- sr-users - kamailio.org

Multiple IPs
by Oz Mortimer 03 Nov '17

03 Nov '17

Hi all, I have Kamailio set as an SBC with freeswitch behind - that all works fine. Freeswitch can send calls out VIA Kamailio to external IP address’, but if I try sending a call to another IP hosted by Kamailio the seems to go nowhere… Supplier -> Kamailio -> Customer1 -> Kamailio -> Customer2 The Kamailio server has multiple assigned IPs, which are used to multi-tennant - but I cant for the like of me to get Tennant to tenant calls to work (via freeswitch) Any pointers?

2 13

onreply_route block not processing replies
by José Seabra 03 Nov '17

03 Nov '17

Hi there, I'm facing an issue regarding with replies coming to kamailio that aren't processes inside of onreply_route block. Anyone here can help me understand why these replies (1XX) aren't entering on onreply_route bock? is there any situation already identified on Kamailio that can originate this behavior? The Kamailio version used is 5.0.1. Thank you Regards -- José Seabra

2 3

Branching Kamailio v5.1
by Daniel-Constantin Mierla 03 Nov '17

03 Nov '17

Hello, I am planning to create the got branch 5.1 (to be used for v5.1.x series) by end of next week (likely on Friday, Nov 10, 2017). After that, the master will be open for new features and fixes will need to be backported to 5.1. Should anyone want a different schedule, reply to the mailing lists and propose your plans. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

WebRTC error code 488 in iOS application
by Denys Pozniak 03 Nov '17

03 Nov '17

Hello! Probably this case is not connected Kamailio directly, but it can be somebody will point me in the correct direction :) We have VoIP solution: FreeSWITCH -> Kamailio + Rtpengine -> WebRTC Chrome. Everything works Ok except iOS application based on React Native WebRTC ( https://github.com/oney/react-native-webrtc) + JsSIP. It answers error 488 on incoming INVITE (wSDP): INVITE SDP to iOS: *......* *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* iOS console log: *11:55:23.670 browser.js:133 JsSIP:WebSocketInterface send() +10ms* *11:55:23.670 browser.js:133 JsSIP:RTCSession session progress +5ms* *11:55:23.671 browser.js:133 JsSIP:RTCSession emit "progress" +1ms* *11:55:23.671 webrtc.js:108 WebRTC.on progress: local* *11:55:23.679 browser.js:133 JsSIP:RTCSession emit "peerconnection" +8ms* *11:55:23.680 webrtc.js:89 WebRTC.on peerconnection* *11:55:23.680 browser.js:133 JsSIP:RTCSession emit "sdp" +1ms* *11:55:23.689 browser.js:133 JsSIP:Transport send() +20ms**11:55:23.689 browser.js:133 JsSIP:Transport sending message:* *SIP/2.0 488 Not Acceptable Here* *11:55:23.701 debug.js:115 JsSIP:ERROR:RTCSession emit "peerconnection:setremotedescriptionfailed" [error:Error: Failed to set remote offer sdp: Failed to create channels.* I found similar issue on thier git https://github.com/oney/react-native-webrtc/issues/293 and main solution is to struct SDP in way like: *a=group:BUNDLE audiom=audio PORT RTP/SAVPF N M Ka=mid:audio* How to modify SDP on Kamailio side (after Rtpengine modification) like below? *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *a=group:BUNDLE audio* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=mid:audio* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* BR, Denys

1 1

User Registration with location awareness
by Daisy Stevens 03 Nov '17

03 Nov '17

Hi Guys Please forgive me if I seem to asking stupid questions but I am a complete dummy at the moment regarding this software. I would love to be able to set up two kamailio servers with the idea of expansion in the future to handle SIP registrations, however having a two server model how would I get the calls to route to another user that is registered on the other kamailio box? Kind Regards Daisy -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

2 1

msilo Content-Type for picture and video?
by sscc 03 Nov '17

03 Nov '17

Hello kamailio folks i am successfully storing and dumping messages for off line users using msilo content type text/plain. i want to store off-line messages for pictures and videos. my particular queries are 1) is msilo store picture and videos? (i have observed blob in silo database table which is used for pictures. 2) in case it does what will be declared in content type hdr parameter? any help in this regard will highly be appreciated. regards, sscc -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Re: [SR-Users] Reading the P-Assserted-identity header
by Gertjan Wolzak 02 Nov '17

02 Nov '17

Alex and Daniel, Thank you for the feedback, the tip from Daniel helped me pinpoint the problem. And the the one from Alex showed that. The problem was between keyboard and chair, I should have seen that the SBC removed the PAI. Once I changed that, the PAI arrived at Kamailio and manipulation was a breeze. Thanks guys.

