- sr-users - kamailio.org

Releasing Kamailio v.5.4.8
by Daniel-Constantin Mierla 09 Mar '22

09 Mar '22

Hello, I am planning to release a new version from branch 5.4, respectively 5.4.8, sometime next week, likely on Wednesday or Thursday (Mar 9/10, 2022). If anyone is aware of issues not yet reported to bug tracker or missing backports, report them in order to try to get the fixes in this release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online March 28-31, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

2 3

rtpengine - SRTP <> RTP missing a=crypto
by Rhys Hanrahan 09 Mar '22

09 Mar '22

Hi Everyone, I have Kamailio sitting between MS Teams and Asterisk, and using rtpengine to terminate SRTP on Kamailio so that all my internal traffic is unencrypted. My current config works fine for inbound calls where I initiate the INVITE and Teams responds, but if Teams sends the INVITE I am having an issue where SRTP cannot finish negotiating. Non SRTP calls work fine with RTPEngine as well, so it's just the RTP to SRTP I am struggling with. According to this I believe I must pass a=crypto in response to the INVITE which also has a=crypto: https://www.dialogic.com/-/media/1f8b54b43087407d9c2b38846c5c2cb5.ashx?h=40… You can see that in the initial invite from Teams, I get RTP/SAVP with a=crypto, but I do not send one in my OK response after 183 Session In Progress. As below - I am wondering if it's because not all audio channels seem to be getting swapped to SAVP? I'd like to do a generic SRTP <> RTP bridge config (I've tried below). However, I am not 100% sure on how to detect when to swap between AVP and SAVP, so I've also tried just doing rtpengine_manage() and relying on other code to swap between SAVP or AVP *only* when going to/from Teams to keep it simple. I also tried both with and without "replace-origin replace-session-connection ICE=remove" but I still get the same behaviour in all cases. Any advice appreciated, as this is my first time dealing with SRTP (and rtpengine). Feeling very stuck. Thanks! branch_route[MANAGE_BRANCH] { ... route(NATMANAGE); route(HANDLE_SRTP); } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } route(HANDLE_SRTP); } route[HANDLE_SRTP] { if (!has_body("application/sdp")) { return; } rtpengine_manage(); return; # As a test, just do rtpengine_manage() and set SAVP/AVP elsewhere. Same behaviour. # Handle bridging of RTP and SRTP # Inbound traffic to SBC should be converted from SRTP to RTP if (proto==TLS) { rtpengine_manage("RTP/AVP"); # Outbound traffic destined to a TLS destination should be converted from RTP to SRTP } else if ($ru =~ "transport=tls") { rtpengine_manage("RTP/SAVP"); } } # INVITE from teams rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP"); # INVITE to teams rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/SAVP"); INVITE sip:+614xxxx@rh.sbc-syd-01.teams.xxxx:5061;user=phone;transport=tls SIP/2.0^M ... v=0^M o=- 57931 0 IN IP4 127.0.0.1^M s=session^M c=IN IP4 52.113.76.53^M b=CT:10000000^M t=0 0^M m=audio 51398 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118^M c=IN IP4 52.113.76.53^M a=rtcp:51399^M a=ice-ufrag:C8ss^M a=ice-pwd:2bV9D6GcXF5f8m0px/wufQD/^M a=rtcp-mux^M a=candidate:1 1 UDP 2130706431 52.113.76.53 51398 typ srflx raddr 10.0.32.179 rport 51398^M a=candidate:1 2 UDP 2130705918 52.113.76.53 51399 typ srflx raddr 10.0.32.179 rport 51399^M a=candidate:2 1 tcp-act 2121006078 52.113.76.53 49152 typ srflx raddr 10.0.32.179 rport 49152^M a=candidate:2 2 tcp-act 2121006078 52.113.76.53 49152 typ srflx raddr 10.0.32.179 rport 49152^M a=label:main-audio^M a=mid:1^M a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:geUHLB1mshmnI5hN83bnO57Hbdm2i7dD14sDAnpA|2^31^M a=sendrecv^M a=rtpmap:104 SILK/16000^M a=rtpmap:9 G722/8000^M a=rtpmap:103 SILK/8000^M a=rtpmap:111 SIREN/16000^M a=fmtp:111 bitrate=16000^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:97 RED/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=rtpmap:13 CN/8000^M a=rtpmap:118 CN/16000^M a=ptime:20^M I correctly convert to/from RTP/AVP and RTP/SAVP for the 183 Session in progress. It is RTP/SAVP before going to Teams: SIP/2.0 183 Session Progress^M ... v=0^M o=- 57931 2 IN IP4 1.2.3.4^M s=NexusOne^M c=IN IP4 1.2.3.4^M t=0 0^M m=audio 37820 RTP/SAVP 9 8 0 101^M a=maxptime:150^M a=mid:1^M a=rtpmap:9 G722/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:37821^M a=ptime:20^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M But then when I send the OK after the 183, I am setting RTP/SAVP before sending to MS Teams, but not setting a=crypto: Also note that I can see there are _some_ channels still as RTP/AVP so maybe this is part of the issue. SIP/2.0 200 OK^M ... v=0^M o=- 57931 2 IN IP4 1.2.3.4^M s=NexusOne^M c=IN IP4 1.2.3.4^M t=0 0^M m=audio 37820 RTP/SAVP 9 8 0 101^M a=maxptime:150^M a=mid:1^M a=rtpmap:9 G722/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:37821^M a=ptime:20^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M Rhys Hanrahan | Chief Information Officer e: rhys(a)nexusone.com.au<mailto:rhys@nexusone.com.au> [www.nexusone.com.au]<http://www.nexusone.com.au/> [signature_1116663581] <http://www.fusiontech.com.au/> NEXUS ONE | FUSION TECHNOLOGY SOLUTIONS p: 1800 NEXUS1 (1800 639 871) or 1800 565 845 | a: Suite 12.03 Level 12, 227 Elizabeth Street, Sydney NSW 2000 www.nexusone.com.au<http://www.nexusone.com.au/> | www.fusiontech.com.au<http://www.fusiontech.com.au/> The information in this email and any accompanying attachments may contain; a. Confidential information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; b. Legally privileged information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; and or c. Copyright material Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties. If you have received this email in error, please notify the sender immediately and delete this message. Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd does not accept any responsibility for loss or damage arising from the use or distribution of this email. Please consider the environment before printing this email.

