- sr-users - kamailio.org

dialog's track_cseq_updates seems to break uac_auth() in this configuration
by George Diamantopoulos 24 Jun '20

24 Jun '20

Hello again, This might turn out to be a bug, but I'm posting here first because my config is fairly complicated and it might not be the sanest it could be. The problem manifests as follows: with the sole change being the enabling of track_cseq_updates for the dialog module, kamailio fails to process the 100 Trying it receives following authentication initiated by the uac_auth() function. Without the track_cseq_updates option enabled, the call proceeds successfully, albeit with the CSeq for the auth-carrying INVITE having the same value as the original. Here's a simplified version of the configuration routes involved in this: request_route { ... route(DISPATCH_PROVIDER); ... } route[DISPATCH_PROVIDER] { if ( !ds_select_dst("10", "8") ) { t_send_reply("503", "Downstream carrier unavailable"); exit; } t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_branch("PROVIDER_BRANCH"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } failure_route[DISPATCH_PROVIDER_FAILURE] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(arealm) = "authrealm.com"; $avp(auser) = "authusername"; $avp(apass) = "verys3kr1t"; uac_auth(); t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); $du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp); // I'll explain why this is here later in this e-mail t_relay(); exit; } } branch_route[PROVIDER_BRANCH] { uac_replace_from("$dlg_var(my_new_from)"); uac_replace_to("$dlg_var(my_new_to)"); $rU = "my new RURI user"; } event_route[tm:branch-failure:DISPATCH_PROVIDER_FAILOVER] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { return; # next DST - only for 5xx or local timeout } else if ( t_check_status("5[0-9][0-9]") || (t_branch_timeout() && !t_branch_replied()) ) { if ( ds_next_dst() ) { t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } else { xlog("L_NOTICE", "--- SCRIPT_DISPATCH_PROVIDER_FAILOVER: Failed to route request to PROVIDER! Giving Up. Negative response will be sent upstream.\n"); } } } One will rightfully wonder why use both branch-failure routes and failure_route to handle things. Well, it's not obvious from this because the relevant parts are removed for brevity, but I'm generally using branch failure event routes to try other destinations in the same destination set, and failure routes to additionally try other downstream "providers" in case all entries is the current destination set have failed. I don't think these actions/routes are relevant because the problem manifests without any of these failover mechanisms (of switching over to the next provider) engaging. Regarding the line "$du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp);" in the failure_route, which might seem a little odd, this was added because without it, uac_auth() will send the auth-carrying INVITE always to the first destination in the destination set even if ds_next_dst has been called from the branch-failure event route. Unfortunately I haven't been able to determine if this actually helped, because I have only received a 5xx error once from the downstream peer before the "fix" and haven't been able to reproduce this since then. But that's another issue unrelated to the one I'm asking about here. So what happens with track_cseq_updates enabled is, after uac_auth() successfully sends the authenticated INVITE and the peer starts sending provisional responses, kamailio doesn't seem to acknowledge them. Instead, it will retransmit the auth INVITE as if there was a firewall preventing the 1xx responses from being admitted. Then, the timeout will engage and the configuration script will behave as if t_branch_timeout returns true (it will do ds_next_dst). So I was wondering if this is to be expected with this configuration, or if this should be reported as a bug. Thanks! Best regards, George

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Kamailio-5.3.3. NAT and RTP not working
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi , Could anybody please give me the approach how to configure NAT in kamailio IMS. I have installed kamailio IMS in openstack environment which has private ips for all four vms (each component in one vm). pcscf has one floating IP attached. I have enabled "WITH_NAT" and configured rtpproxy . when i make a call from outside of openstack using 2 zoiper clients with outbound proxy keeping floating ip of pcscf, I am getting 200 ok with all SIP packets correctly flowing but there is no RTP Packet. When i try the same within openstack environment without using public ip, I am getting RTP Packets. Kindly tell me the procedure of NAT Enablement or do i need to do anything other than this. Kindly help because i am stuck with this for long time. Thanks.