1 0

Tcp socket error with active/active kamailio schema
by Donat Zenichev 02 Nov '17

02 Nov '17

Hi guys. I'm trying to realize active/active schema with kamailio. Topology is as following: active/active kamailio -> routing server (asterisk) Kamailio servers work as registrar/location servers and sbc. Asterisk server performs routing functions for calls. We use sip / sips / ws / wss for signaling. Schema built on aws platform and both kamailio have virtual ip address. So when first kamailio crashes, second one assigns ip address of first kamailio. And vice-versa. Both kamailio listens on both shared addresses all the time (with help of sysctl non lolcal bind function). Problem occurs when first kamailio goes down and second kamailio gets ip address of first one. Asterisk tries to send in-dialog requests (for dialogs that were established on first kamailio), request goes to second kamailio and asterisk gets a socket error. And this is proper behaviour I think, because first kamailio established tcp connection with asterisk from certain port, that is not listened on second kamailio. So that, I think that the solution is to use force_send_socket function, but I'm not confident about it. If anyone has ideas, please send your advices. Thanks in advance. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Kamailio not calling each time
by hanoulia salameh 02 Nov '17

02 Nov '17

Hello, I have a problem with kamailio (v4.4), when i try to call from a sip phone through kamailio, the call is interrupted and hang up automatically but after 4s it continues to retry (I can see in the log file Session Progress) and goes through the asterisk (certified/13.13-cert4) and establishes a normal call (It rings at the client side) while it's still disconnected from the sip phone side But when i try again it works fine sometimes. Am I missing something in my configuration file? What can I show in order to help me? Regards

2 2

Kamailio Realtime Integration with Asterisk
by Floimair Florian 02 Nov '17

02 Nov '17

Hi! I found a few tutorials from Daniel on this topic , the latest being https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astd… . Has anyone ever tried this with a more recent Asterisk version (13 or newer) and using chan_pjsip instead of chan_sip? I'd be willing to update the tutorial once I get it running but I would like to avoid going through obstacles that were already handled ;-) My desired setup is 2 Asterisk boxes (which share configuration, useres and registrations via DB) behind a Kamailio as Load-balancer (got that working already). With best regards Florian Floimair COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 f.floimair(a)commend.com<mailto:f.floimair@commend.com> http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg

1 0

Mutual TLS with Skype for Business 2015
by Francisco Valentin Vinagrero 02 Nov '17

02 Nov '17

Hello, I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster. I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file) [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs [client:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved. The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate. As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this? Thanks, Francisco.

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Reading the P-Assserted-identity header
by Gertjan Wolzak 01 Nov '17

01 Nov '17

Hello Kamailions, I receive anonymous calls where the PAI has a prefix which I need to strip before forwarding the call to a proxy. I am trying to read the the header with $hdr(P-Asserted-Identity) and I tried the $ai pseudovariable, but I get an empty reply back every time. xlog("PAI is $ai"); results in ; "Nov 1 16:19:55 proxy04 /usr/local/sbin/kamailio[14814]: ERROR: <script>: PAI is <null>" Same for xlog("PAI is $hdr(P-Asserted-Identity). This works for the Privacy header but not the P-Asserted-Identity header. Hope someone can tell me what I am doing wrong.... Rgds, Gertjan

3 2

Record-Route IP Value
by Soltanici Ilie 31 Oct '17

31 Oct '17

Hi All, Having the following Header: Record-Route: <sip:192.168.1.1;lr;did=637.07c7c2d7> How can i extract only IP value from this header? Thank You.

2 3

module version mismatch for /usr/local/lib64/kamailio/modules/kex.so
by Abdul Basit 31 Oct '17

31 Oct '17

Hi, My kamailio installation with 5.0.3 was working fine. I update kamailio source to 5.0.4 and make all; make install I started receiving error below: /etc/init.d/kamailio start [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(7426) ERROR: <core> [core/sr_module.c:431]: version_control(): module version mismatch for /usr/local/lib64/kamailio/modules/kex.so; core: kamailio 5.0.3 (x86_64/lin[FAILmodule: kamailio 5.0.4 (x86_64/linux) ... failed! failed! kamailio.cfg is same as earlier. I removed kamailio sources and modules folder /usr/local/lib64/kamailio/modules/ I also dropped old kamailio postgres database and created new one after fresh source install of version 5.0.4 from git. # git clone https://github.com/kamailio/kamailio kamailio # cd kamailio # git checkout -b 5.0 origin/5.0 But this error persists. Whats the resolution? Is this a bug? -- regards, abdul basit

2 2

Source comparisons by DNS hostname
by Alex Balashov 31 Oct '17

31 Oct '17

Hi, Is there a reasonable way, in Kamailio, to determine whether the source of a given SIP message is equivalent to the IP address to which a given DNS hostname resolves? i.e. if(src_ip == 'fqdn.domain.xyz') but not src_ip, obviously. I can't find any core keywords that would seem to fit the bill for a composite comparison. Any other options? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Radius accounting and cdr's
by Jan Hazenberg 31 Oct '17