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Kamailio + KEMI + topos + change_reply_status
by Marrold 09 Mar '22

09 Mar '22

Hi, I'm using Kamailio 5.5.3 with a KEMI / Python based configuration on Debian 11. In my reply route I am trying to convert a 183 to a 180 with: KSR.textopsx.change_reply_status(180, "Ringing") I've noticed with TOPOS enabled, it doesn't work and I still see a 183 go out. But with it disabled, it works as expected and it's converted to a 180. I get the same behaviour with / without msg_apply_changes() Config looks like this: def ksr_onreply_manage(self, msg): KSR.info(f"ksr_onreply_manage\r\n") scode = KSR.pv.get("$rs") if KSR.permissions.allow_source_address_group()<0: if scode == 183: KSR.info(f"ksr_onreply_manage: Converting 183 to 180\r\n") KSR.textopsx.change_reply_status(180, "Ringing") KSR.textopsx.msg_apply_changes() if scode>100 and scode<299 : self.ksr_route_natmanage(msg) return 1 The debug log doesn't reveal anything too obvious, I can share if its of any use. Does anyone have any suggestions? Thanks Matthew

2 2

kamailio.cfg : behind the NAT (listen and advertise) understanding
by Youssef Boujraf 07 Mar '22

07 Mar '22

Hi Dear I am trying to understand how "listen" and "advertise" work for sip signaling purpose ( corebooks : listen <http://www.kamailio.org/wiki/cookbooks/5.4.x/core#listen> and advertise ) Flow : kamailio behind NAT using softphones on private and public networks. - kamailio : Network : IPv4_private 192.168.1.6 / IPv4_public kamailio.cfg : listen=udp: IPv4_private: 5060 advertise IPv4_public : 5060 - uac (softphone) : zoiper, microsip, cisco spa508g USE CASE : I 'd like to force all softphone on the same local network to reach kamailio through a public ip address. ??? QUESTION : Do I need to REGISTER on IPv4_private address of Kamailio OR REGISTER on IPv4_public address of kamailio or DEFINE another "listen" with another PORT WITHOUT "advertise" for softphone on same kamailio network ? listen=udp: IPv4_private: 5060 advertise IPv4_public : 5060 listen=udp: IPv4_private: 5066 Do I need to use "kamctl ps" to verify which interface is listening on port ? Thanks in advance Best Regards, Youssef BOUJRAF Email: yboujraf(a)gmail.com

3 5

simple GUI for Kamailio as a SBC - could be paid
by Krzysztof Drewicz 05 Mar '22

05 Mar '22

Hello, We searched for a simple GUI that could drive a kamailio - a bunch of trunks, some custom logic, tried to use open source Sipwise C5 - a way too big and heavy. Any hints ? links ? Mabe a paid subscription? Our goal is - to have a GUI that could - put a version number to configuration, and have a simple way to audit changes - only to log - what was done, when and from which management account. vi + svn/git - it's working but this is no GUI, ACLs in sqlite, also work, but - need to have a ssh console, keep files in some repository - well this is simply inconvenient when you have more than 5 trunks with registration etc... BR, -- *Krzysztof Drewicz* Senior Infrastructure Administrator CLUDO | ul. Grochowska 306/308, 03-840 Warszawa t+48221223977 kdrewicz(a)cludo.pl | www.cludo.pl <http://www.cludo.pl/>

5 7

OS-assigned TLS source Port#?
by Matthias Urlichs 02 Mar '22

02 Mar '22

Hi, My Kamailio installation originates TLS connections using source port 5061. Is there a way to teach it not to do that? Seting tcp_reuse_port=no doesn't have any effect. -- -- Matthias Urlichs

2 2

ndb_redis - redis-cluster error
by Maharaja Azhagiah 02 Mar '22

02 Mar '22

Hi All, I am new to Kamailio and testing location server using kamailio and redis-cluster. My setup is simple: All SIP REGISTER messages will be forwarded to this location-server. The server checks for AUTH and then needs to save the usrloc in redis-cluster. I have hiredis and hiredis-cluster libraries installed I am using ndb_redis to support redis-cluster with below configuration: modparam("ndb_redis", "server", "name=172.17.0.3:7000 ;addr=172.17.0.3;port=7000") modparam("ndb_redis", "server", "name=172.17.0.4:7001 ;addr=172.17.0.4;port=7001") modparam("ndb_redis", "server", "name=172.17.0.5:7002 ;addr=172.17.0.5;port=7002") modparam("ndb_redis", "server", "name=172.17.0.6:7003 ;addr=172.17.0.6;port=7003") modparam("ndb_redis", "server", "name=172.17.0.7:7004 ;addr=172.17.0.7;port=7004") modparam("ndb_redis", "server", "name=172.17.0.8:7005 ;addr=172.17.0.8;port=7005") modparam("ndb_redis", "cluster", 1) If SIP method is REGISTER, Kamailio sends redis_cmd to store as below: redis_cmd("172.17.0.3:7000", "sadd %s %s", "location:usrdom::437780082: mazhagiah.sip.zang.io", "location:entry::$xavp(ulrcd=>ruid)", "r"); However, redis-cluster replies with MOVED message, but kamailio / ndb_redis throughs error as below: 0(3915) DEBUG: ndb_redis [redis_client.c:391]: redisc_get_server(): Hash 2590462363 (172.17.0.3:7000) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2591109923 (172.17.0.8:7005) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590232843 (172.17.0.7:7004) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590175467 (172.17.0.6:7003) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590347603 (172.17.0.5:7002) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590290227 (172.17.0.4:7001) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590462363 (172.17.0.3:7000) 0(3915) DEBUG: ndb_redis [redis_client.c:896]: redisc_exec(): rsrv->ctxRedis = 0x558e4d739210 0(3915) DEBUG: ndb_redis [redis_client.c:903]: redisc_exec(): rsrv->ctxRedis = 0x558e4d739210 0(3915) DEBUG: ndb_redis [redis_client.c:957]: redisc_exec(): rpl->rplRedis->type:6 0(3915) ERROR: ndb_redis [redis_client.c:959]: redisc_exec(): Redis error:MOVED 10082 172.17.0.4:7001 0(3915) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-621e8438-f4b-b] (11 / 19) 0(3915) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-621e8438-f4b-b 0(3915) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-621e8438-f4b-c] (12 / 19) Please help if I am missing anything with my configuration or library. Thanks Raja

