- sr-users - kamailio.org

Register without user create
by José Antonio Gutiérrez Delgado 29 Jul '19

29 Jul '19

Hello, I would like to know if it is possible to enable registration for users that have not been created with kamctl. For example, if I can register a user with a random name without a password, or obtain from another database other than kamailio; I would also like it to be allowed at the same time as there are users with password registered with kamctl. Thanks you, José Antonio Gutiérrez Delgado Responsable técnico Oficina: (+34) 923040031 <http://www.arsoft-company.com> *De conformidad a lo establecido en el REGLAMENTO (UE) 2016/679 DEL PARLAMENTO EUROPEO Y DEL CONSEJO de 27 de abril de 2016 relativo a la protección de las personas físicas en lo que respecta al tratamiento de datos personales y a la libre circulación de estos datos y por el que se deroga la Directiva 95/46/CE (Reglamento general de protección de datos), le informamos que todos los datos pasaran a formar parte de un fichero automatizado cuyo responsable es AUGMENTED REALITY SOFTWARE S.L siendo la única finalidad de dicho fichero la gestión de carácter comercial y el posible envío de comunicaciones comerciales sobre nuestros productos. Si lo desea puede ejercer sus derechos de acceso, rectificación, cancelación y oposición de sus datos en C/ Duero 12, Parque Científico, ARSOFT, 37185 o enviando un mensaje electrónico a info(a)arsoft-company.com <info(a)arsoft-company.com> indicando en el asunto el derecho que desea ejercitar.Este mensaje y sus archivos adjuntos van dirigidos exclusivamente a su destinatario, pudiendo contener información confidencial sometida a secreto profesional. No está permitida su reproducción o distribución sin nuestra autorización expresa. Si usted no es el destinatario final por favor elimínelo e infórmenos por esta vía.*

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Dialog missing when BYE is received from trunk
by Andrew White 28 Jul '19

28 Jul '19

Hi all, I’ve got a call flow where PBX makes a call via Kamailio, which uses the UAC modules to make the call to the trunk. This works great, however I’ve noticed recently that when the BYE comes from the trunk, Kamailio doesn’t recognise the dialog and throws back a 404. Due to this, the PBX never knows the session has ended, and keeps its session open. Here’s the flow from Wireshark: https://i.imgur.com/BxPRrXP.png <https://i.imgur.com/BxPRrXP.png> As you can see, the BYE is received from the trunk, and Kamailio throws a 404. I checked with an rpc dlg.list and at this point the dialog had gone. 30 seconds later, I threw a BYE from the PBX, and because the dialog no longer existed, it got a 481 back. The INVITE to the trunk has the following Record-Route header: Record-Route: <sip:kamailio.public-ip;lr;ftag=mjKtcKaBteQ4Q;vst=AAAAAAYFBQUHAwYHBgRzMwIRGRMcAgJBWBEVCwAbBhtADxBjdGVkLmNvbS5hdQ--;did=624.b2e1> When the first BYE comes in, it still has same did in the Route header. Below I’ve put the output from the dialog list rpc call (before the first BYE) and the relevant log lines. { h_entry: 1062 h_id: 7723 ref: 2 call-id: a9d32620-2b68-1238-048a-0cc47a18705a from_uri: sip:61400123123@pbx.local:5060 to_uri: sip:61400123123@10.0.0.2 state: 4 start_ts: 1564269942 init_ts: 1564269929 end_ts: 0 timeout: 1564313142 lifetime: 43200 dflags: 643 sflags: 0 iflags: 16 caller: { tag: mjKtcKaBteQ4Q contact: sip:pbx.local cseq: 7581172 route_set: socket: udp:10.0.0.2:5060 } callee: { tag: SD5ptp899-1177933201-1564269931305 contact: sip:SDd3mt8-nn9jnirnmqcrvfmvkpm0hmhvgggjgv2io788of70s84u9@trunk.remote:5060;transport=udp cseq: 0 route_set: socket: udp:10.0.0.2:5060 } profiles: { } variables: { { cseq_diff: 1 } } } Jul 28 09:27:30 ip-10-0-0-2 /usr/local/sbin/kamailio[11061]: NOTICE: {2 450917085 BYE a9d32620-2b68-1238-048a-0cc47a18705a} acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1564270050;method=BYE;from_tag=SD5ptp899-1177933201-1564269931305;to_tag=mjKtcKaBteQ4Q;call_id=a9d32620-2b68-1238-048a-0cc47a18705a;code=404;reason=Not here;src_user=0400123123;src_domain=trunk.remote;src_ip=trunk.remote;dst_ouser=0400123123;dst_user=0400123123;dst_domain=kamailio.public-ip I’m assuming the 404 is because the Contact header I sent from Kamailio -> Trunk is rewritten, that Contact is Kamailio’s public contact IP. However the dialog has the caller contact within it referencing back to the local PBX, so I would think when Kamailio gets this dialog ID back with a BYE it would relay it back to the original contact, wouldn’t it? Thanks! Andrew