31 Oct '17

Hi all, I have a question regarding radius accounting. I would like to have a config where i store local cdr records (and not the ACC records) and send accounting to radius as well. I have the following config: # ----- acc params ----- /* General settings */ modparam("acc", "log_level", 0) modparam("acc", "log_facility", "LOG_LOCAL2") /* Enable cdr's */ modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_log_enable", 1) modparam("acc", "cdr_start_on_confirmed", 0) modparam("acc", "cdr_facility", "LOG_LOCAL2") modparam("acc", "cdr_on_failed", 1) /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) /* Accounting triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc_radius", "radius_flag", FLT_ACC) modparam("acc_radius", "radius_missed_flag", FLT_ACCMISSED) modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") This works but stores the raw accounting records as well. If i disable the acc log_flag it stops sending radius accounting records (while the acc_radius log_flag is still enabled). Is it possible to send only radius accounting records and disable the local logging of those records, but leave the cdr logging enabled? Or is there a option where i can set only the cdr_log_level so accounting records will only show up when i increase the debug level? Another option is to use a seperate log facility for accounting and cdr's, but it would be nice if it is possible to disable the logging of accounting records and only keep the cdr's. I'm running kamailio 5.0.3. Thanks, Jan

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Re: [SR-Users] Kamailio for Unified Communication
by Angel Elena 31 Oct '17

31 Oct '17

kamailio is a "proxy" server.... this features may be added by a media-server... freeswitch, asterisk, seems. You can put kamailio as "sip proxy" and you can add these services behind kamailio proxy -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Zarafa Webapp. -----Mensaje original----- De: Jayanth Acharya <jayachar88(a)gmail.com> Enviado: Mar 31-10-2017 10:40 Asunto: [SR-Users] Kamailio for Unified Communication Para: Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org>; > Hi, > > Can Kamailio be used to provide following broad-type of services ? > > 1. Unified Communication services > 2. Contact Center services > 3. Instant Messaging / Chat services > 4. Voicemail service > 5. Presence aware communication services > 6. Desktop sharing service > > For many/some of the above, I'd expect back2back user-agent capabilities to be > mandatory, and if I am not mistaken Kamailio doesn't support B2BUA mode. Or are > there some modules that may be used for this purpose ? > > thanks, > Jay > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > >

1 0

Kamailio for Unified Communication
by Jayanth Acharya 31 Oct '17

31 Oct '17

Hi, Can Kamailio be used to provide following broad-type of services ? 1. Unified Communication services 2. Contact Center services 3. Instant Messaging / Chat services 4. Voicemail service 5. Presence aware communication services 6. Desktop sharing service For many/some of the above, I'd expect back2back user-agent capabilities to be mandatory, and if I am not mistaken Kamailio doesn't support B2BUA mode. Or are there some modules that may be used for this purpose ? thanks, Jay

1 0

Reinvite/ACK race
by Alex Balashov 30 Oct '17

30 Oct '17

Hi, I've got a scenario like so: UA A -----> Kamailio P ----> UA B 1. UA A initiates call through Kamailio P; 2. Dialog is established and confirmed, with Record-Route; 3. UA B sends reinvite #1 through P to A; 4. UA B sends 2xx reply; 5. UA B sends end-to-end ACK for reinvite #1 and almost simultaneously sends reinvite #2. The temporal delta is between reinvite #2 and ACK for reinvite #1 on the wire is 3 ms. So, the result — for all kinds of stochastic processing and userspace scheduling type reasons — is that the reinvite is forwarded first, before the ACK. That leads to a 500 / 491 scenario UA A. The cause, as we all know, is that Kamailio's worker threads are loosely coupled, and incoming UDP datagrams are distributed directly by the kernel. Is there any general guidance on what to do with these scenarios? I looked at RFC 5407 § 3.1.4, which appears to describe a similar, but not identical scenario involving an initial INVITE and subsequent reinvite. As far as I can tell, the recommendation in that standard is "space the messaging out more in time". Switching to TCP would presumably help, since any given flow would involve a single connection to a single worker thread and the transport would guarantee ordering. However, that's not really feasible in this implementation for a host of reasons. I know Kamailio has some config locking primitives, but I am extremely wary of complex synchronisation on dialog-identifying attributes, particularly if there is a possibility that such locks could stall a worker perpetually. Any other thoughts welcome! Cheers, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 2

unexpected behavior of save from registrar module
by Vasiliy Ganchev 30 Oct '17

30 Oct '17

Hi, community! I have in my Kamailio script following block (while REGISTER processing, after auth succeed): ..... /* Save to usrloc */ lock("LOCATION_LOCK"); if (!save("location")) { unlock("LOCATION_LOCK"); sl_reply_error(); exit(); } unlock("LOCATION_LOCK"); ...... According to module documentation - if "save" - failed (for any reason) - I expected the control to be returned to the script with error code "-1" But, there are cases - when this does not happen - which lead to deadlock for workers processing REGISTERs (I never unlock("LOCATION_LOCK")): - module answers with internal error - and return 0 from "save" function: https://github.com/kamailio/kamailio/blob/master/src/modules/registrar/save… - line 1013 My use case happened due to malfunctioning device that in some moment of time: - send REGISTER update - kamailio answers 200OK - device retransmit the same REGISTER (same CSeq) - during 20 seconds (hardcoded value in registrar module) - kamailio answers 200OK (and everything is ok) - if retransmission continue more than 20s - kamailio answered 400 Bad request, and did not return control to script (as a result I had no chance to unlock("LOCATION_LOCK") from code above) Tested it on kamailio 4.2.5 and 5.0.3. as for me, possible solutions are: - return -1 instead of 0 from "save" by default - if there are some reasons not to do this - we should change documentation, and explicitly mention that there are cases - when script execution is stopped. thanks in advance! cheers -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