2 1

Problem with auth_ephemeral and parse_uri(): bad port in uri
by Vlasis Chatzistavrou 01 Mar '22

01 Mar '22

Hello, I have a problem with Kamailio 5.4.6 and auth_ephemeral. I have the following in the Kamailio configuration loadmodule "auth_ephemeral" modparam( "auth_ephemeral", "sha_algorithm", 3 ) modparam( "auth_ephemeral", "username_format", 0 ) modparam( "auth_ephemeral", "secret", 1234 ) as per https://kamailio.org/docs/modules/4.1.x/modules/auth_ephemeral.html#auth_ep… and registrations fail. In the logs we see: Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:777]: check_parse_uris(): looking up From header Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:817]: check_parse_uris(): parsing From URI Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} <core> [core/parser/parse_uri.c:1296]: parse_uri(): bad port in uri (error at char 5 in state 2) parsed: <sip:3518929:16411>(17) /<sip:3518929:1641150726@192.168.2.99> (35) Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: WARNING: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:820]: check_parse_uris(): failed to parse From uri Apparently Kamailio is confused by the timestamp following the username separated by the : character. The REGISTER message is below: REGISTER sip:192.168.2.99 SIP/2.0 Via: SIP/2.0/WSS 192.0.2.202;branch=z9hG4bK5452321 Max-Forwards: 70 To: "3518929" <sip:3518929:1641148397@192.168.2.99> From: "3518929" <sip:3518929:1641148397@192.168.2.99>;tag=ht76o8b2b6 Call-ID: phkj9mi2n3s3ju7uu3qq2f CSeq: 274 REGISTER Contact: <sip:edh7mmti@192.0.2.202;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:ca5e9372-dfa1-459a-b6ba-4398d23bd896>";expires=300 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, outbound User-Agent: Raspberry Phone (SipJS - 0.11.6) Content-Length: 0 and Kamailio parses it as sip:<username>:<port> instead of sip:<username>:<timestamp>. Is this a bug that should be reported or is there any setting that I am missing?

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t_on_failure => what variables being reset to which values?
by Benoît Panizzon 01 Mar '22

01 Mar '22

Hi List We t_on_failure re-route calls to another destination in case one fails for example announcement servers... Before sending to the announcement server I do add some extra header to define which announcement in which language should be played: append_hf("IMP-annc: $avp(announcecode)\r\n"); append_hf("IMP-mandate: $avp(mandate)\r\n"); append_hf("IMP-language: $avp(callerlang)\r\n"); Also $rU is being set prior to sending the call to the announcement server. Now one of our announcement servers became unavailable and t_on_failure was triggered. The 2nd announcement servers was receiving the calls without any of the additional header and with the 'inbound' $ru without the altered $rU https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_on_failu… I'm a bit confused by that, as this states the URI is reset to the value it had on 'relaying'. Is this on 'inbound replaying' or on 'outbound relaying' when route(RELAY) was called to send the call TO the first annoucement server? Is there a bit more specific information on that behaviour? -- 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 ______________________________________________________

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KEMI jsdt CommonJS-based module
by Ian Carlson 01 Mar '22

01 Mar '22

Greetings, I'm new to Kamailio and working to add Kamailio to an existing Asterisk solution. I'm attracted to the idea of routing via KEMI jsdt since our Asterisk ARI is using nodejs. Playing with KEMI jsdt I was trying to use require() to break up the monolithic routing script. It seems that Kamailio uses Ducktape 2 without module-node or module-ducktape, requiring all routing in a single file? I'd appreciate any suggested practices for managing Kamailio configurations. Cheers, Ian

2 4

RTP Engine early media issue one to many
by Filippo Graziola 28 Feb '22

28 Feb '22

Hi everyone, in my environment (kamailio - http API for retrieve destinations - rtpengine) it is used a lot parallel forking, the flow is as follows: A -> INVITE -> Kamailio -> http API/rtjson / t_relay -> Kamailio -> client B / C / D (only one accepting the call) Kamailio loops to itself to have a parallel forking, it involves rtpengine correctly. Both audio and video work during call. The issue comes with one of destinations sending 183 with SDP, the caller will send then a video preview as early media before call is established. I noticed that only the first one "asking" for early media is receiving this video preview (sometimes none of clients requesting it), all others are ignored by caller. This is a case of 1-to-many, I searched a lot about this but it seems at least not working or not supported, could you confirm that there is no configuration trick possible to make work this early media to many destinations? Another issue in above call flow, is that if client B is receiving correctly this early media video and client C answers the call (200OK) client C does not see any video, audio working and video is completely missing from rtp, only a reinvite makes possible to see video. Is this a client side only issue (so only solution is the reinvite) or there is something missing on my kamailio configuration? Thanks Filippo

1 0

ACK and DNS failover
by Julien Klingenmeyer 25 Feb '22

25 Feb '22

Hello, I use DNS failover feature for routing some calls, and I wonder about its limitations. I noticed that INVITE and ReINVITE requests are correctly routed based on SRV priority/weight in case of failure, but ACK requests do not use them. Let’s say I have Kamailio A relaying requests to a pool of Kamailios B1 and B2 in TLS. DNS records are the below ones. kamailiob.net. 60 NAPTR 20 100 S SIPS+D2T _sips._tcp.kamailiob.net. _sips._tcp.kamailiob.net. 60 SRV 1 10 5061 kamailiob-1.net. _sips._tcp.kamailiob.net. 60 SRV 2 10 5061 kamailiob-2.net. If B1 is down, I expect requests being relayed to B2 (because of priority 2 in the SRV record). * For initial INVITEs: R-URI is sip:kamailiob.net * If B1 is down, the request is retried to B2 => OK * For in-dialog requests: Route header in incoming request is sip:kamailiob.net so it is set as next destination with loose_route function * If B1 is down, the request is retried to B2 for ReINVITE => OK * But once the ACK comes in, with the same Route header (sip:kamailiob.net), Kamailio tries to send it to B1 only (no retries to B2, and even no retransmissions to B1 either). * When B1 is back UP again a few seconds later, Kamailio does not try to relay the ACK again. The ACK request is attempted to be retransmitted only once to B1. It fails and no more retries after that. Is it the expected behavior? Or is it something misconfigured on Kamailio A? Or is it a TLS connection issue? Here is the DNS configuration on Kamailio A (TM module is enabled): dns_try_naptr=yes use_dns_failover=yes use_dns_cache=yes dns_srv_lb=yes enable_tls=yes tcp_max_connections=4096 tls_max_connections=4096 tcp_reuse_port=yes tcp_connect_timeout=10 tcp_send_timeout=10 tcp_connection_lifetime=3600 I had a look into the DNS tutorial (https://github.com/kamailio/kamailio/blob/master/doc/tutorials/dns.txt) without finding any hint about this. If anyone has already played with SIP DNS failover in Kamailio, your help would be appreciated, thanks! Julien