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multiple domains with microsoft team
by Leanderson Correia 24 Jun '20

24 Jun '20

I'm testing here to use multiple domains with microsoft teams and I'm having trouble making and receiving calls. It seems that when connecting a call, kamailio keeps the tls connection active and reuses it, making the domain parameters used the wrong ones for new call. Can someone help me, or give a tip? -- Att, Lean

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Presence and call queueing - Kamailio and Asterisk
by Pover A.V. 24 Jun '20

24 Jun '20

Summary: Using Kamailio to load balance between Asterisk servers. Trouble getting call queuing working because can't get device state hints to work reliably. - Hello I'd be grateful for any advice that can be offered for this problem to do with presence and call queueing. I have 3 Kamailio (v5.3.4) servers that load-balance incoming traffic from various sources: a legacy PBX, soft phone clients, and a SIP trunking service. Sitting behind the 3 Kamailio servers are 2 Asterisk (v16 LTS) servers. Kamailio acts as a registrar and user location service. Asterisk is routing calls and (should be) providing device state information for each extension. An external script builds a list of PJSIP endpoints based on the content of the Kamailio subscriber table, and deploys it to the Asterisk servers whenever a change is detected. The AOR for each record is configured to point to Kamailio. No authentication details are attached to these endpoints, as Kamailio is doing authentication before dispatching a request to Asterisk. Whenever Kamailio receives an INVITE, it passes this to an Asterisk server using the dispatcher module (round-robin algorithm '4'); asterisk then processes the request. At the moment, everything is working except for presence/device states and call queueing. I've tried to set up 'presence', 'pua', 'dialog' and '(presence/pua)_dialoginfo' modules on Kamailio with limited success. The presence state when a user toggles 'do not disturb' on their soft phone is passed to other users who are subscribed (watching), however when a user makes a call, no presence updates are seen by subscribed users (the extensions show as 'Idle' or 'Available' instead of 'In Use' or 'Busy'). Asterisk is able to report that an extension is 'Ringing' or 'InUse' when a call is made to that extension, because an INVITE to an internal extension matches a pattern in the dialplan and Asterisk routes it to a PJSIP endpoint with the same name. However, when somebody makes a call, Asterisk is unable to track the state of the extension making the call because the call is going via Kamailio so Asterisk is not seeing the caller as an internal extension. We need a way to make Asterisk think the call is coming from an internal device if the caller ID matches a PJSIP endpoint. Both Asterisk servers are configured to share device state updates with each other, and this is working. Without being able to see device state for an internal calling party, we will be unable to reliably set up call queues in Asterisk. Using the 'mohqueue' module on Kamailio is not something we want to do because of its reliance on a database, which is creating a single point of failure - if the database server is down for any reason, Kamailio refuses to start. Ideally, we would achieve some form of distributed call queueing, however the simplest option seems to be setting up a queue on one Asterisk server and having an active-passive high availability configuration, should a server go down. I'm looking for advice on how best to keep track of what is happening in a call, and for Asterisk to know the states of each extension and make these available in a dialplan hint. Or any other suggestions that might help to achieve call queueing in this setup. Thank you

2 1

Dialog Profiles
by Ilie Soltanici 24 Jun '20

24 Jun '20

Hello, Is there any way in Kamailio to get the profile values the current dialog is part of? For example, I have one dialog with the following profiles: profiles: { { test: abcd } test: 1234 } I need in the reply route to get the values for the test dialog profile. (abcd, 1234). I know I can use get_profile_size, but that will give the number of profiles of the current dialog, but I need the values to loop through. Thanks.

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Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin

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Kamailio ims video calls
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi, Could anyone please give me some idea on how video call works in IMS. Does it need rtp proxy to work? Or is it the same way how voice call works ? It would be good , if anybody share some links on video calls as well voice conferencing. Thanks, pavithra

2 1

kamailio 5.2.2 enabling hashing_type
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello, I'm revisiting some initial issues I had with P-CSCF. I could not set the hashing type in my PCSCF (kamalio 5.2.2) so I simply commented it out and moved on. But now, I may need it as the P-CSCF is not handling INVITES coming from I-CSCF that originated from external SIP server, where the endpoint (UE) is registered to the IMS but the P-CSCF sends the INVITE back to the S rather that the registered UE and fails. kamailio.cfg: modparam("ims_usrloc_pcscf", "hashing_type", 1) So with that, when I set the hashing_type to 1,1 or 2 I get the following starting the pcscf: Starting the P-CSCF: Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Not starting Kamailio IMS P-CSCF: invalid configuration file! Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * 0(5766) ERROR: <core> [core/modparam.c:140]: set_mod_param_regex(): parameter <hashing_type> of type <2> not found in module <ims_usrloc_pcscf> Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: 0(5766) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 364, column 47: Can't set module parameter Am I missing some module that I did not include at compile time? According to the docs, it has no additional dependencies other than the ims_usrloc_pcscf.so https://kamailio.org/docs/modules/5.0.x/modules/ims_usrloc_pcscf.html#idp35… Any help is appreciated. Thanks, _Martin