3 5

sanity passes invalid uri
by Juha Heinanen 30 Oct '17

30 Oct '17

I noticed that K 5.0 sanity() test passes uri that contains ` (back quote) character. Config: modparam("sanity", "default_checks", 1024) /* URI checks */ modparam("sanity", "uri_checks", 3) /* RURI, From */ xlog("L_INFO", "Checking $ru\n"); if (!sanity_check()) xlog("L_INFO", "Check failed\n"); else xlog("L_INFO", "Check passed\n"); Syslog: Oct 27 17:17:38 lohi /usr/bin/sip-proxy[31946]: INFO: Checking sip:jh@te`st.fi Oct 27 17:17:38 lohi /usr/bin/sip-proxy[31946]: INFO: Check passed According to RFC3261: hostname = *( domainlabel "." ) toplabel [ "." ] domainlabel = alphanum / alphanum *( alphanum / "-" ) alphanum Is this a bug or am I missing something? -- Juha

2 2

Remove all cpecial characters (\n,\t etc)from xml body
by Yuriy Gorlichenko 30 Oct '17

30 Oct '17

Hi. I building external presentity service for web based phones Main trouble that i have at the xml body of PUBLISH message \n \t characters and because of it can't set body as string to the redis So I trying to find a way to remove special characters form the xml body for get oneline string instead of multiline I use native .cfg file for now and, unfortunattely, can not use lua or phyton I tried $var(body)=$(rb{s.rm,$var(RemoveThisStaff)}) where $var(RemoveThisStaff) = "\n" and "\t" but it was unsuccesfull. Maybe someone will help to kno how to do it.

2 2

Issue with sipcapture
by Nathan Leyton 28 Oct '17

28 Oct '17

Hi There I am new to Kamailio and have just installed and got it up and running as a sipcapture node. It is working well and logging all the hep sip traffic in my homer database. I have an issue with rtcp extracted to the msg column is not json. It is some kind of weird encoded string (not base64), this is not readable by homer in any way. When I use nonsip_hook I am able to extract the json from the hep and put it in the msg field. However when I return 1 from the hep route on sip to go back to main route I get parsing errors like so on all the sip messages:- Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: sipcapture [sipcapture.c:2711]: nosip_hep_msg(): couldn't parse sip message Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: <core> [core/receive.c:186]: receive_msg(): core parsing of SIP message failed (178.78.120.2:44389/1) Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: <core> [core/parser/msg_parser.c:331]: parse_headers(): bad header field [(null)] Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: WARNING: <core> [core/receive.c:193]: receive_msg(): parsing relevant headers failed Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22517]: ERROR: <core> [core/parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22517]: ERROR: <core> [core/parser/msg_parser.c:675]: parse_msg(): ERROR: parse_msg: message=<HEP3#002,> It never actually enters the main route. With nonsip_hook off, I enter the main route and all my sip messages are there in homer, just missing rtcp as the data is a weird encoded string. Hope someone can help Thanks Nathan -- Nathan Leyton / Chief Technology Officer nathan(a)callhandling.co.uk / 07482 917 412 Call Handling Ltd Office: 0333 321 0888 / Fax: 0333 321 0877 Bakerloo Chambers304 Edgware Road, London, W2 1DY www.callhandling.co.uk [image: htmlsig.com] <https://htmlsig.com/t/0000001BREBFR>

2 1

fix_nated_contact and IPv6
by Sebastian Damm 27 Oct '17

27 Oct '17

Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: <sip:bob@192.168.8.132> outbound Contact: <sip:bob@1234:1234:0:1234:0:0:0:2> The correct way of transforming the Contact URI would be: Contact: <sip:bob@[1234:1234:0:1234:0:0:0:2]> This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian

4 4

Too Long Contact URI
by Soltanici Ilie 27 Oct '17

27 Oct '17

Hi All, I cannot Register some softphones because of too long Contact header when using TCP protocol. Using UDP - everything works ok and this is how the Contact header looking: sip:XXXXXXXXX@XXX.XXX.XXX.XXX:64773;rinstance=6502234ef65ea6e7 But, switching to the TCP i'm receiving the following warning: WARNING: registrar [sip_msg.c:194]: check_contacts(): contact uri is too long:[sip:XXXXXXXXX@192.168.0.1:60471;rinstance=bf4d7aae0f099bf0;transport=tcp] How can i solve this issue? Thank You.