3 4

kamailio and certificate renewal
by Aymeric Moizard 25 Feb '22

25 Feb '22

Hi all, I'm switching my kamailio to use Let's Encrypt certificates. Do I need to force a restart of kamailio when the certificate is renewed or will this be automatic? The doc indicates that reload is possible (but not advised) but it looks to be for a "configuration change" (tls.cfg file change) and not for certificate change. Maybe certificates are automatically reloaded? Regards Aymeric -- Antisip - http://www.antisip.com

2 1

Re-using an existing TLS connection
by Matthias Urlichs 25 Feb '22

25 Feb '22

Hello, My problem: our provider opens a TLS connection to us. They want us to use this channel for outgoing calls, instead of opening a new SIP connection to their server. Is there a way to teach Kamailio to do that? -- -- Matthias Urlichs

3 3

Info: SIP resources collection
by Daniel-Constantin Mierla 24 Feb '22

24 Feb '22

Hello, over the 20 years I used various SIP applications, libraries and tools but after a while I forgot many of them, therefore I tried to collect many in a list hosted on github at: * https://github.com/miconda/sip-resources Hoping it proves useful for some of you! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

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kamailio : how to xlog the value of "myself" ?
by Youssef Boujraf 24 Feb '22

24 Feb '22

Dear I 'd like to xlog the value of the "myself" ? if(uri==myself) { log("the request is for local processing\n"); };Best Regards, -- Youssef BOUJRAF

3 3

dispatcher.reload
by David Villasmil 24 Feb '22

24 Feb '22

Hello guys, We're dispatching with hash over user, if i execute dispatcher.reload <http://www.kamailio.net/docs/modules/5.0.x/modules/dispatcher.html#dispatch…> does the dispatcher list gets calculated? i.e. user A is always dispatched to box3, if i reload, is the hash recalculated and will probably be dispatched differently? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 4

Dispatcher Load Balance with SIP Trunk
by warawich 24 Feb '22

24 Feb '22

Hello, I would like to do a load balance between Asterisk SIP trunks. You can see a diagram from this link: https://drive.google.com/file/d/1Qy66L5rQCfxysYQpSd2-ek_8-by0T8PR/view?usp=… SIP Packets capture log: https://drive.google.com/file/d/1CHGUOwoRDAC93MMBtyfa8gBISKIVC-ng/view?usp=… Details 1. Asterisk1 makes SIP trunk connection with Kamailio. 2. Kamailio makes SIP trunk connection with Asterisk2 and Asterisk3 3. Caller register SIP phone with Asterisk1 Caller extension = 8002 4. Asterisk2 makes a blind transfer to Kamailio. (Call to 8009 then transfer to ARI APP) exten = 8009,1,Transfer(SIP/3802@<kamailio public IP>) 5. ARI app extension is 3802 My problem is that Kamailio LB only works when I try to connect with Asterisk1 and Asterisk2 or Asterisk1 and Asterisk3. If I have two Asterisks in dispatcher.list, it doesn't work and it appears *SIP/2.0 401 Unauthorized* in sip packets capture log. Kamailio version: 5.5.2 Kamailio.cfg modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 3) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } dispatcher.list # setid(int) destination(sip uri) flags(int,opt) priority(int,opt) attrs(str,opt) 1 sip:10.148.0.31:5123 1 sip:10.148.0.44:5123 Thanks, Warawich

2 2

SQLOPS sql_query_async() not functioning in an async manner?
by Ben Kaufman 23 Feb '22

23 Feb '22

I'm not sure if this is by design or not. I have things I want to log to a SQL DB, but it's not nearly as important as call handling, so the sql_query_async() function looked ideal, but in testing, if the DB is not responsive, the call processing gets blocked. In the example below, I'm using just a short hostname which isn't resolvable, but if I have an IP address there that isn't really assigned (for example if the DB server is down), the same blocking behavior is exhibited, but the error is slightly different (can't connect instead of unknown host). Here's my config. This specific simple config was tested on Kamailio 5.5, but I observe similar behavior (in a more complicated config) in the dev branch as well. - - - - #!KAMAILIO async_workers=2 loadmodule "db_mysql" loadmodule "sqlops" loadmodule "sl" loadmodule "xlog" loadmodule "pv" force_rport=yes ## Host sql does not resolve. modparam("sqlops","sqlcon","cb=>mysql://kamailio:kamailiorw@sql/kamailio") ## But start anyways. modparam("sqlops", "connect_mode", 1) request_route { sl_send_reply("100", "Trying"); ## Prevent re-transmissions from the UAC xlog("L_NOTICE", "Time before sql_query_async(): [$TV(Sn)]\n"); sql_query_async("cb", "INSERT INTO table1 (col) VALUES ('val')"); xlog("L_NOTICE", "Time before sl_send_reply(): [$TV(Sn)]\n"); sl_send_reply("404", "Not found"); } Here's the log if I send a simple INVITE with SIPp. Note the timestamps 9(16) NOTICE: <script>: Time before sql_query_async(): [1645574694.546165] 9(16) ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Unknown MySQL server host 'sql' (-3) 9(16) ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool 9(16) ERROR: sqlops [sql_api.c:190]: sql_reconnect(): failed to connect to the database [cb] 9(16) ERROR: sqlops [sqlops.c:232]: sql_check_connection(): failed to connect to database 9(16) ERROR: sqlops [sqlops.c:264]: sql_query_async(): invalid connection to database 9(16) NOTICE: <script>: Time before sl_send_reply(): [1645574699.552440] Ben Kaufman Sr. VoIP Engineer - BCM One +1.612.735.9309 bkaufman(a)bcmone.com<mailto:bkaufman@bcmone.com> 24 hour client support: +1.855.639.6300 [https://www.bcmone.com/files/bcmone-logo.png]<https://www.bcmone.com/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://www.facebook.com/bcmonenyc/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://twitter.com/bcm_one/> [https://www.bcmone.com/files/linkedin-icon-circle.png] <https://www.linkedin.com/company/business-communications-management/> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

