byte packets from the sip server. After 15 minutes my calls are cancelled because of the missing pings. Without proxy, the phone send the pings directly to the sip server and the call stays alive. When I experiment with some other nathelper settings, kamailo start results in errors. Any help would be appreciated. Best regards, Bernd

3 11

enable topology hiding on one side or dynamic ip in topoh module Contact and Via Header
by Patrick Karton 17 Feb '23

17 Feb '23

Hello , im using kamailio with two interfaces external and internal. i need a way either to : * enable topoh hiding only when outgoing interface is external ( mask contact and Via ip only when ougoing interface is external * or if it is not possible to enable it only in one direction.i want to know how to configure dynamic(for example in xavp or avp) ip to put in Contact and Via when topoh is enabled. i see that the 'mask_ip' parameter of topoh module is a string. so we can not set a dynamic value here unfortunately. Thanks

4 8

Calling uac_replace multiple times?
by Benoît Panizzon 17 Feb '23

17 Feb '23

Hey all I have started using the uac module to be able to mirror username changes on both legs of a call. But now I found an issue: Assume this situation: Call IC => Customer IC is e164, Customer wants 'local' numbers. IC From: +41991234567 => Customer From: 0991234567 uac_replace_from("sip:" + $avp(from_local) + "@" + $(fu{uri.host})); Customer is absent, after timeout in failure_route call is dispatched to Voicemail Server, which is configured for e164 numbers: uac_replace_from("sip:" + $avp(from_e164) + "@" + $(fu{uri.host})); But then uac complains: replace_uri(): Already called uac_replace for this dialog Why isn't just the most from version kept and used? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

TLS version and cipher from WSS
by Michal Hajek 17 Feb '23

17 Feb '23

Hi, I can log incoming SIPS TLS version and cipher with PV $tls_version and $tls_cipher_info. How can I log these variables on incoming HTTP/WebSocket. Thanks Michal H.

2 2

Identify error in log
by Евгений Мастратов 16 Feb '23

16 Feb '23

Hello colleagues. I'm starting to use Kamailio for educational purposes, as an IMS test zone. Registration of subscribers, calls go successfully, but the following errors are constantly pouring into the log: Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Apparently they don’t carry anything critical, but they clog the log, natively I can’t understand the meaning of the errors. Please tell me what they mean and how to eliminate their cause? If necessary, I will provide the pcscf config.

2 1

uac reg reload frequency
by Dmytro M 16 Feb '23

16 Feb '23

When you do "kamcmd uac.reg_reload" frequently enough, you'll get an error: # kamcmd uac.reg_reload error: 500 - Failed to shift records - check log messages Logs: ERROR: uac [uac_reg.c:416]: uac_reg_ht_shift(): shifting in-memory table is not possible in less than 150 secs What is the reason behind 150 sec limitation for how frequent reloads can be done? Is it configurable in any way? Use case: Remote app needs to disable uac registration. It deletes record in DB and calls kamcmd uac.reg_reload to activate the change. With 150 sec limit it can't apply DB changes immediately. Please advise. Thanks, Dmytro

2 2

ACK on RE-Invite is send to UAC NATed Internal port instead of public port
by Angelo Sipper 15 Feb '23

15 Feb '23

Hi, My setup has: (TLS)UAC 77.1.1.1 -> (TLS)Kamailio 17.17.17.211 (UDP) -> (UDP)Kamailio 17.17.17.181 (UDP) -> Carrier(UDP) Both Kamailio are 4.3 with minor modifications on default cfg, and Topoh and TLS modules are used. I am having an issue on some outbound calls from nated UACs. On these calls our SBC sometimes receives reverse invites from the carrier side within the call. When this happens, my SBC(17.17.17.211) sends the invite to UAC and the UAC sends back 200 OK (SDP). Unfortunately my SBC(17.17.17.211) sends back the ACK not to the correct UAC public port but instead to the UAC internal port making the call to crash with several retries from my SBC to send 200 OK as ACK was never received from UAC! If you look the below traces of reverse invite past within the call, the reinvite has port private 20125 and alias the public port 2314 of UAC but the ACK bellow is send to 20125 and not to 2314 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060 INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314 INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 68 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Content-Length: 0 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 276 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Any help will be much appreciated. Thank you! Regards, Angelo

2 5

SCA Presence : Record-Route Filed Not Update in sca_subscriptions Table
by Vinayak Makwana 14 Feb '23

14 Feb '23

Hello all, I am using the sca module in kamailio script.so when i subscribe with my ip phone for SCA Presence i get all the information in a sca_subscriptions table for the first time. When I reboot the device I get a new record-route header in trace but It does not update *record-route *Filed in sca_subscriptions table. Can anyone help me out with this issue ? *Here's modparam configuration * modparam("sca", "db_update_interval", 10 ) modparam("sca", "db_url", DBURL_LOCAL) modparam( "sca", "line_seize_max_expires", 30) modparam( "sca", "purge_expired_interval", 1800) -- -- * <https://www.ecosmob.com/itexpo/> * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

3 8

SCA Feature:: Record-route query.
by satyaprakash ch 13 Feb '23

13 Feb '23

Hello I have a query regarding the record-route field update in DB for SCA SUBSCRIBE. Here are my test cases *Initial SCA subscription,* We got the below SUBSCRIBE packet in kamailio SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081381rdb66 *Record-Route* : <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=CFFBAF19-19A68CBC Call-ID: bd4b50d9384523c14ea05c532d895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 also in the sca_subscriptions table record-route header is updated as per Initial SUBSCRIBE record-route. *record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> * But After rebooting the device, we got a new record-route in SUBSCRIBE Packet but in DB Record-route Field not updated after getting an updated record-route in Trace. *Kamailio Trace *SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081415rdb48 *Record-Route*: <sip:sansay1081415rdb48spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=35852031-CF749E20 Call-ID: 41fa031d79e35fae4a61c61984895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 as per logic it should update the record-route field in sca_subscriptions table . but it didn't update it : record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> So can you suggest to us how we can update the record-route field as per SUBSCRIBE's packet new record-route ?

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