1 1

How to send OPTIONS message by kamailio server
by 吴妙灵 27 Oct '17

27 Oct '17

Hello, I don't know how to send OPTIONS message from kamailio server to user client,can you give me same advice? Cheers, Ada

4 3

Timer function didnot trigger the routing function
by RAJA SREE RATHNA gomathinayagam 27 Oct '17

27 Oct '17

Hi, I tried using the timer function to call a function after specific time but it was not working as specified. Could you please help me with some example using timer to trigger a function. I tried it using the description given in below link. www.kamailio.org/docs/modules/devel/modules/timer.html Thanks and regards, Rathna.

2 3

Kamailio v5.0.4 Released
by Daniel-Constantin Mierla 27 Oct '17

27 Oct '17

Hello, Kamailio SIP Server v5.0.4 stable release is out! This is a maintenance release of the latest stable branch, 5.0, that includes fixes since the release of v5.0.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous 5.0.x versions are strongly recommended to be upgraded to v5.0.4. For more details about version 5.0.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/10/kamailio-v5-0-4-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.twitter.com/miconda -- www.linkedin.com/in/miconda

2 4

Set timeout before 180/183 reply arrived
by Yu Boot 27 Oct '17

27 Oct '17

Hello. Have a problem with some uplinks. They sometimes respond "100 Trying" on initial INVITE and... that's all. A caller waits for about 30 seconds and cancel a call deciding there's something wrong with our network. How to use failure route when 180/183 answers don't arrive in 5 or 10 seconds after INVITE?

2 2

Get value of a custom header param
by Arsen 26 Oct '17

26 Oct '17

Hi guys! Is it possible to get value of a custom parameter of the authorization header? I've checked selects in the textops mod and try the following: if (is_present_hf("Authorization")) { xplog("L_ERR", "custom auth $sel((a)hf_value.authorization.edatech_opaque) \n"); } Arsen Semionov www.eurolan.info cell: +442035198881

3 5

Upgrading from 1.5 to 3.x
by Brandon Shiers 26 Oct '17

26 Oct '17

I have a client that has an old Kamailio 1.5 installation. They would like to migrate and get off their old hardware and old OS. I've read the upgrade page for the 1.5.x migration to 3.0.0, are there any gotchas to be aware of other than what's there? Brandon Shiers, Network Engineer 937 West Main Street Riverton, WY 82501 307.857.6704 (o) 307.840.2366 (c) 307.856.1499 (f) brandon.shiers(a)cerento.com<mailto:Brandon.Shiers@cerento.com> [http://www.cerento.com/emailsignatures/logo.jpg]

2 1

Regarding rtimer usage under specific condition
by RAJA SREE RATHNA gomathinayagam 26 Oct '17

26 Oct '17

Hello, While going through the rtimer module we noticed that it execute the routing function repeatedly after the given interval of time. But our specification is to execute the routing function only once and timer should be enabled based on specific condition. Also there is no enable or disable option for rtimer module as in timer module, whether rtimer module can be enabled based on specific condition.If it is possible could you please suggest the solution. Thanks & Regards, Rathna.

2 1

shared dialogs
by : Paolo Visintin - Time-Net S.r.l. 26 Oct '17

26 Oct '17

Hi kamailio users, I'm wondering if there's something already done with shared dialog among more kamailio instances, in order to manage branches initalized by another kamailio instance. A ctually seems not possible (also with db mode) because kamailio is looking into caller/callee_sock and if it's different does not manage in a "standard" failover environment (master / slave) with vIP and keepalived and $fs = VIRTUAL_IP everything is working fine but in a distributed environment this could not happen as the relay is managed by a kamailio proxy Thanks in advance Paolo

4 7

I-CSCF Prefixes IMPU with + Symbol
by Ron McLeod 25 Oct '17

25 Oct '17

I am working with v5.0.3 and the IMS extensions. When I attempt to send a SIP MESSAGE from the application server towards the UE, the I-CSCF used to determine the S-CSCF for the subscriber by sending a Location-Info request to the HSS. I found that this request was failing because the IMPU (IMS Public User Identity) was being mangled by the I-CSCF, prefixing the user part with a + symbol. The example configuration file for the I-CSCF does in-fact does cause the r-uri to be prefixed with a + symbol if one is not already present. I'm wondering why since it breaks the LIR/LIA lookup. AS I-CSCF HSS | MESSAGE | | | ---------------------> | | | sip:ron@provider1.test | | | | | | | | | | LIR | | | ----------------------> | | | sip:+ron@provider1.test | | | | | | | | | DIAMETER_UNKNOWN_USER | | | <---------------------- | | | | | | | | 403 | | | <--------------------- | | | FORBIDDEN | | route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } ... Thanks, Ron

1 0

Testing for user=phone with siputils uri_param() doesn't work
by George Diamantopoulos 25 Oct '17

25 Oct '17

Hello, In my script I've had to test for the user=phone parameter in URIs. I thought uri_param() from siputils would be handy for this operation on the RURI, so I did: if ( !uri_param("user","phone") ) { add_uri_param("user=phone"); } Unfortunately this wouldn't work (uri_param returns -1), and I end up with double user=phone params in the RURI. I was able to work around this by changing the test to: if ( !($(ru{uri.uparam}) == "phone") ) { add_uri_param("user=phone"); } But I wanted to ask if this is intentional or if I should file a bug or something. Thanks. George