4 4

kamailio :?==?utf-8?q? ROUTE[AUTH] - IPAUTH - use case
by Youssef Boujraf 23 Feb '22

23 Feb '22

Dear, I enabled two softphone with ADDRESS. - codec1 : ip 192.168.30.241/32 5060 - codec2 : ip 192.168.30.242/32 5060 I enable one softphone with SUBSCRIBER - codec3 : is registered (login/passwd) codec4 is not part of the domain, is not subriber and address. Kamailio proxy - kamailio : ip 192.168.30.240/32 5060 In the kamailio.cfg ROUTE[AUTH]# IP authorization and user authenticationroute[AUTH] {#!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; }#!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return;} Regarding the condition #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; }#!endif USE CASE : If codec4 (not registered or same domain) tries to INVITE codec3 (in subscriber) the codec, an INVITE MESSAGE is sent and the codec3 could ACK. - But I don't want to allow the INVITE, I 'd like to not allow the codec4 to reach codec3. - But if codec1 or codec2 tries to reach codec3, kamailio will allow the INVITE. Best Regards, -- Youssef BOUJRAF

1 0

Issue with Authenticating outbound Call
by Victor ku 22 Feb '22

22 Feb '22

Hi, I'm new to sip and kamailio, I have recently setup a docker container based on kamailio/kamailio:5.5.0-stretch to check things out. While I was trying to authenticate an outbound call to a 3rd party sip provider from my kamailio container, I noticed there are times that the 2nd invite with the authorization header that's supposed to be sent from the failure route doesn't get sent out. There's no error recorded in the log nor does wireshark picking up any reply to sip phone or message to 3rd party sip provider. Anyone know what might have caused this? Some other things to note: Kamailio does sent out the INVITE with authorization header and authenticated successfully when my sip phone sent the sip request with FROM ip set to the 3rd party sip server's ip. When I change the sip phone to send INVITE with FROM Domain using kamailio container's ip, I'm able to see header is being built from the trace but no INVITE with authorization header is being sent. Changing the FROM uri using $fu or uac_replace_from() does help either. ---------------------------------------------------------------------------------------------------------------- > > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} uac [auth_hdr.c:408]: > build_authorization_hdr(): hdr is <Authorization: Digest > username="xxxxxxxx", realm="xxx.xxx.xxx.xxx", nonce="07f4a5779568", > uri="sip:+0123456789@xxx.xxx.xxx.xxx", qop=auth, nc=00000001, > cnonce="3782993246", response="d5ebd0ef00d44db2292c7b8508fa23bc", > algorithm=MD5 > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1026 a=16 > n=if > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1021 a=24 > n=t_relay > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} <core> [core/msg_translator.c:1797]: > check_boundaries(): no multi-part body > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1030 a=2 > n=exit > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483650 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1564]: > t_should_relay_response(): rps discarded - uas status: 100 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1924]: relay_reply(): > reply status=2 branch=0, save=0, relay=-1 icode=0 msg status=401 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:596]: > receive_msg(): reply-route executed in: 1753 usec > kamailio-kamailio-1 | 14(22) DEBUG: dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483652 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:606]: > receive_msg(): cleaning up ---------------------------------------------- package capture ---------------------------- 192.168.0.165 xxx.xxx.xxx.xxx SIP/SDP 1383 Request: INVITE > sip:+0123456789@ xxx.xxx.xxx.xxx | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 585 Status: 100 Trying | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 820 Status: 401 Unauthorized | > 192.168.0.165 xxx.xxx.xxx.xxx SIP 396 Request: ACK sip:+0123456789 > @ xxx.xxx.xxx.xxx | ------------------------------------------nothing more till sip phone cancel ----------------- # ----------------- below is failure route config --------------------- > failure_route[MANAGE_FAILURE] { > xlog("~~~~~ MANAGE_FAILURE ($rm) ($rs-$rr) (from $fu) (to $tu) (req :$ru) > (dest $du)\r\n"); if (t_is_canceled()) { exit; } xlog("Checking status code\r\n"); #!ifdef WITH_UAC if(t_check_status("401|407")) { if (isflagset(7)) { t_reply("503","Authentication failed"); exit; } $avp(auser) = "xxxxxxxxxx"; $avp(apass) = "xxxxxxxxxx; if(uac_auth()) { setflag(7); xlog("after uac_auth $rm (from $fu)(to $tu)(req $ru)(dest $du)\r\n"); if(!t_relay()) { xlog("Failed to relay ($rm) ($rs-$rr) (from $fu)(to $tu)(req $ru)\r\n"); t_reply("500","Error"); } } else { xlog("uac_auth() not available\r\n"); t_reply("503","Authentication invalid"); } exit; } #!endif } > # ----------- end failure route config --------------------

2 2

reinvite with 2 video lines in SDP
by Filippo Graziola 22 Feb '22

22 Feb '22

Hi all, I'm facing a strange behaviour from a client (iOS application based on pjsip library) which is doing as follow when receiving a call (on phone locked): - accept the call rejecting video (m=video 0...) in SDP - reinvite to negotiate again video media Specifically the reinvite passing through kamailio (+rtpengine) has this SDP body after rtpengine offer (the client use TLS so I can't see the original message): v=0 o=- 3854443139 3854443142 IN IP4 PUBLIC_RTP_ENGINE_IP s=pjmedia b=AS:386 t=0 0 a=X-nat:0 m=audio 46734 RTP/AVP 8 101 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:96000 a=ssrc:1456418416 cname:441b90ac6db72f22 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:46735 m=video 0 RTP/AVP 96 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 m=video 46776 RTP/AVP 98 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 a=rtpmap:98 H264/90000 a=fmtp:98 profile-level-id=42e01e; packetization-mode=1 a=sendrecv a=rtcp:46777 from this I can understand that the client is doing something strange, because there is still the rejection line (I know that this question should go to the pjsip guys) but I would also like to know if it is possible to "clean" this sdp by kamailio. A few clients receiving this sdp are answering with only audio. Reading the docs, sdpops/textops modules allow modifying lines by prefix or media, but till now I have no luck to exclude only the line "m=video 0 RTP/AVP 96" and his attributes without touching the correct video request. Could someone please give me some hints on it?

