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37396 discussions

Branching 5.4 in git repository
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, so far it looks pretty calm during the testing phase, some static analysis didn't reveal any new major issues (well, this is done also during development phase). Therefore, if nothing else is proposed or shows up, I am considering to create the git branch 5.4 next week on Thursday -- this branch will hold the code for 5.4.x release series. After branching 5.4, the master branch will be open again for new features. The 5.4.0 enters the release candidate phase, likely to have the first release out in 1-2 weeks after branching. 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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Maintenance work on kamailio.org server - Wed, July 8, 2020
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, on Wednesday, July 8, 2020, there will be some work on Kamailio project infrastructure. Therefore there could be short intervals of downtime for online resources like the web server, wiki, documentation and downloads portals, mailing lists or the GIT repository mirror. Once the maintenance work is finished, we will post an update. 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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30X redirect in interconnects - better alternative?
by Gerry | Rigatta.com 08 Jul '20

08 Jul '20

Hi, I would like to use Kamailio for load balancing incoming carrier traffic. We do currently IP authentication and call logic in Yate boxes. Ideally I would like to distribute calls with 30X redirects with the Kamailio dispatcher so that IP authentication and all logic can stay in the Yate boxes. However I have doubts that 30X redirects are generally accepted in interconnects. What is your experience with this? What is the possible alternative to redirects if one wants to keep IP authentication and call logic in the boxes behind the Kamailio SIP router? E.g. how can one reliably check the carrier source IPs behind Kamailio? Custom headers injected by Kamailio? Of cause I can check source IPs with a database lookup in Kamailio but I try to avoid that as this makes the setup much more complicated and error prone. Thank you for your ideas. Gerry