3 3

Planning Kamailio v5.0.4
by Daniel-Constantin Mierla 25 Oct '17

25 Oct '17

Hello, I am considering to release Kamailio v5.0.4 later this week, likely on Wednesday or Tuesday. If you are aware of any issue in branch 5.0 which is not yet reported on github.com/kamailio/kamailio issues tracker, then do it as soon as possible to have a chance to look at it. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

3 5

usrloc module does not work with db_cluster
by Uri Shacked 25 Oct '17

25 Oct '17

Hi, I started using db_cluster module. DB are master-slave replication both ways. #!define DBURL "cluster://kamailio" when i stop the primary mysql server, all subscribers fails to register again. seems like the usrloc module does not work with the cluster. modparam("usrloc", "db_mode", 1) modparam("db_cluster", "cluster", "kamailio=>primary=9s9s;secondary=8s8s") it works OK for modules like dispatcher and htable to reload from DB. it shows the sql driver errors, but reloads from the secondary server. I need auto expire, so, p_usrloc is not a good option. working with version 4.2 any ideas ? Uri

2 1

FW: Mutual TLS with Skype for Business 2015
by Francisco Valentin Vinagrero 25 Oct '17

25 Oct '17

Hello, I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster. I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file) [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs [client:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved. The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate. As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this? Thanks, Francisco.

1 0

Advice on taking the best path to build a Presence & IM server
by José Seabra 24 Oct '17

24 Oct '17

Hello there, I'm writing here in order to get your experiences about a Presence & IM server. I would like to know what is the best solution to follow: - SIP Simple or XMPP? I've more knowledge about SIP Simple with kamailio but i did some researches and i found several open sources unified communications software that's implement Presence and IM using an XMPP server. So what is your opinion about which is the best solution to choose, taking in considerations the following aspects: 1. scalability 2. interopeability 3. permornace 4. features(like: Personal assistance, voicemail integration, chat group, integration with social media) Should i choose Kamailio SIP Simple with xcap or an XMPP server(Prosody or ejabberd, etc) I will appreciate a lot your opinion. Thank you Regards José -- Cumprimentos José Seabra

3 4

lcr
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

(no subject)
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

How to check 4XX/5XX response in general.
by vivek＠advaitamtech.com 24 Oct '17

24 Oct '17

Hi, Is there any way to catch all 4XX and 5XX SIP error codes rather then specifying individually ie. 404, 500, 400 etc in kamailio. Thanks, Vivek.

2 2

NatHelper ignoring force_socket module parameter
by Asgaroth 24 Oct '17

24 Oct '17

Hi All, I have a strange issue, I have set the module parameter for nathelper's force_socket to a specfic ip/port, however, when I perform a sip trace I can see that all locally generated options messages are not sent from the socket defined in the modules parameters. I am not setting $fs anywhere in the script, so I am not over-writing it. Do any of the other module parameters have a bearing on if nathelper uses the force_socket parameter? My current nathelper settings are: modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "natping_interval", 20) modparam("nathelper", "natping_processes", 4) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:keepalive@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", NAT_BFLAG) modparam("nathelper", "force_socket", "1.2.3.4:5060") I am using kamailio v4.3.1: # kamailio -V version: kamailio 4.3.1 (x86_64/linux) f38e67 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: f38e67 compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7 Thanks

2 5

No music on hold with mohqueue
by Fedor 24 Oct '17

24 Oct '17

I got in my kamailio.cfg: loadmodule "mohqueue.so" # --- music on hold moh ---- modparam ("mohqueue", "db_url", "mysql://kamailio:PASSWORDHERE@localhost/ kamailio") modparam ("mohqueue", "mohdir", "/etc/kamailio/moh") modparam ("mohqueue", "moh_maxcalls", 3000) Databases for mohqueue exist rttproxy configured fed@vps1:/etc/kamailio# ll /etc/kamailio/moh/ total 185424 drwxr-xr-x 2 kamailio kamailio 4096 Oct 8 09:32 ./ drwxr-xr-x 3 root root 4096 Oct 8 10:04 ../ -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.0 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.1 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.10 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.11 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.12 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.13 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.14 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.15 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.16 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.17 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.18 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.19 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.2 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.20 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.3 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.4 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.5 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.6 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.7 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.8 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.9 -rw-r--r-- 1 kamailio kamailio 8627660 Jan 2 2017 hold.wav rtpproxy configured, but no music on hold(srtp works fine) Can anybody help me, please?