2 2

Enable switch redirect to other SIP server source
by bayu p 21 Feb '22

21 Feb '22

Dear Kamailio Expert, Could you help me switching redirect sip server to another sip server ? As described below architecure. I have some uac_redirect module to implement , but kamailio cannot perform running states. Pleas help to reditect sip server to another sip server. Thanks & Best regards,

2 1

kamailio success story
by Valentin Christoph 21 Feb '22

21 Feb '22

Hi all, We had to connect to an SBC, which was not compliant to RFC 3261. It was urgent, because management was waiting for first call. Within minutes, I could implement a workaround in kamailio routing configuration: ## CV experimental 2022-02-17: fake Supported HF if (is_present_hf("Require") && !is_present_hf("Supported")) { append_hf("Supported: $hdr(Require)\r\n", "Require"); msg_apply_changes(); } Thank you, kamailio! All the best Christoph

2 1

SIP server port service cannot be changed from default value (5060)
by bayu p 18 Feb '22

18 Feb '22

Hello Experts, Please help me to change port service from default 5060 port into other port due to security reason. Since I have change port in kamailio.cfg file, but services cannot fly normally that shown client unable login online (status not online). [image: image] <https://user-images.githubusercontent.com/76675592/154027407-24ab423b-302b-…> Besides, I need to run the kamailio services after install default kamailio, I have not found where procedure found in this installation process on github.com/kamailio. Could you please share if there another steps of compiling Kamailio source, install, deploy process, and running command for this specific installation process? Thanks Gb

4 9

Kamailio v5.5.4 Released
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, Kamailio SIP Server v5.5.4 stable release is out. This is a maintenance release of the latest stable branch, 5.5, that includes fixes since the release of v5.5.3. 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 v5.5.4. For more details about version 5.5.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/02/kamailio-v5-5-4-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

2 2

Releasing Kamailio v5.5.4
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, I am considering to release Kamailio v5.5.4 later this week, likely on Thursday, Feb 17, 2022. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

Info: sipexer v1.0.0 - sip cli tool
by Daniel-Constantin Mierla 16 Feb '22

16 Feb '22

Hello, I want to announce the availability of sipexer v1.0.0 - a sip cli tool that can facilitate testing and monitoring of SIP signalling systems. It tries to have a modern approach, with a flexible templating system, supporting both IPv4 and IPv6 with all the common transport layers, respectively UDP, TCP, TLS and WebSocket (for WebRTC). The project can be found at: * https://github.com/miconda/sipexer It is written in Go language for better portability, binaries for Linux, MacOS and Windows are made available for download in the release page: * https://github.com/miconda/sipexer/releases/tag/v1.0.0 Among its features: * send OPTIONS request (quick SIP ping to check if server is alive) * do registration and un-registration with customized expires value and contact URI * authentication with plain or HA1 passwords * set custom SIP headers * template system for building SIP requests * fields in the templates can be set via command line parameters or a JSON file * variables for setting field values (e.g., random number, data, time, environment variables, uuid, random string, …) * simulate SIP calls at signalling layer (INVITE-wait-BYE) * respond to requests coming during SIP calls (e.g., OPTIONS keepalives) * send instant messages with SIP MESSAGE requests * color output mode for easier troubleshooting * support for many transport layers: IPv4 and IPv6, UDP, TCP, TLS and WebSocket (for WebRTC) * send SIP requests of any type (e.g., INFO, SUBSCRIBE, NOTIFY, …) One usage example that could ease the testing of Kamailio is initiating registrations or simulating calls over WebSocket without the need of having a JavaScript soft phone application running in a web browser. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

10 12

allow_source_address_group in failure route
by Marrold 16 Feb '22

16 Feb '22

Hello, I'm using Kamailio as a proxy between several voice gateways and if the call fails for whatever reason (far end is busy etc) I am forwarding to an Asterisk box to play an announcement. Within the failure route I am using allow_source_address_group() to ascertain if the failure response was from a Voice Gateway or Asterisk itself, however it appears to return the group for the first branch when the reply is from the second branch to Asterisk. To handle the failure I am adding some headers, and setting a new destination indirectly with ds_select_dst before t_relay. I am using 5.5.3 on Debian 11, using KEMI / Python. Does anyone have any ideas? Thanks Matthew

2 1

Missing contact in 180 ringing causes wrong routing
by Xuo Guoto 15 Feb '22

15 Feb '22

Hello all, I have a kamailio ( kamailio 5.5.3 (x86_64/linux)) with asterisk (Asterisk 16.2.1~dfsg-2ubuntu1) with following architecture ┌─────────────┐ ┌─────────────┐ ┌─────────────────────┐ │ Asterisk ├───────┤ Kamailio │ 188.xxx.59.171 ───────── 103.xxx.xx.134 │ Linphone │ └─────────────┘ └─────────────┘ └─────────────────────┘ 10.10.1.13 10.10.1.9 In case the diagram is not displayed correctly its pasted at https://pastebin.com/b0UtE4Lj Registrations are handled by Kamailio and calls from linphone are routed to Asterisk. Similarly all calls originating in asterisk are routed over to Kamailio which in turn looks up AOR and routes the calls over to Linphone. Along with Linphone I also have a WebRTC endpoint at Kamailio. When calls are sent to WebRTC from asterisk, it gets correctly routed over to Asterisk, but if the client is Linphone the ACK of 200OK is not routed correctly to Linphone. I get this error in the logs: [core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced tls:188.xxx.59.171:5061, to udp:188.xxx.59.171:5060) Digging into the difference between WebRTC and Linphone, I could see that in 1800 Ringing packet from WebRTC has a Contact headers while its missing in the packet from Linphone This is the 180 ringing from WebRTC: tag: rcv pid: 695615 process: 17 time: 1644860126.209155 date: Mon Feb 14 23:05:26 2022 proto: wss ipv4 srcip: 103.xxx.xx.134 srcport: 52508 dstip: 188.xxx.59.171 dstport: 443 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/WSS 188.xxx.59.171:443;branch=z9hG4bK730a.2cc7461b72f0c895e7cce5a0cedce4f7.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Content-Length: 0 While this is from Linphone tag: rcv pid: 695614 process: 16 time: 1644860097.970935 date: Mon Feb 14 23:04:57 2022 proto: tls ipv4 srcip: 103.xxx.xx.134 srcport: 25843 dstip: 188.xxx.59.171 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/TLS 188.xxx.59.171:5061;branch=z9hG4bKb5b1.d3231367383d07b1063098dcb35cc2b0.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPjd16f1379-1be7-448e-894a-c42d96f63146 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=cf7a7770-529d-4450-8bed-880382dce1dd To: <sip:user_jbjp@10.10.1.9>;tag=2ooYDia Call-ID: 8a025999-b096-4fe0-a623-9f6e6a69b8ae CSeq: 27838 INVITE User-Agent: Linphone Desktop/4.3.2 (Ubuntu 20.04.3 LTS, Qt 5.12.12) LinphoneCore/5.0.49 Supported: replaces, outbound, gruu Record-route: <sip:188.xxx.59.171:5061;transport=tls;r2=on;lr=on;did=3b9.b1b2> Record-route: <sip:10.10.1.9;r2=on;lr=on;did=3b9.b1b2> Content-Length: 0 When the 180 ringing is passed on to the Astrerisk it has the contact header modified with alias. tag: snd pid: 695615 process: 17 time: 1644860126.210189 date: Mon Feb 14 23:05:26 2022 proto: udp ipv4 srcip: 10.10.1.9 srcport: 5060 dstip: 10.10.1.13 dstport: 7000 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;alias=103.xxx.xx.134~52508~6;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Content-Length: 0 Now, I am not sure if this is the reason for routing the packets to wrong destination, but this is all I could figure out. Any clue to get this resolved will be much appreciated. My Kamailio config is pasted at https://pastebin.com/qyRS0usw with regards, X. Sent with ProtonMail Secure Email.