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Re: [SR-Users] sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Update: Disabling the topoh module on the proxy which produces the error seems to stop the failure from manifesting. I'll try using topos_redis instead, but should this be treated as a bug? BR, George On Wed, 8 Jul 2020 at 12:37, George Diamantopoulos <georgediam(a)gmail.com> wrote: > Hello again, > > Indeed $mb seems to contain garbage: > > SCRIPT_MB: ACK <BC><EA><8F> SIP/2.0#015#012Via: SIP/2.0/UDP > 172.30.154.189;TH=dcv;branch=z9hG4bK629b.6af9302cd78dc58dffe817e60124f4ed.0#015#012Route: > <sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@172.30.155.1 > ;lr;received=sip:2.2.2.2:32768 > ;ob;r2=on>,<sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@3.3.3.3 > ;lr;received=sip:2.2.2.2:32768;ob;r2=on>#015#012Max-Forwards: > 68#015#012From: "Anonymous" <sip:unknown@voip.domain.com>;tag=as4bc9e324#015#012To: > <sip:voip-test-user-02@voip.domain.com>;tag=jw7z5s0zvc#015#012Call-ID: > 0138c6370346d1dd7f1b47f604b01f92(a)voip.domain.com#015#012CSeq > <http://0138c6370346d1dd7f1b47f604b01f92@voip.domain.com#015%23012CSeq>: > 102 ACK#015#012Content-Length: 0#015#012TH: dch#015#012#015#012 > > How can this be possible? Capturing traffic on wire shows the RURI I > pasted in my original message and there are no script operations on the > RURI before sanity_check() (message buffer above is printed just before > sanity_check() is run in REQINIT). > > BR, > George > > On Wed, 8 Jul 2020 at 11:18, George Diamantopoulos <georgediam(a)gmail.com> > wrote: > >> I'll post the message buffer ASAP, but in the meantime I don't see how >> config operations could affect the RURI. Here's everything that's happening >> until the sanity check involved: >> >> request_route { >> >> if(is_method("KDMQ")) { >> dmq_handle_message(); >> } >> >> # no connect for sending replies >> set_reply_no_connect(); >> >> if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { >> # silent drop for scanners - uncomment next line if want to reply >> # sl_send_reply("200", "OK"); >> exit; >> } >> >> if (!mf_process_maxfwd_header("10")) { >> force_rport(); >> sl_send_reply("483","Too Many Hops"); >> exit; >> } >> >> # OPTIONS and NOTIFYs directed to myself >> if(is_method("OPTIONS|NOTIFY") && uri==myself && $rU==$null) { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> # All keep-alive methods regardless of destination >> if ( $hdr(Event) == "keep-alive") { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> if(!sanity_check("17895", "7")) { >> xlog("Malformed SIP request from $si:$sp\n"); >> exit; >> } >> >> BR, >> George >> >> On Wed, 8 Jul 2020 at 10:58, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> check your config operations, because the R-URI seems to be the next >>> string (without quotes): "<BC><EA><8F>" >>> >>> You can try to print $mb in such case to see the entire SIP message >>> buffer. >>> >>> Cheers, >>> Daniel >>> On 08.07.20 09:48, George Diamantopoulos wrote: >>> >>> Hello Daniel, >>> >>> Thanks for the reply. Indeed there is, not sure how I managed to miss >>> that. And it wasn't about the schema after all: >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1254]: parse_uri(): uri too short: <<BC><EA><8F>> >>> (3) >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1328]: parse_sip_msg_uri(): bad uri <<BC><EA><8F>> >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: WARNING: >>> {1 <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:282]: check_ruri_scheme(): failed to parse request uri >>> [<BC><EA><8F>] >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: ERROR: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <script>: Malformed >>> SIP request from 172.30.154.189:5060 >>> >>> Still, not sure what the problem is though... >>> >>> BR, >>> George >>> >>> On Wed, 8 Jul 2020 at 09:30, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> when the ruri scheme check fails, there should be another debug message >>>> saying that. Have you pasted all log messages for the failure case? >>>> >>>> Cheers, >>>> Daniel >>>> On 07.07.20 22:23, George Diamantopoulos wrote: >>>> >>>> Sorry, I realised I copy pasted wrong log messages for Call 1. Here's >>>> the relevant part showing success for call 1 in contrast with Call 2: >>>> >>>> grep 2a859fcc4e1c8f840191a81d7c16e76d kamailio.log | egrep >>>> 'check_ruri_scheme|w_sanity_check' | grep ACK >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 1 >>>> >>>> On Tue, 7 Jul 2020 at 21:34, George Diamantopoulos < >>>> georgediam(a)gmail.com> wrote: >>>> >>>>> Hello all, >>>>> >>>>> I'm not 100% sure this is the only culprit in an issue I'm >>>>> investigating, but superficially it appears that RURI scheme sanity module >>>>> checks from the default config (flags 17895 in REQINIT) fail if the >>>>> RURI in an ACK following a 487 includes parameters. Example from two calls >>>>> from a kamailio instance acting as registrar/usrloc server, INVITE RURIs >>>>> are after usrloc lookup: >>>>> >>>>> Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> These INVITEs produce no complaints. Later, the same registrar >>>>> produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses >>>>> from the next proxy in the path following a CANCEL: >>>>> >>>>> Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> The next proxy (which produced/relayed the 487) processes the ACK for >>>>> Call 1 successfully, but sanity_check at the proxy drops the request for >>>>> Call 2 with: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>>>> >>>>> whereas Call 1 seems OK: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:305]: check_required_headers(): check_required_headers entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:313]: check_required_headers(): check_required_headers passed >>>>> >>>>> Could this be a bug in sanity module? Is there anything one can do in >>>>> config which could result in illegal ACKs being produced for hop-by-hop >>>>> transactions? schema appears to be sip: in both cases... >>>>> >>>>> Thank you. Best regards, >>>>> George >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Funding: https://www.paypal.me/dcmierla >>>> >>>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Funding: https://www.paypal.me/dcmierla >>> >>>

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Setting up Kamailio as Teams SBC
by Роман С. 08 Jul '20

08 Jul '20

Hello. I'm trying to set up Kamailio as SBC for Teams using https://skalatan.de/en/blog/kamailio-sbc-teams. Setup is completely default (except things mentioned at article), but I use wildcard certificate for TLS. Well, I can't even pass dispatcher: kamcmd dispatcher.list | egrep "RI|FLAG" URI: sip:sip.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 3 URI: sip:sip2.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 2 URI: sip:sip3.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 1 Where do I start to dig?