2 1

Multi crashes in the same day
by igor.potjevlesch＠gmail.com 23 Oct '17

23 Oct '17

Hello ! I have multi crashes on a Kamailio 4.2.8 instance. Already 6 coredump today, same time with the same scenario. Below the output of the last coredump: (gdb) where #0 0x0000003d6f6328a5 in raise () from /lib64/libc.so.6 #1 0x0000003d6f634085 in abort () from /lib64/libc.so.6 #2 0x000000000061aad2 in qm_debug_frag (qm=0x7f9a8a7aa010, f=0x7f9a8aca5968) at mem/q_malloc.c:161 #3 0x000000000061bf1c in qm_malloc (qm=0x7f9a8a7aa010, size=24, file=0x7131d0 "<core>: re.c", func=0x71483e "subst_run", line=440) at mem/q_malloc.c:388 #4 0x0000000000505e2c in subst_run (se=0x7f9a8ac08b78, input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, count=0x0) at re.c:440 #5 0x0000000000506fff in subst_str (input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, se=0x7f9a8ac08b78, count=0x0) at re.c:515 #6 0x00007f9a87ff629d in subst_uri_f (msg=0x7f9a8abfef38, subst=0x7f9a8ac08b78 "X\215\300\212\232\177", ignored=0x0) at textops.c:744 #7 0x000000000041d51b in do_action (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1094 #8 0x0000000000429ba3 in run_actions (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1583 #9 0x000000000042a208 in run_actions_safe (h=0x7fffed081380, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1648 #10 0x0000000000542994 in rval_get_int (h=0x7fffed081380, msg=0x7f9a8abfef38, i=0x7fffed080890, rv=0x7f9a8ab64ef8, cache=0x0) at rvalue.c:924 #11 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed081380, msg=0x7f9a8abfef38, res=0x7fffed080890, rve=0x7f9a8ab64ef0) at rvalue.c:1918 #12 0x000000000041cf77 in do_action (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1064 #13 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1583 #14 0x000000000041d3f6 in do_action (h=0x7fffed081380, a=0x7f9a8ab6b070, msg=0x7f9a8abfef38) at action.c:1079 #15 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab607b8, msg=0x7f9a8abfef38) at action.c:1583 #16 0x000000000042a2d0 in run_top_route (a=0x7f9a8ab607b8, msg=0x7f9a8abfef38, c=0x7fffed081380) at action.c:1669 #17 0x00007f9a891f6d0f in prepare_new_uac (t=0x7f9a6de86518, i_req=0x7f9a8abfef38, branch=0, uri=0x7fffed0814d0, path=0x7fffed0814b0, next_hop=0x7f9a8abff1a8, fsocket=0x0, snd_flags=..., fproto=0, flags=2, instance=0x7fffed0814a0, ruid=0x7fffed081490, location_ua=0x7fffed081480) at t_fwd.c:410 #18 0x00007f9a891faded in add_uac (t=0x7f9a6de86518, request=0x7f9a8abfef38, uri=0x7f9a8abff1a8, next_hop=0x7f9a8abff1a8, path=0x7f9a8abff568, proxy=0x0, fsocket=0x0, snd_flags=..., proto=0, flags=2, instance=0x7f9a8abff578, ruid=0x7f9a8abff590, location_ua=0x7f9a8abff5a0) at t_fwd.c:855 #19 0x00007f9a89201ddd in t_forward_nonack (t=0x7f9a6de86518, p_msg=0x7f9a8abfef38, proxy=0x0, proto=0) at t_fwd.c:1723 #20 0x00007f9a891ee374 in t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, proto=0, replicate=0) at t_funcs.c:355 #21 0x00007f9a8922bd89 in _w_t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, force_proto=0) at tm.c:1517 #22 0x00007f9a8922ceee in w_t_relay (p_msg=0x7f9a8abfef38, _foo=0x0, _bar=0x0) at tm.c:1718 #23 0x000000000041d48d in do_action (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1088 #24 0x0000000000429ba3 in run_actions (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1583 #25 0x000000000042a208 in run_actions_safe (h=0x7fffed0848f0, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1648 #26 0x0000000000542994 in rval_get_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, i=0x7fffed082668, rv=0x7f9a8a83eb60, cache=0x0) at rvalue.c:924 #27 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082668, rve=0x7f9a8a83eb58) at rvalue.c:1918 #28 0x0000000000546fc2 in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082af0, rve=0x7f9a8a83f280) at rvalue.c:1926 #29 0x000000000041cf77 in do_action (h=0x7fffed0848f0, a=0x7f9a8a83fbd0, msg=0x7f9a8abfef38) at action.c:1064 #30 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a838c88, msg=0x7f9a8abfef38) at action.c:1583 #31 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996598, msg=0x7f9a8abfef38) at action.c:712 #32 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a9955b8, msg=0x7f9a8abfef38) at action.c:1583 #33 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996a50, msg=0x7f9a8abfef38) at action.c:1079 #34 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a8fa068, msg=0x7f9a8abfef38) at action.c:1583 #35 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:712 #36 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:1583 #37 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830db8, msg=0x7f9a8abfef38) at action.c:1079 #38 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38) at action.c:1583 #39 0x000000000042a2d0 in run_top_route (a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38, c=0x0) at action.c:1669 #40 0x00000000005095b6 in receive_msg ( buf=0xa78b60 "INVITE sip:0123456789@A.B.C.181:5060;transport=UDP;user=phone SIP/2.0\r\nFrom: <sip:0987654321@A.B.C.148:5060>;tag=BN1507883839-1-1507047459-332794206;tran sport=UDP\r\nTo: <sip:0123456789@H.G.F.D"..., len=1119, rcv_info=0x7fffed084be0) at receive.c:216 #41 0x000000000060945a in udp_rcv_loop () at udp_server.c:521 #42 0x00000000004a6e62 in main_loop () at main.c:1629 #43 0x00000000004acfa6 in main (argc=7, argv=0x7fffed085018) at main.c:2581 Regards, Igor.