1 0

get kamailio sip processes id
by Patrick Karton 14 Feb '22

14 Feb '22

hello, as we are all aware kamailio create a lot of processes for many tasks. im using kamailio in kubernetes and i would like to monitor only sip processes with top command. i only need a way to get only kamailio sip processes id (udp receiver that handle sip request) . but i dont knwo how to get it. top -c -p id1,id2,id3,id4 Thanks

3 5

SEMS license with kamailio and rtpengine
by Patrick Karton 14 Feb '22

14 Feb '22

Hello im using kamailio and rtpengine to build a custom SBC. i was using freeswitch as my b2bua but i noticed a lot of issues when cps and simultaneous increased so i tried sems and i have better results. But i do not really understand sems license so i want to know if i can use it. i read from sems github "It is licensed under dual license terms, the GPL (v2+) and proprietary license. This program is released under the GPL with the additional exemption that compiling, linking, and/or using OpenSSL is allowed. For a license to use SEMS under non-GPL terms, please contact FRAFOS GmbH at info(a)frafos.com" so my question is if i want to use sems as b2bua with kamailio and rtpengine and make profit of it do i need to purchase a license each time i want to sell it to a customer ? Thanks.

7 39

Kamailio server behind NAT
by Tihomir Culjaga 13 Feb '22

13 Feb '22

Hi all I'm wondering what is best practice to configure kamailio for a setup shown below. Carriers only support interconnection over a private network so I need to terminate every separate network on kamailio machine (and on RTPEngine) . That will turn very quickly into an administration nightmare as we add more and more carriers. My plan is to terminate carrier private networks on a router and NAT the traffic to kamailio where it leads me to a kamailio behind NAT setup. Well, in theory set_advertised_address and set_advertised_port per message can do the trick... at least I hope. Also, some carriers require a SIP registration to activate SIP trunks. I do that using UAC module (remote registration) but here again I got a problem with contact address as you can set only one contact address (modparam("uac", "reg_contact_addr", "10.1.1.11:5060")). /--RTP 10000-19999-- RTPEngine1 -----RTP------+-- IPPBX1 carrier1 --- NAT ---| | | | | | carrier2 --- NAT ---+-----SIP 5060------ kamailio ----SIP---- Dispatcher | | | carrier3 --- NAT ---| | | \--RTP 20000-29999-- RTPEngine2 -----RTP------+-- IPPBX2 So bottom line, I'm wondering what would be the best practice for such a scenario ? That would spare me from hitting many pitfalls. Would really appreciate a hint. Regards, Tihomir.

3 3

Vo5G with open5gs
by Hornbuckle, Mark A 11 Feb '22

11 Feb '22

Hello, I have been trying to find documentation on Kamailio IMS and Vo5G support for open5gs. I wanted to ask you directly if Kamailio IMS does support Vo5G with open5gs? Thank you

1 0

resending same message from UE after receiving RP-ACK
by Eugene B 11 Feb '22

11 Feb '22

Hi, I trying to use smsops module as SMSC. When I sending message from UE I see re-sending same message after each RP-ACK(~10 attempt). What could be the problem? Thanks! pcap in attach. kamailio 5.3.2 -- *Yours faithfully,* *Eugene B.*

2 1

kamilio open5gs 5G SA compatible?
by Sandland, Seth P 11 Feb '22

11 Feb '22

Were looking to test a kamilio on Open5gs 5g stand alone ( non lte, non NSA) can you conform or provide documents or install instructions on how this is accomplished. I can not find much of any details if Kamilio support Open5Gs 5g SA and what features it does support on a 5g SA core. Seth Sandland Wireless Communications IT Engineer Pacific Northwest National Laboratory Tel: (509) 375-6842 Email: seth.sandland(a)pnnl.gov<mailto:seth.sandland@pnnl.gov> www.pnnl.gov<http://www.pnnl.gov/> [A close up of a logo Description automatically generated]

1 0

Publish XML and xpath
by Igor Olhovskiy 10 Feb '22

10 Feb '22

Hello, I'm trying to get state of endpoint from PUBLISH message in a most simple way using xmlops module Trying to make this $xml(body=>doc) = $rb; log("L_ERR", "$rU is now in state $xml(body=>xpath:/dialog-info/dialog/state)\n"); XML in PUBLISH itself <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:88881@dev.company"> <dialog id="v6qmEYf5yI" direction="initiator"> <state>confirmed</state> </dialog> </dialog-info> Actual text in log is ... <script>: 88881 is now in state Means it's empty. xmlops module is loaded with no parameters. Tried also string $xml(body=>xpath:/dialog-info/dialog/state/text()) -> same result What am I missing here? kamailio 5.4.6 -- Thanks in advance, Igor