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IMS_ISC technical specification
by Ali Shirvani 26 Jul '19

26 Jul '19

Hi all, I read some portion of imc_isc_mod.c, specially isc_from_as function. In that function there is some function call that searching for some marks in the SIP message. Also I read section 5.7.3 from TS 24.229 but I didn't find the marking mechanism that used in isc_from_as function. Would you please point me to the technical specification that implemented in IMS_ISC module? Regards, Ali

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Concatenating strings in $hdr
by David Cunningham 26 Jul '19

26 Jul '19

Hello, I'm having an issue where this doesn't work: #!trydef PRODUCT "Foo" if( $hdr("X-" + PRODUCT + "-Transport") != "" ) { add_uri_param( $hdr("X-" + PRODUCT + "-Transport") ); } It seems Kamailio doesn't like the concatenation within the $hdr(). Concatenation does work in other places though: append_hf( "X-" + PRODUCT + "-Transport: tcp\r\n" ); Can anyone advise how to make the concatenation within $hdr() work? Thanking you in advance. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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IM Chat facility in Kamailio
by Gaurav Bmotra 25 Jul '19

25 Jul '19

hi i m using kamailio 5.1.2 , on Ubuntu 18. LTS existing setup is working fine with audio and video call ,,, ut now i wan to add chat feature in it ... so plz help me how can i implement chat IM in to it thanks -- *Regards:* Gaurav Kumar

2 1

Malformed Route Header
by Duarte Rocha 25 Jul '19

25 Jul '19

Greetings, When a in-dialog request has Route headers, Kamailio will remove the Route Header with it's address when routing the packet. This works perfectly for me in every situation. However, in one of the calls I found a weird behaviour in the removal. So, this is the scenario : Kamailio has two IP's 1.1.1.209 and 1.1.1.205. This is BYE request received in the on 1.1.1.209 The header Route looks like this when it arrives at 1.1.1.209 : Route : <sip:1.1.1.209;r2=on;lr;ftag=5AE0303037363132017D0E65;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQECA3ByAwMcHwIdHQsWHwUMOA--;vst=AAAAABgHDQgBAwsIBQECdnIDAxwfAh0aBB4cAgU5;did=8f5.67c2>,<sip:1.1.1.205;r2=on;lr;ftag=5AE0303037363132017D0E65;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQECA3ByAwMcHwIdHQsWHwUMOA--;vst=AAAAABgHDQgBAwsIBQECdnIDAxwfAh0aBB4cAgU5;did=8f5.67c2> It reaches 1.1.1.205 like this : Route : As you can see, the header is empty and it should contain the 1.1.1.205 URI. Is the header Router malformatted and creating this error ? Best Regards, Duarte Rocha

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Re: [SR-Users] [EXT] Re: P-CSCF and I-CSCF ports -RESOLVED
by Woscek, Martin W. 24 Jul '19