2 2

Presence WARNING after publish and no NOTIFY
by Yuriy Gorlichenko 23 Oct '17

23 Oct '17

Hi use 5.0.1 with presence for TCP/UDP endpoints presence works PERFECT but for WS socket presence i see next (small cleaned log here with all described bellow https://pastebin.com/8a77ggj0 ): When WS endpoint subscribes - kamailio After kamailio sends 202 to SUBSCRIBE message (see it on client) it sends fast NOTIFY correcly without any trouble, and i see at the kamailio reply from WS endpoint - 200 WS endpoint added to active_watchers table. But when Subscribed endpoint makes for example some call and presence server sends PUBLISH i see: presence [presentity.c:426]: delete_presentity_if_dialog_id_exists(): Presentity already exists - deleting it At the logs and then presence server doesn't send NOTIFY to WS endpoint What is the issue of this? Can not understand why it happens. Can someone to help me understand this? And the dump i see that kamailio send first NOTIFY form UDP port but then translates it to tls for websocket and i see correct NOTIFY at the WS endpoint

2 4

MoS(QoS) measurement in Kamailio based IMS network
by Kranti Kumar 23 Oct '17

23 Oct '17

Hi All, I am using Kamailio IMS VMimage which is provided by Franz. I am successfully established the call between the IMS clients. Here I would like to measure the MoS mean opnion score (QoS) of Calls (Voice quality). Can any one suggest me / guide me how to measure the same in this IMS network. Thank you so much in advance. Regards, -kranti

2 1

how to bridge two landline users
by 赵国杰 23 Oct '17

23 Oct '17

Hello, In a standard sip flow, the call goes like: sip user A --> kamailio --> pstn --> landline user B. However, when user A has a bad internet access, the audio is broken. So what I want is to let sip user A send a invite to kamailio first, then kamailio send invite to user A and B's landline number through pstn, then bridge the two call together. I understand this can be achieved by using FREESWITCH originate and bridge command. I've tried but there's no audio both ways, which really makes me feel stupid of myself. So I'm wondering if this can be done with kamailio? If so, how? Thanks Jesse

2 1

uac_replace_from and header adulterations
by Alex Balashov 23 Oct '17

23 Oct '17

Hello, I've got a bizarre problem caused by bad UA behaviour: UA A ---> Kamailio (P) ---> UA B 1. UA A sends initial INVITE through P to B; 2. Kamailio (P) makes some modifications to the From header using uac_replace_from() and passes along to B. 3. B sends an in-dialog request (e.g. BYE or reinvite) to A through P; in doing so, it modifies the To (formerly From) value slightly, replacing the hostname portion in the To URI with a different value to the one that was received in the From header. 4. Kamailio relays this in-dialog request to A, but with an adulterated/clipped/truncated/grammatically invalid To header now. 5. A responds with 400 Bad Request due to invalid To header. I would not dispute that UA B should not be modifying the remote URI in this manner. But since it does, it gives rise to two questions: 1. Why does Kamailio not simply discard the modified To header and restore the original value stored in the Record-Route rider parameter? Is it because the Record-Route parameter does not contain the original header value, but rather some data complementary to the current header value? Or is it because the UAC code takes a checksum of the original remote URI header value and stores it, and just checks it when restoring on principle? If so, what's the motive for that? 2. Wouldn't it be better behaviour to simply reject a request so malformed, rather than passing it on with a corrupt restored value? If Kamailio can detect that the header has been tampered with, why pass it on? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Help on Video Conference
by Kranti Kumar 23 Oct '17

23 Oct '17

Hello , I’m trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ My configuration details: HSS - 10.0.0.9 (./hss.sh) PCSCF - 10.0.0.10 (kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg) ICSCF - 10.0.0.11(kamailio -f /usr/local/etc/kamailio/kamailio-icscf.cfg) SCSCF - 10.0.0.12 (kamailio -f /usr/local/etc/kamailio/kamailio-scscf.cfg) host machine: 10.0.0.5 I am able to register mobile clients ( IMSdroid) and make voice/video calls between them successfully. Now I would like to make a conference call among 3 IMSdroid users but its failing. Could you please suggest me the way forward here. I am thinking to integrate Application server (AS) to enable Video Conferences. Could you please guide me how to enable AS in this VM image and configuration details. Is there any possibility to do Video conference using this current setup(VM image). Could you please suggest me. Thank you so much in advance. Regards, -kranti

2 1

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