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37517 discussions

tsilo: ts_append() cannot find the ts_store() data
by Anthony Alba 25 Nov '20

25 Nov '20

Hi, having tsilo + PUSH problem. My probIem is no matter how I ts_store() the data, ts_append() cannot find any matching uri ERROR: tsilo [ts_append.c:64]: ts_append(): failed to retrieve record for sip:david@voice.example.com In route[LOCATION] { ts_store("sip:$tU@voice.example.com") } in route[PUSHJOIN] { ts_append("location", "sip:$tU@voice.example.com") } ts.dump looks like this { "jsonrpc": "2.0", "result": { "Size": 2048, "R-URIs": { "R-URI": "sip:david@192.168.122.7:5060;ob", "Hash": 57095964, "Transactions": { "Transaction": { "Tindex": 15146, "Tlabel": 805874386 } } }, "Stats": { "RURIs": 1, "Max-Slots": 1, "Transactions": 1 } }, "id": 43582 } Thanks Anthony Alba

3 7

kamailio.cfg fomatter
by marek 25 Nov '20

25 Nov '20

hi, i'm using https://github.com/miconda/vscode-kamailio-syntax in VScode. its great! But i need format code. Are there some formatters for kamailio.cfg? It can be for other editors than VScode. Marek

6 9

ACK Relay Issue Transaction Not Found
by me.projects＠yahoo.com 24 Nov '20

24 Nov '20

Hi When ACK is received on Kamailio from PBX t_check_trans() fails. 200 OK sent PBX from Teams and ACK from PBX is provided here. What could be the issue. SIP/2.0 200 OK FROM: "Ext1"<sip:1102@PBX_IP>;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f TO: <sip:1101@KAM_IP>;tag=e667397cb47c457a860222931e320b41 CSEQ: 25772 INVITE CALL-ID: a6685754-b402-4a15-82b2-5fb2551e5b58 VIA: SIP/2.0/UDP PBX_IP:7790;received=PBX_IP;branch=z9hG4bKPj170912fc-0908-40bb-8248-9f6cb560332d;rport=7790 RECORD-ROUTE: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>,<sip:sbc.callsprout.com;transport=tls;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f;lr>,<sip:KAM_IP:7790;transport=udp;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f;lr> CONTACT: <sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;ias=52.114.132.46~5061~3> CONTENT-LENGTH: 353 SUPPORTED: timer CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY REQUIRE: timer SESSION-EXPIRES: 1800;refresher=uac SERVER: Microsoft.PSTNHub.SIPProxy v.2020.11.16.5 i.USEA.5 X-Siptrace-Fromip: tls:KAM_IP:7790 X-Siptrace-Toip: udp:PBX_IP:7790 X-Siptrace-Time: 1606205334 607029 X-Siptrace-Method: INVITE X-Siptrace-Dir: out ACK sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;alias=.114.132.46~5061~3 SIP/2.0 Via: SIP/2.0/UDP PBX_IP:7790;rport;branch=z9hG4bKPjaa30c14c-c965-473d-bc49-9f223f9f634c From: "Ext1" <sip:1102@PBX_IP>;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f To: <sip:1101@KAM_IP>;tag=e667397cb47c457a860222931e320b41 Call-ID: a6685754-b402-4a15-82b2-5fb2551e5b58 CSeq: 25772 ACK Route: <sip:KAM_IP:7790;lr;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f> Route: <sip:sbc.callsprout.com;transport=tls;lr;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f> Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 X-Siptrace-Fromip: udp:PBX_IP:7790 X-Siptrace-Toip: udp:KAM_IP:7790 X-Siptrace-Time: 1606205334 615495 X-Siptrace-Method: ACK X-Siptrace-Dir: in

5 10

How to route to calls to asterisk
by Mihai Cezar 23 Nov '20

23 Nov '20

Hi all, I need some help on getting to work my scenario: I have 3 asterisk servers, and every agents (soft phones like zoiper) is connecting to which server they need by contacting the PublicIP of each server. (the problem is that I need to reuse the IP address on others servers) . I want if is possible to use a proxy like "haproxy" but for sip to redirect the agents (softphones), to their coresponding server. For example: - james(a)ast.example.com -> must register to asterisk server 1, - sarah(a)ast2.exampe.com -> must register to asterisk server 2 - andy(a)ast3.example.com-> must register to asterisk server 3 ast.example.com, ast2.example.com, ast3.example.com = will share all the same ip, when the request arrives in the proxy it should check the vhost and redirect to the coresponding server for registering. Any hints and how to do this scenario?