24 Jul '19

Hi Ali, Looking through the source code, its not that the port for I-CSCF is hard coded to 5060, its just that it defaults to 5060 if the following is set in the Kamailio.cfg file: dns_try_naptr=no which we had left it disabled. Well I assumed I had all the DNS worked out, I had NAPTR records set for the P,I and S-CSCF. Looking though the pcscf.cfg there is a setting for NETWORKNAME which is set to ims522.ims.org, and is a separate FQDN from the P,I,S-cscf FQDN’s. I always kept the NAPTR rec for ims522 as the same port of the P-CSCF (pcscf522) in DNS. Two things I needed to fix: 1. Enable the DNS NAPTR lookups: dns_try_naptr=yes 2. Modify the NAPR for ims522 to be that of 4060 Thanks to you and Amar for the pointing to DNS. _Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Ali Shirvani Sent: Tuesday, July 23, 2019 1:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] P-CSCF and I-CSCF ports Hi Martin, As Amar pointed out it may some DNS configuration. You can check the DNS entry of I-CSCF with: $ dig -t srv <FQDN of I-CSCF> Regards, Ali On Tue, Jul 23, 2019 at 7:15 PM Amar Tinawi <amar.tinawi(a)gmail.com<mailto:amar.tinawi@gmail.com>> wrote: Hello Martin I think you are missing the DNS part. On Tue, Jul 23, 2019, 5:30 PM Woscek, Martin W. <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> wrote: Hi IMS users, I can configure the P-CSCF to use 5060, and the I-CSCF to use 4060 and both come up listening to their respective ports when each module is started. But the P-CSCF is hardcoded to use 5060 for the I-CSCF. Before I make code changes to the executable: Is there some additional configuration to allow for me to switch ports? Why is Kamailio IMS hardcoded and why not enforce it at the .cfg load time? Thanks, Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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kamailio 4.4.4 fills IPv6 contact address when sending packet over IPv4
by Pali Rohár 24 Jul '19

24 Jul '19

Hi! I detected strange problem that sip.linphone.org refuse to accept presence information re-transmitted by kamailio 4.4.4 server. I debug this problem with tcpdump and I found out that problem is in kamailio which fills IPv6 address into UDP datagram and that datagram is sent via IPv4 socket to IPv4 address, to sip.linphone.org server. And sip.linphone.org server does not have IPv6 connectivity, so correctly return over IPv4 to sender just "400 Bad Contact Header" error. On my server is running kamailio 4.4.4 from Debian Stretch and I can 100% reproduce this problem against public sip.linphone.org server. My server has both IPv4 and IPv6 connectivity and kamailio is listening for both IPv4 and IPv6 connections. So why is kamailio sending IPv6 address over IPv4 and therefore makes it impossible to communicate with non-IPv6 enabled servers? Looks like a problem with choosing default/correct socket for Contact header. And how to fix this problem? Can you help me? I would like to have working interconnection with linphone servers. Just to note I'm seeing this problem only for presence information packets. Other requests, like INVITE or MESSAGE seems to work. Below is relevant tcpdump output. Some parts were replaced by {VAR}. PS: I'm not subscribed to list, so please CC my address when sending reply. Thank you! 17:22:58.121719 IP (tos 0x10, ttl 64, id 21629, offset 0, flags [none], proto UDP (17), length 1266) {MY_IPV4_ADDRESS}.5060 > 91.121.209.194.5060: [bad udp cksum 0xa099 -> 0x9825!] SIP, length: 1238 NOTIFY sip:{REMOTE_NAME}@{REMOTE_USER_IPV4_ADDRESS}:5060;registering_acc=sip_linphone_org SIP/2.0 Via: SIP/2.0/UDP {MY_IPV4_ADDRESS};branch=z9hG4bK2b55.88f93c20000000000000000000000000.0 To: <sip:{REMOTE_NAME}@sip.linphone.org>;tag=75559182 From: <sip:{MY_SIP_URI}>;tag=97d8e785fdf42bf9622a64c13c504961-2708 CSeq: 2 NOTIFY Call-ID: 26cf9d5c019af2dc3302b770887bcc2e@0:0:0:0:0:0:0:0 Route: <sip:91.121.209.194:5060;lr> Content-Length: 597 User-Agent: kamailio (4.4.4 (x86_64/linux)) Max-Forwards: 70 Event: presence Contact: <sip:{MY_IPV6_ADDRESS}:5060;transport=udp> Subscription-State: active;expires=3600 Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="Pali <sip:{MY_SIP_URI}>"> <tuple id="sg89ae"> <status><basic>open</basic></status> <contact priority="0.8">Pali <sip:{MY_SIP_URI}></contact> </tuple> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="TA0C538B2"> <status> <basic>closed</basic> </status> <contact priority="1">sip:{MY_SIP_URI}</contact> <timestamp>2019-04-19T17:20:36+02:00</timestamp> </tuple> </presence> 17:22:58.151188 IP (tos 0x0, ttl 52, id 22949, offset 0, flags [none], proto UDP (17), length 373) 91.121.209.194.5060 > {MY_IPV4_ADDRESS}.5060: [udp sum ok] SIP, length: 345 SIP/2.0 400 Bad Contact Header Via: SIP/2.0/UDP {MY_IPV4_ADDRESS};branch=z9hG4bK2b55.88f93c20000000000000000000000000.0;rport=5060 From: <sip:{MY_SIP_URI}>;tag=97d8e785fdf42bf9622a64c13c504961-2708 To: <sip:{REMOTE_NAME}@sip.linphone.org>;tag=75559182 Call-ID: 26cf9d5c019af2dc3302b770887bcc2e@0:0:0:0:0:0:0:0 CSeq: 2 NOTIFY Content-Length: 0 -- Pali Rohár pali.rohar(a)gmail.com