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kamailio5.2.2 calling t_relay(host, port)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, Im trying to call t_relay() with host and port as documented in: https://www.kamailio.org/docs/modules/5.2.x/modules/tm.html However, I get a subst_str(): no match on P-CSCF start and therefore fails. Is t_relay(host, port) supported as documented? Am I missing something? Thanks, _Martin

2 1

Expired transaction
by Marat Gareev 24 Jun '20

24 Jun '20

Hello! I'm trying to manually reproduce expired transactions which couldn't be cleared. I had one idea, but it was unsuccessful. I tried to send SIGSTOP to a group of Kamailio processes while the transaction is live. The idea is to freeze processes for a while, then continue to work, so that by this time the transaction deletion time was missed. But after a few seconds, the transaction is still cleared. Do you have any ideas for reproduction? P.S. I'm sorry If you received my incoherent mails. These were attempts to reply to the thread, seems to be my mail client or sr-users mailing list doesn't support In-Reply-To header.

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ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found
by Denys Pozniak 23 Jun '20

23 Jun '20

Hello! Could someone point me to the possible source of the problem? Jun 16 09:51:06 proxy-edge /usr/sbin/kamailio[9645]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9640]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9600]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9660]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Modules settings: *loadmodule "pua.so"* *loadmodule "pua_dialoginfo"* *loadmodule "dialog.so" * *modparam("pua", "default_expires", 1200)* *modparam("pua", "outbound_proxy", LOCAL_IP_ADDR)* *modparam("pua_dialoginfo", "include_callid", 1)* *modparam("pua_dialoginfo", "include_localremote", 1)* *modparam("pua_dialoginfo", "include_tags", 1)* *modparam("pua_dialoginfo", "send_publish_flag", 8)* *modparam("dialog", "dlg_flag", 4)modparam("dialog", "db_mode", 0)modparam("dialog", "enable_stats", 1)modparam("dialog", "dlg_match_mode", 1)modparam("dialog", "default_timeout", 7200) modparam("dialog", "hash_size", 8192)* *request_route{* *...* *setflag(8);* *...* *}* Generated SIP PUBLISH: *PUBLISH sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com> SIP/2.0* *Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bK3e55.afdce977000000000000000000000000.0* *To: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>* *From: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>;tag=e98d9fc94bada9746d67946ab287a76f-ba77* *CSeq: 10 PUBLISH* *Call-ID: 72ef85f374c8e703-9659(a)10.10.10.10 <72ef85f374c8e703-9659(a)10.10.10.10>* *Content-Length: 578* *User-Agent: proxy* *Max-Forwards: 70* *Event: dialog* *Expires: 7201* *Content-Type: application/dialog-info+xml* *<?xml version="1.0"?>* *<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>">* * <dialog id="162092997-5060-464(a)BJC.BGI.CA.BBI" call-id="162092997-5060-464(a)BJC.BGI.CA.BBI" direction="initiator">* * <state>Trying</state>* * <remote>* * <identity>sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com></identity>* * <target uri="sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com>"/>* * </remote>* * <local>* * <identity>sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com></identity>* * <target uri="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>"/>* * </local>* * </dialog>* *</dialog-info>* -- BR, Denys Pozniak