2 3

ACK Not Relayed to MS Teams
by me.projects＠yahoo.com 23 Nov '20

23 Nov '20

Hi, I am integrating PBX with Teams via Kamailio Call is working but ACK delivery has an issue This is the ACK Kamailio receives from PBX I process the ACK with handle_ruri_alias(); and record_route_preset(AS per GUIDE LINES) But this change is not reflected in RELAY ACK from PBX ACK sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=0c344c6a-cd75-4095-9891-bf9445c28b82;x-c=5691aa770226562db7d3e16ab090e196/s/1/e857255ef2344e03bd1d12e270ef5e6c;ias=52.114.132.46~5061~3 SIP/2.0 Via: SIP/2.0/UDP PBX_IP:7790;rport;branch=z9hG4bKPjc50440d0-77db-4467-94cb-2f93c4f88538 From: "USER" <sip:13300@PBX_IP>;tag=ab87b9b5-eee3-420c-8472-2eec82a769ac To: <sip:1508@KAMAILIO_IP>;tag=a2ce378b106949cf8bac8f978bb573fb Call-ID: 3dbc6680-fec3-4ba5-8232-6dda8c669bca CSeq: 23811 ACK Route: <sip:KAMAILIO_IP:7790;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sbc.KAMAILIO_FQDN.com;transport=tls;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 X-Siptrace-Fromip: udp:PBX_IP:7790 X-Siptrace-Toip: udp:KAMAILIO_IP:7790 X-Siptrace-Time: 1606135546 516717 X-Siptrace-Method: ACK X-Siptrace-Dir: in

2 3

textops.subst_uri does not change all branches
by Anthony Alba 23 Nov '20

23 Nov '20

Hello kamailioers, Version 5.4.2. I have a R-URI that I need to change TLS -> TCP in all branches. I do this in request route before t_relay: // AoR has multiple contacts, all with transport=TLS route[REQUEST] { textops.subst_uri("/transport=TLS/transport=TCP/i") //parallel forking //why doesn't this change all branches? t_relay() } If I print out the R-URI in the branch route, I see that only the first R-URI is changed, in the other branches the R-URI transport=TLS is not changed to transport=TCP. Any suggestions on how to get this to work across all Contact addresses? Thanks Anthony Alba

2 1

Issue with ca-list
by George Goglidze 20 Nov '20

20 Nov '20

Hi Folks, I was wondering if somebody could help me with an issue. I’m a newbie here, just installing Kamailio sip server. I’ve enabled TLS, and am trying create a SIP Trunk to external SIP Service which is TLS enabled port 5061. I’ve configured the following in tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certs/sbc-private.pem certificate = /etc/kamailio/certs/godaddy.pem ca_list = /etc/kamailio/certs/calist.pem In the section above – ca_list = calist.pem contains all the CA’s and Subordinates of the destination server. Private_key and certificate are of my own server (public godaddy signed) [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certs/sbc-private.pem certificate = /etc/kamailio/certs/godaddy.pem ca_list = /etc/kamailio/certs/godaddyca.pem In the section above the ca_list is godaddy’s ca and subordinate. In the wireshark I can see that I’m sending out SIP OPTIONS PING (I’m using dispatcher module). Then the server replies with tls SERVER HELLO which includes it’s certificate But for some reason we are rejecting it: Alert (level: fatal, Description: Unknown CA) How should I set this up to make sure the remote server CA’s are verified? Thank you,