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How to hook to Register And Unregister event in Kamailio
by Anuran Barman 24 Jul '19

24 Jul '19

Hi, I am integrating Kamailio into my application. I want to hook to the successful REGISTER and unregister event into Kamailio into my application. For now, I am able to hook into INVITE event and can hit my server to send a email to the callee user that user X is calling you. The way I am doing is by this if (is_method("INVITE")) { xlog("LOG_LOCAL3","L_INFO","invite came ($fU) ($tU)"); $var(res) = http_connect("sipnodejsserver", "/","text/plain","src_user:$fU,dst_user:$tU" ,"$avp(route)"); xlog("LOG_LOCAL3","L_INFO","request sent $avp(route) $var(res)"); setflag(FLT_ACC); # do accounting } I need to show into my application that their SIP phone is online or not. How can I hook into this?

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Re: [SR-Users] [EXT] Re: P-CSCF and I-CSCF ports
by Woscek, Martin W. 23 Jul '19

23 Jul '19

DNS is fine. I flip the NAPTR records when I flip back and forth the 4060/5060 btwn the P and I. Martin Woscek The MITRE Corp. mwoscek(a)mitre.org<mailto:mwoscek@mitre.org> Phone: 703-983-2650 FAX: 703-983-7142 From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Ali Shirvani Sent: Tuesday, July 23, 2019 1:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] P-CSCF and I-CSCF ports Hi Martin, As Amar pointed out it may some DNS configuration. You can check the DNS entry of I-CSCF with: $ dig -t srv <FQDN of I-CSCF> Regards, Ali On Tue, Jul 23, 2019 at 7:15 PM Amar Tinawi <amar.tinawi(a)gmail.com<mailto:amar.tinawi@gmail.com>> wrote: Hello Martin I think you are missing the DNS part. On Tue, Jul 23, 2019, 5:30 PM Woscek, Martin W. <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> wrote: Hi IMS users, I can configure the P-CSCF to use 5060, and the I-CSCF to use 4060 and both come up listening to their respective ports when each module is started. But the P-CSCF is hardcoded to use 5060 for the I-CSCF. Before I make code changes to the executable: Is there some additional configuration to allow for me to switch ports? Why is Kamailio IMS hardcoded and why not enforce it at the .cfg load time? Thanks, Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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P-CSCF and I-CSCF ports
by Woscek, Martin W. 23 Jul '19