2 5

KEMI KSR.async Python3
by Ahmed Moubayed 10 Feb '22

10 Feb '22

Hi, I'm trying to use the async module from a python3 Kemi script but it conflicts with the keyword async. Any suggestions on how it can be used? Is there an alternative to KSR.async.route()? Thanks, Ahmed

1 0

CDR preserve original call direction
by Ahmed Fouad 10 Feb '22

10 Feb '22

Hi, I've enabled ACC and CDR logging and it's working fine for the logging but I noticed something wrong in the CDR table , the call source and destinations are depending on the BYE message not the initial invite dialog ACC table +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | id | method | callid | time | src_ip | src_domain | src_user | dst_ouser | dst_domain | +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | 216 | BYE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:57:51 | x.x.x.x | domain.com | DST_CLID | SRC_CLID | domain.com | | 215 | INVITE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:55:17 | x.x.x.x | domain.com | SRC_CLID | DST_CLID | dst_domain.com | The INVITE is the correct one with SRCIP x.x.x.x and domain.com but the BYE reversed the order since the the DST_CLID who sent the BYE so it's now logged as the source and the final CDR log looks like +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | cdr_id | src_username | src_domain | dst_username | dst_domain | call_start_time | call_end_time +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | 195 | DST_CLID | domain.com | SRC_CLID | domain.com | 2022-02-09 06:55:12 | 2022-02-09 06:57:51 | the CDR as shown is reversed regarding the source/dst username I tried to enable the direction ACC direction flag but didn't help and here is my current paramters modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdrs_table", "cdrs") Can you please advise how to configure ACC/CDR to log the CDR with the correct initial INVITE call flow regardless of the BYE direction ?

3 5

REGISTER with http_async_client ?
by Cyril Ramière 10 Feb '22

10 Feb '22

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); } } ##### END SNIP Best regards!

4 15

Kamailio doc request
by Mustafa Rahman 10 Feb '22

10 Feb '22

2 1

Failure route with t_relay()
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I am trying to make a re-transmission mechanism from the failure route, I've tried to use $dp and $du as a relay, but variables are not accepted in t_relay(), When using ip and port straight forward t_relay("144.0.0.0", "1111"), I am getting: [t_fwd.c:1752]: t_forward_nonack(): no branches for forwarding [tm.c:1753]: _w_t_relay_to(): t_forward_noack failed Is there a way to re-transmit the call to the same carrier again? Thanks in advance, and have a great day, Max Kemel

2 1

Failure route mechanism with counter
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I would like to use a failure route with a retry mechanism, which provides a MAX_NUM_OF_RETRIES (max number of times reaching the failure route), Which can be achieved by a simple global counter, from what I've seen, the Counters module doesn't provide a GET function in script, What is the easiest solution for this kind of behavior? Thanks a lot, Max Kemel

2 1

Timers firing too soon
by Alberto Izquierdo 04 Feb '22

04 Feb '22

Hi all, I'm using Kamailio timers to execute a series of calls with a certain delay. I have found out that from time to time, the timers are triggered too soon, sometimes they are triggered 2/3 seconds before they should. I have read in the documentation that it is normal to have a bit of delay in some of the timers, but it doesn't mention anything about this behavior. Is this normal? Thank you.

1 0

Debian kamailio.init vs. kamailio.service
by Juha Heinanen 04 Feb '22

04 Feb '22

By comparing pkg/kamailio/deb/bullseye dir kamailio.init and kamailio.service files, it appears to me that they are not equal alternatives. Is this correct interpretation? -- Juha

1 0

Homer7 and Kamailio 5.3
by Angelo Sipper 03 Feb '22

03 Feb '22

Hi to all, We have a Kamailio 5.3 that sends all traffic to a homer7 using siptrace module with parameters force_send_sock and hep_mode_on,1 and hep_version,3 and trace_mode,1. Everything works fine getting all traffic to Homer7 but, in case Homer7 is not reachable, Kamailio then no longer accepts traffic or registrations and it is like it crashes. When we revive Homer7 and is it again reachable accepting traffic, Kamailio then works again fine without any restart or actions. Any ideas why this may be happening? Is there anything we need to add to the config to avoid this behavior as having Homer7 now is a single point of failure for Kamailio? Thank you! Kind Regards, Angelo

6 11

uac problem with 5.3.8
by Richard Robson 03 Feb '22

03 Feb '22

we have been running 5.3.4 with the uac_auth_mode() function ans it has been working we have upgraded to 5.3.8 and with the same script we are getting an error with the uac module saying it was unable to find the cmd auc_auth_mode I have tried this with uac_auth as well with the same result. it looks like the uac module is not loaded or is not loading but running a core.modules reports the uac module can anyone shed any light on this problem? is it a version pecific problem?

3 4

Updating $du for uac REGISTERs in event_route[tm:local-request]
by Raffaella Seuß 03 Feb '22

03 Feb '22

Hello all, I tried to update $du for REGISTERs sent by the uac module in event_route[tm:local-request]. (I want to overwrite the auth_proxy under some conditions.) However, the requests are still sent to the auth_proxy defined in uacreg. Am I missing something here, why doesn't it work? It's basically just: event_route[tm:local-request] { if (is_method("REGISTER")) { # ... $du = "sip:PROXY_IP:PROXY_PORT"; } } My Kamailio version is 5.5.2. With kind regards, Raffaella Seuß foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: raffaella.seuss(a)foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon? Unter https://www.foncloud.net/wissen finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services. Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

1 0

Siremis configuration
by Tim Mastarone 02 Feb '22

02 Feb '22

Hi, I am using Siremis v 5.3 with kamailio. My users can log into the Siremis site just fine but the field to enter their email address is not appearing. My sql queries seem to be working besides that. Does this feature need to be enabled? I would like to use email to confirm the user login with a confirmation code. (mysql Ver 15.1 Distrib 10.1.48-MariaDB, for debian-linux-gnu (x86_64) using readline 5.2) Thank you, [cid:image001.png@01D81830.52C21EA0]<http://www.vikingelectronics.com/> TIM MASTARONE Associate Engineer 715-386-8861 x208 vikingelectronics.com<http://www.vikingelectronics.com/>

1 0

two SIP proxies
by Uwe Sigleur 31 Jan '22

31 Jan '22

2 1

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