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icscf start error
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I have installed kamailio - 5.3.4 version where i am facing below issues in icscf while starting kamailio. db_mysql is loaded but .. what could be the issue? icscf : Aliases: *: ims.sip.example.com:* 0(789) INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(791) INFO: pike [pike.c:97]: pike_init(): PIKE - initializing 0(791) INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size *0(791) ERROR: <core> [db.c:204]: db_bind_mod(): Module db_con1=>mysql not found. Missing loadmodule? 0(791) ERROR: ims_icscf [db.c:83]: ims_icscf_db_bind(): ims_icscf_db_bind: cannot bind to database module! Did you forget to load a database module ? 0(791) ERROR: ims_icscf [ims_icscf_mod.c:270]: mod_init(): Failed to initialise ims_icscf module 0(791) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_icscf (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_icscf.so)* ERROR: error while initializing modules 0(791) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... 0(791) INFO: cdp [diameter_peer.c:367]: diameter_peer_destroy(): destroy_diameter_peer(): Terminating all children... 0(791) INFO: cdp [diameter_peer.c:386]: diameter_peer_destroy(): destroy_diameter_peer(): All processes terminated. Cleaning up. 0(791) INFO: cdp [worker.c:139]: worker_destroy(): Unlocking workers waiting on empty queue... 0(791) INFO: cdp [worker.c:142]: worker_destroy(): Unlocking workers waiting on full queue... 0(791) CRITICAL: cdp [diameter_peer.c:426]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test 0(791) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped 0(791) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized

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Kamailio v5.3.5 Released
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, Kamailio SIP Server v5.3.5 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.5. For more details about version 5.3.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/06/kamailio-v5-3-5-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

1 0

reason=404 (Not Found - destination user not found on this S-CSCF)
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I am using kamailio ims-5.3.4 where i am facing this kind of problem during route(term). call registration is happening successfully. In call Establishment , during route(term) , I am facing destination user not found . what could be the problem?. can anyone please help me. Thanks, Pavithra

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Releasing Kamailio v5.3.5
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, I am considering to release next week (likely on Monday, June 22) the next minor version out of branch 5.3, respectively 5.3.5. Should anyone be aware of issues not reported to the bug tracker or commits for fixes in master not backported to 5.3 branch, let us know. For new issues, the best is to report them on the tracker. For commits, you can provide links/hashids or make pull requests. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Re: [SR-Users] 403 domain not served error in scscf
by David Villasmil 20 Jun '20

20 Jun '20

you shouldn't write to me directly. You can write to the list and address me. Anyway, kamailio thinks the message received is NOT directed to it. You should add an "alias" with the IP you're sending to (or the FQDN) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 On Sat, Jun 20, 2020 at 9:01 PM Pavithra Mohanraja <pavimohan3004(a)gmail.com> wrote: > Hi David, > Could you please help me in this issue > > ---------- Forwarded message --------- > From: Pavithra Mohanraja <pavimohan3004(a)gmail.com> > Date: Fri, Jun 19, 2020, 1:36 PM > Subject: 403 domain not served error in scscf > To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > > > Hi, > I am using kamailio IMS - 5.3.4 > when I am making a call between two clients using zoiper , Call > establishment is done successfully . > > But during BYE packet sent from caller to callee , it is reaching till > S-CSCF and giving 403 Domain not served .. Kindly help me in this regard. > > > I have sent lot of mails regarding this error and no replies .. Kindly > help .. > > Error in scscf : > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7fa1e6945ca8 alloc'ed > from core: core/rvalue.c: rval_new_empty(224) > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/socket_info.c:628]: grep_sock_info(): checking if host==us: 10==11 && > [*10.45.5.22] == [10.45.4.181]* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/socket_info.c:635]: grep_sock_info(): *checking if port 4080 > (advertise 0) matches port 40629* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/forward.c:412]: *check_self(): host != me* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: > sip:alice at sip.example.com > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>:* > Domain not served 403* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: BYE ( > sip:bob@sip.example.com (10.45.4.177:4060) to sip:alice@sip.example.com;transport=TCP, > y-B071ZDjFS-z4NLxREyVw..) > > > Kindly help .. > 10.45.5.22 - sip client alice(a)sip.example.com > 10.45.4.181 - scscf > > > please help.. >

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How do I test the Kamailio service ？
by zhaosl＠shinetechchina.com 20 Jun '20

20 Jun '20

Hi : Now I set up a Kamailio Service on the server, it works now. But I didn`t know how to test the server by trying the voice tool.Can you give me some advices , tools or methods ? Thanks. zhaosl(a)shinetechchina.com

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Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

Hi can you share your kamaili config. I want to add some extra columns to cdr table like what you did. Thanks. Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

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Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