23 Jul '19

Hi IMS users, I can configure the P-CSCF to use 5060, and the I-CSCF to use 4060 and both come up listening to their respective ports when each module is started. But the P-CSCF is hardcoded to use 5060 for the I-CSCF. Before I make code changes to the executable: Is there some additional configuration to allow for me to switch ports? Why is Kamailio IMS hardcoded and why not enforce it at the .cfg load time? Thanks, Martin

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Debugging kamailio module
by Ali Shirvani 23 Jul '19

23 Jul '19

Hi all, I want to trace function calls in a kamailio module, e.g. ims_isc module. I installed kamailio-dbg package and started the kamailio with gdb: `gdb kamailio` I also set break point on target function, but it seems gdb doesn't stop on that break point. How should I start gdb and set break points? it seems gdb couldn't handle kamailio child processes that forked in start. Regards, Ali

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by Carlos Cruz 23 Jul '19

23 Jul '19

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by Carlos Cruz 23 Jul '19

23 Jul '19

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Two GWs with the same IP
by Duarte Rocha 22 Jul '19

22 Jul '19

Greetings, I'm building some testing procedures for my Kamailio proxy. In order to do that i have a SipTester (Acting as Calling Number) from where i want to originate calls to Kamailio. After Kamailio processes the call, i want it to route it to SipTester again so it can act as Called Number. This should work with only one SipTester in one machine, so i created two SipTester GW's on Kamailio, belonging to different users and GW groups. Also, the two GWs use different Kamailio addresses to communicate with the proxy. However, in this combination, when routing i get this error " ERROR: dialog [dlg_handlers.c:666]: pre_match_parse(): bad request or missing CALLID/TO hdr :-/". If i change the destination or the origin of the INVITE the calls work without a problem. Does this happen because both GW's have the same address? Best Regards

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Reading presence state
by Gertjan Wolzak 19 Jul '19

19 Jul '19

Hello Kamailions, I have been able to get the presence configured, subscribers see the status of their subscriptions, this is working fine, for internal, incoming external and external calls. This was solved by enabling use_pubruri_avps in the pua_dialoginfo module. Now I want to be able to route calls based on the presence state. Using the presentity table could be used by doing a query based on the userpart of the ruri, which works fine when the call is still active, but it can take a while for the record to be removed from the presentity table, as that is dependend on the expires value. So when a query is made after a call is finished, a record would still be present in the presentity table till clean_period timer has hit again and the expires value is expired. So I should get the status from memory, but I have not been able to find the information on how to do that. Can someone enlighten me on where to find that information? Rgds, Gertjan

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I need to hold Leg A of a call for a random amount of time
by Gaurav Bmotra 19 Jul '19

19 Jul '19

hi i m using kamailio where i m tring to hold the call bitween switch A and Switch B PSTN -> Switch A -> SWITCHB -> SwitchC -> DESTINATION When PSTN hangs up, I need the call to stay up between SwitchA and B for an arbitrary number of seconds or minute plz help thank you -- *Regards:* Gaurav Kumar

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Kamailio 5.2.3 failed to start due to incompatible openssl version
by Andrew Chen 18 Jul '19

18 Jul '19

Hey support, Today I ran into this error on my latest 5.2.3 Kamailio box: Jul 18 15:05:28 sjomaintpsg51 /usr/sbin/kamailio[5525]: CRITICAL: tls [tls_init.c:671]: init_tls_h(): installed openssl library version is too different from the library the kamailio tls module was compiled with: installed “OpenSSL 1.1.1 11 Sep 2018” (0x1010100f), compiled “OpenSSL 1.1.0g 2 Nov 2017" (0x1010007f).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) There's not support for openssl1.1.1 ? 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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kamailio + asterisk integration RTP flow question
by Satish Patel 18 Jul '19

18 Jul '19

I am configuring kamailio as registar and asterisk for media services. in kamailio + asterisk integration how two SIP clients RTP traffic will flow, are they go via Asterisk or directly from kamailio ?