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sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Hello all, I'm not 100% sure this is the only culprit in an issue I'm investigating, but superficially it appears that RURI scheme sanity module checks from the default config (flags 17895 in REQINIT) fail if the RURI in an ACK following a 487 includes parameters. Example from two calls from a kamailio instance acting as registrar/usrloc server, INVITE RURIs are after usrloc lookup: Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 These INVITEs produce no complaints. Later, the same registrar produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses from the next proxy in the path following a CANCEL: Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 The next proxy (which produced/relayed the 487) processes the ACK for Call 1 successfully, but sanity_check at the proxy drops the request for Call 2 with: DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 whereas Call 1 seems OK: DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:305]: check_required_headers(): check_required_headers entered DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:313]: check_required_headers(): check_required_headers passed Could this be a bug in sanity module? Is there anything one can do in config which could result in illegal ACKs being produced for hop-by-hop transactions? schema appears to be sip: in both cases... Thank you. Best regards, George

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Some Functions not defined in Documentation of Module ims_registrar_pcscf
by Hamid Hashmi 08 Jul '20

08 Jul '20

I have been using a sample code of PCSCF<https://github.com/kamailio/kamailio/blob/master/misc/examples/ims/pcscf/ro…>, where a function "pcscf_assert_identity" is used but it's not defined in module ims_registrar_pcscf documentation. Can you please explain its functionality. The same way other functions are not defined in the documentation. pcscf_assert_called_identity reginfo_handle_notify pcscf_unregister Regards Hamid R. Hashmi

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Kamailio 5.3.4 - Segfault on http_async_client
by José Seabra 07 Jul '20

07 Jul '20

Hello there, Kamailio is terminating with signal 11 and is generating a coredump file. I opened the core file generated by it and it seems to be related to the module http_async_client. In the following link you can find the output of core dump backtrace full: https://pastebin.com/rapPZy9N Below is the http_async_client config that I have: modparam("http_async_client", "workers", 2) modparam("http_async_client", "connection_timeout", 500) modparam("http_async_client", "hash_size", 2048) modparam("http_async_client", "curl_verbose", 0) modparam("http_async_client", "memory_manager", "shm") If the information provided is not enough to troubleshoot this issue let me know. Many thanks Best Regards -- Cumprimentos José Seabra

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Kamailio Buildroot
by Benjamin Marty 07 Jul '20

07 Jul '20

Hello I would like to ask if someone already tried to build Kamailio for the Embedded Distribution Buildroot. I tried it myself but it looks like Kamailio is not that easy to cross compile. Thanks Benjamin

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siptrace not sending data to hepic
by Andrew Chen 07 Jul '20

07 Jul '20

Hi all, I've been assigned a task to investigate why hepic doesn't generate MOS scores ever since we changed the kamailio interface MTU from 9001 -> 1500. During my investigation, this is what I have found out so far: - Putting MTU back to 9001 removes this condition where no MOS was generated. - I did a tcpdump of port 9063 which is the port used to send data to Hepic capture server and saw it reports packets are fragmented: 3:48:55.895505 IP (tos 0x68, ttl 64, id 38049, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP 13:48:55.896451 IP (tos 0x68, ttl 64, id 38050, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP - Hepic support also caused by using 1500 MTU on the interface. - I tried using the following core parameters to see if they could remedy this issue but didn't seem to work: ## MTU related core settings udp4_raw=-1 udp4_raw_mtu=1500 I can't seem to find any parameters in the siptrace module that controls fragmentation of the packets to Hepic. Any suggestions on how else to troubleshoot this? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

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