1 0

403 domain not served error in scscf
by Pavithra Mohanraja 19 Jun '20

19 Jun '20

Hi, I am using kamailio IMS - 5.3.4 when I am making a call between two clients using zoiper , Call establishment is done successfully . But during BYE packet sent from caller to callee , it is reaching till S-CSCF and giving 403 Domain not served .. Kindly help me in this regard. I have sent lot of mails regarding this error and no replies .. Kindly help .. Error in scscf : Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7fa1e6945ca8 alloc'ed from core: core/rvalue.c: rval_new_empty(224) Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/socket_info.c:628]: grep_sock_info(): checking if host==us: 10==11 && [*10.45.5.22] == [10.45.4.181]* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/socket_info.c:635]: grep_sock_info(): *checking if port 4080 (advertise 0) matches port 40629* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/forward.c:412]: *check_self(): host != me* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: sip:alice at sip.example.com Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>:* Domain not served 403* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: BYE ( sip:bob@sip.example.com (10.45.4.177:4060) to sip:alice@sip.example.com;transport=TCP, y-B071ZDjFS-z4NLxREyVw..) Kindly help .. 10.45.5.22 - sip client alice(a)sip.example.com 10.45.4.181 - scscf please help..

1 0

Development for v5.4.x is frozen
by Daniel-Constantin Mierla 19 Jun '20

19 Jun '20

Hello, the formal notification that the development for the next major version 5.4.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.4 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, pull requests can still be done, but they will be merged after branching 5.4. Can still be done commits with documentation improvements, enhancements to related tool (e.g., kamctl, kamcmd), exporting missing KEMI functions and completing the functionality of the new modules added in 5.4. Once the branch 5.4 is created, new features can be pushed again to master branch as usual. From that moment, the v5.3.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.4 is going to be built at: * https://www.kamailio.org/wiki/features/new-in-5.4.x Upgrade guidelines will be collected at: * https://www.kamailio.org/wiki/install/upgrade/5.3.x-to-5.4.0 Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.3.x to 5.4.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

MD5 and SHA-256 instead of MD5 or SHA-256...
by Aymeric Moizard 18 Jun '20

18 Jun '20

Hi All, I'd like to improve my setup by switching to SHA-256. However, as a first step, I would like to offer both MD5 and SHA-256 in 2 different WWW-Authenticate header. If I'm correct, this is not doable with the latest auth module? Is this a planned feature? As an alternative, I would like to decide the algorithm in the script instead of a module parameter. It looks to me this is also not doable? Again, is this a planned feature? Thanks to all, Regards Aymeric -- Antisip - http://www.antisip.com

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Freezing development for Kamailio v5.4.0
by Daniel-Constantin Mierla 18 Jun '20

18 Jun '20

Hello, based on the discussions during the last online devel meeting, the plan is to release the next major version (Kamailio v5.4.0) around the mid of this summer. Therefore I propose to freeze the development on Thursday, June 18, 2020. Then we will have about 4-6 weeks of testing before the release is out. Master will be unfrozen when the branch 5.4 is created (usually that was done after 3-4 weeks in the testing phase). If you are planning to introduce new features in the 5.4 release series, be sure you push them to master branch or create a pull request till the freezing date. If anyone else wants a different time line for 5.4.0 release, let us know and we can discuss to match the best for everyone. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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removing xavp from the stack.
by David Villasmil 18 Jun '20

18 Jun '20

Hello guys, if i do something like: $(xavp(my_var)) = "1"; $(xavp(my_var)) = "2"; $(xavp(my_var)) = "3"; it is store like: $(xavp(my_var)) = "1"; -> index 2 $(xavp(my_var)) = "2"; -> index 1 $(xavp(my_var)) = "3"; -> index 0 and later i need to use and remove from the stack, like this: use and remove: $(xavp(my_var)) --> would remove index 0 Then the stack would remain like: $(xavp(my_var)) = "1"; -> index 1 $(xavp(my_var)) = "2"; -> index 0 Simply popping the index 0 and reindex the remaining... is this possible? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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dispatcher seems to use the default client certificate
by Mack Hendricks 18 Jun '20

18 Jun '20

Hey All, I'm attempting to use dispatcher to send probe messages using TLS for two different domains. I'm providing the socket attribute, which maps to a certificate in /etc/kamailio/tls.cfg. But, it seems to always select the default client cert, which is not the certificate I want to use. My attrs column in dispatcher looks like this: socket=tls:142.93.159.231:5061;ping_from=sip:mack.dopensource.com socket=tls:142.93.159.231:5062;ping_from=sip:levin.dopensource.com Is there some way to force dispatcher to do TLS cert matching based on the host:ip? Thanks -Mack

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