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SIP-DTLS support
by Martin Meadows 17 Jul '19

17 Jul '19

Hello, Has there been any movement on the support of SIP-DTLS (datagram-TLS) within Kamailio since: * sr-users:2012-October:075185.html <https://lists.kamailio.org/pipermail/sr-users/2012-October/075185.html> I'm interested to see if Kamailio can proxy 5061/udp to a FreeSWITCH 5060/udp. In general, I have been unable to find any material on implementations of: * draft-jennings-sip-dtls-05.html <https://tools.ietf.org/id/draft-jennings-sip-dtls-05.html> Cheers - Martin

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opt_reply(): called for non-OPTIONS request
by Alexandru Covalschi 17 Jul '19

17 Jul '19

Hello list, if KSR.is_OPTIONS() and KSR.is_myself_ruri(): KSR.x.modf("options_reply"); exit(); with a valid URI (no username) gives the subj output (opt_reply(): called for non-OPTIONS request) How could that be? Python KEMI, 5.2.3, docker. ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

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Kamailio IMS to IP PBX - how?
by David Dean 16 Jul '19

16 Jul '19

I'm trying to connect a VoLTE smartphone to an IP PBX using Kamailio IMS. I've managed to get the smartphone to connect to Kamailio IMS and make calls with our devices registered with Kamailio. The problem is that I need the smartphones to register with the IP PBX which is a VigorBX 2000n. This is so they can use an internal extension number and make calls to other local extensions with hard wired desk phones. Should I be using Kamailio IMS to "proxy" the smartphone SIP connection to the VigorBX PBX? Or would it be better to connect the smartphone directly to the VigorBX using SIP? If so, how? I've read absolutely every how-to document I could lay my hands on but none of them describe integrating IMS with a PBX. Thanks for any help / advice you can offer. Dave

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KEMI beginner question on alpine Linux
by Karsten Horsmann 16 Jul '19

16 Jul '19

Hi List, I play around with alpine Linux 3.10.x and Kamailio 5.2.x and I search for app_python3 and KEMI support. But I don't found it. AFAIK if I want to use KEMI with Python3 I need of course Kamailio and app_python3 (maybe something else?). Is the package maintainer on this list? Cheers Karsten

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How to get total number of dialogs in profile with value
by Nathan Angelacos 15 Jul '19

15 Jul '19

In kamailio < 5.1.x it used to be possible to get total number of items in a dialog_with_profile by skipping the "value" paramter, such as: modparam ( "dialog", "profiles_with_value", "outbound" ) . . . get_profile_size ( "outbound", "$var(myvar)" ); However, 5.1.x and 5.2.x report: ERROR: dialog [dialog.c:912]: w_get_profile_size_helper(): invalid value parameter The documentation ( https://kamailio.org/docs/modules/5.2.x/modules/dialog.html#dialog.f.get_pr… ) Indicates the "value" paramter is still optional. I'm sure I must be missing something obvious... Any ideas? Thanks!

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UDP fragmentation issues
by Gerry | x164 15 Jul '19

15 Jul '19

Hi, I have setup Kamailio on Centos 7 as a proxy in front of Asterisk in the same datacenter, using the dispatcher module. I have noted that large INVITEs sent by Kamailio do not arrive at the Asterisk box. It seems the network is dropping fragmented UDP traffic. This does not happen however when I contact the Asterisk box directly from outside the datacenter. Then large INVITES are arriving are Asterisk. So the problem is either in Kamailio or in the datacenter. I have analysed the problem with ngrep and tcpdump/wireshark. In normal mode (udp4_raw=no setting) Kamailio is producing UDP packets above 1500 bytes and leaves all fragmentation to the operating system. Enabling raw udp in Kamailio and lowering the MTU in the server didnt help. I even disabled offloading of UDP fragmentation to the NIC to get to the bottom of this. However nothing helps. Only forcing TCP in the dispatcher works when having large INVITES going to Asterisk. In any case, it appears that there is no guarantee that fragmented UDP arrives at all. As far as I am aware, the recommendation is to use TCP if messages are bigger than 1400 bytes. If so, how can I force Kamailio to send messages over 1400 bytes only in TCP mode? Or is there some other way how to deal with this? Thanks for any hints. Gerry

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