3 9

t_uac_send function with From tag
by João Vitor Arruda 20 Nov '20

20 Nov '20

Hello, I'm trying to use the function t_uac_send inside a failure_route as described in https://kamailio.org/docs/modules/5.3.x/modules/tm.html#tm.f.t_uac_send to send a CANCEL out. Although the documentation says "it can include From/To tags" i was not able to get the from tag provided to be used. Another random generated one is used. I've tried the following format without success: t_uac_send("CANCEL", "$ru", "", "", "From: <$fu>;tag=$ft\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") t_uac_send("CANCEL", "$ru", "", "", "From: $fu;tag=$ft\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") t_uac_send("CANCEL", "$ru", "", "", "From: bob(a)kamailio.org;tag=2w3e\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") In the last one which is copied from the documentation that is how the >From header going out looks like: From: <bob(a)kamailio.org;tag=2w3e>;tag=3393f0703fb0ccaca74109ff37de39f5-36d71ef0 Any tips on how to get the From tag passed to the function to be used? Thanks, Joao Arruda

2 6

Kamailio suddenly stopes responding to sip messages
by Uttam Giri 19 Nov '20

19 Nov '20

Hi, We are using Kamailio with Jitsi client where we observed sometimes Kamailio stops responding to sip messages. As a result, my Jitsi client goes offline and never back online. Also in the Kamailio syslog there are no error logs(debug enabled) related to this issue. As a temporary workaround, we had to restart the Kamailio server all the time. Does anyone observed a similar issue and can suggest any resolution. Kamailio version:- version: kamailio 5.3.5 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 7.5.0 UDP process when Kamailio stops responding:-. root@ip-10-68-16-14:/home/ugiri# kamctl ps { "jsonrpc": "2.0", "result": [ { "IDX": 0, "PID": 15436, "DSC": "main process - attendant" }, { "IDX": 1, "PID": 15442, "DSC": "udp receiver child=0 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 2, "PID": 15443, "DSC": "udp receiver child=1 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 3, "PID": 15444, "DSC": "udp receiver child=2 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 4, "PID": 15445, "DSC": "udp receiver child=3 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 5, "PID": 15446, "DSC": "udp receiver child=4 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 6, "PID": 15447, "DSC": "udp receiver child=5 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 7, "PID": 15448, "DSC": "udp receiver child=6 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 8, "PID": 15449, "DSC": "udp receiver child=7 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 9, "PID": 15450, "DSC": "slow timer" }, { "IDX": 10, "PID": 15451, "DSC": "timer" }, { "IDX": 11, "PID": 15452, "DSC": "secondary timer" }, { "IDX": 12, "PID": 15453, "DSC": "JSONRPCS FIFO" }, { "IDX": 13, "PID": 15454, "DSC": "JSONRPCS DATAGRAM" }, { "IDX": 14, "PID": 15455, "DSC": "ctl handler" }, { "IDX": 15, "PID": 15456, "DSC": "Http Async Worker" }, { "IDX": 16, "PID": 15457, "DSC": "Http Async Worker" }, { "IDX": 17, "PID": 15458, "DSC": "TIMER NH" }, { "IDX": 18, "PID": 15459, "DSC": "tcp main process" } ], "id": 17709 } Thanks, Uttam

2 3

Re-Invite from callee With RTPEngine between RTP--SRTP
by me.projects＠yahoo.com 19 Nov '20

19 Nov '20

Hi, I have have Kamailio as SBC one-end is SRTP and other end is RTP So I m using rtpengine_manage("replace-origin replace-session-connection DTLS=passive OSRTP-offer ICE=force SRTP")rtpengine_manage("replace-origin replace-session-connection DTLS=off SDES-off ICE=remove RTP") 1-Asterisk(RTP) -----2-Kamailio/RTPEngine ------ 3- Carrier with SRTP I detect source IPs and to manage this all. When call is made from 1 to 3 all works and during call 3 send re-Invite to Hold Call .. We need to call rtpengine_manage and also when on 200OK reply of re-invite from 1. How shall I detect that 1- if this is Re-Invite (has_totag() )and 2- its from Callee. Please advise.

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