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37378 discussions

Change in LCR Debug
by Social Boh 26 Jan '24

26 Jan '24

Hello, using DEBUG with LCR module to see how the module apply the rules, I see he use Gateway Index in the list. The Gateway Index could be different to Gateway ID and maybe can confuse the correct interpretation about the use of the rules. Is it possible change the code: matched_gws[gw_index].gw_index = t->gw_index; matched_gws[gw_index].rule_id = rule->rule_id; matched_gws[gw_index].prefix_len = pl->prefix_len; matched_gws[gw_index].priority = t->priority; matched_gws[gw_index].weight = t->weight * (kam_rand() >> 8); matched_gws[gw_index].duplicate = 0; LM_DBG("added matched_gws[%d]=[%u %u, %u, %u, %u]\n", gw_index, t->gw_index, pl->prefix_len, t->priority, matched_gws[gw_index].weight); to use gateway ID and not gateway Index? Thank you Regards -- --- I'm SoCIaL, MayBe

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How to get media IP
by mm e 26 Jan '24

26 Jan '24

When receiving a call I get the following message： INVITE sip:123456789@192.168.1.118:29363 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.118:65178;branch=z9hG4bK-d87543-4a4a6b6c49541b2c-1--d87543-;rport Max-Forwards: 70 Contact: <sip:96341@192.168.1.118:65178> To: "123456789"<sip:123456789@192.168.1.119:29363 > From: "96341"<sip:96341@192.168.1.119:29363 >;tag=af3c7c31 Call-ID: OTFhN2M2YmRmNmU2N2I1ZmQxNmM4ODg4YzRiZmQyNTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1011d stamp 40820 Content-Length: 339 v=0 o=- 3 2 IN IP4 192.168.1.118 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.118 t=0 0 m=audio 32170 RTP/AVP 0 8 18 101 a=alt:1 1 : +YhwgPgy ayLEx4wo 192.168.1.118 32170 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv How to print the media IP in SDP separately in the log? I didn't find the Pseudo-Variables..can you help me

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412 Conditional request failed
by satyaprakash ch 26 Jan '24

26 Jan '24

Hi There were two PUBLISH requests were sent by the Kamailio Proxy server to the Kamailio Presence server. Processing of the first PUBLISH request resulted in the presentity table being updated - the "etag" changed from "a.1705053846.16611.21.1" to "a.1705053846.16613.23.2" Processing of the second PUBLISH failed because it also attempted to update the etag in the presentity table but at this point in time there was no match to the original etag value in the database Logs in kamailio.log: ------ ps_db_update_presentity(): No E-Tag match a.1705053846.16611.21.1 DEBUG: tm [t_reply.c:637]: _reply_light(): reply sent out - buf=0x7f81e5ca5fc0: SIP/2.0 412 Conditio... shmem=0x7f81de6da068: SIP/2.0 412 Conditional request failed. Will any one suggest how to resolve the issue of 412 conditional request failed.

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Is the Contact header required for the dialog module?
by Unai Rodriguez 26 Jan '24

26 Jan '24

Dear List, I’m trying to set up the dialog module and I keep getting the error "bad sip message or missing Contact hdr”: --- Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:219]: populate_leg_info(): bad sip message or missing Contact hdr Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:952]: dlg_new_dialog(): could not add further info to the dialog --- This is for MRCP traffic and the SIP INVITEs do not contain the Contact header. They look like this: ---- INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0 Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ Max-Forwards: 70 From: <sip:10.10.10.10:5099>;tag=134a3eKD2a8Ua To: <sip:some.mrcp.server.example.com:8060> Call-ID: 051df0ff-3541-123d-f090-02f373ff014c CSeq: 78491159 INVITE User-Agent: google_stt Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE Supported: timer, 100rel Content-Type: application/sdp Content-Disposition: session Content-Length: 329 v=0 o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10 s=- c=IN IP4 10.10.10.10 t=0 0 m=application 9 TCP/MRCPv2 1 a=setup:active a=connection:new a=resource:speechrecog a=cmid:1 m=audio 12034 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 L16/8000 a=sendonly a=mid:1 ---- According to the RFC, the Contact header is not mandatory (https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t have an easy way of adding the Contact header to the INVITEs. Is there a way to configure Kamailio’s dialog module to do without the Contact header? Any pointers appreciated. Thank you so much With best wishes, Unai Rodriguez

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How to assign xavp array to htable?
by Benoit Panizzon 26 Jan '24

26 Jan '24

Hi List Via sql_xquery I get a stacked xavp aka array. I would like to store that stacked xavp into a hash table for later usage in other transactions. Is this possible? Via xarp_parameters explode or implode does not work as that would stringy the same key multiple times. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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How to set a branch specific variable?
by Benoit Panizzon 25 Jan '24

25 Jan '24

Hi Gang I would like to set a variable specific to the branch. Is there a way? append_branch($var(aor1)); $(branch(dst_uri)[-1]) = $var($dst1) $(branch(want_plus)[-1]) = 1; append_branch($var(aor2)); $(branch(dst_uri)[-1]) = $var($dst2) $(branch(want_plus)[-1]) = 0; t_on_branch("BR_T"); branch_route[BR_T] { if ($branch(want_plus)) { $rU = "+" + $rU; } } Or would there be a way to query the index of the added branch so I could stack an avp with that index to access in the branch route? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Re: STIR/SHAKEN
by Anthony Wittig 25 Jan '24

25 Jan '24

Hey everyone, I've set it up so that when the invite comes into Kamailio I suspend the transaction (save the transaction info in sht) and use dispatcher to send a request to our NSS server. On the reply from NSS I throw the identity header into a sht and continue the original transaction. Does that seem like the right way to do it? The next things I need to look at are: * if first NSS server doesn't respond quickly, try the second (change some timeouts and use dispatcher to try the backup) * if both NSS servers aren't working, I still want to resume the original call and let it through. Maybe I can keep track of the failures in a sht and if I have two failures, resume the original invite without the identity header. Sound like an ok plan? -- Anthony Wittig Mango Voice Developer > > ---------- Forwarded message ---------- > From: Anthony Wittig <awittig(a)mangovoice.com> > To: sr-users(a)lists.kamailio.org, david.villasmil.work(a)gmail.com > Cc: > Bcc: > Date: Tue, 23 Jan 2024 20:39:09 -0700 > Subject: [SR-Users] STIR/SHAKEN > Thanks! > > This is what I've got so far: > > 1. Kamailio invite -> NSS > 2. NSS 302 -> Kamailio > 3. Kamailio ack -> NSS > > I can see the identity header on the 302 response in my onreply_route. How > can I replay the same invite from step #1 with the identity header I get > back in step #2? Should I be doing something with a transaction on step #1, > maybe start a branch that goes out to NSS and when it comes back resume the > transaction (these are all words I've heard smart people say, but I don't > fully understand them...). > > Any pointers would be greatly appreciated, thanks! > -- > Anthony Wittig > Mango Voice Developer > > ---------- Forwarded message ---------- > From: David Villasmil <david.villasmil.work(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Tue, 23 Jan 2024 14:56:52 +0100 > Subject: [SR-Users] Re: STIR/SHAKEN > Ditto, i do first approach. > > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > > > On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> These are all valid approaches, depending on preference. Catching 302s >> and extracting exactly the desired info might be easier with Kamailio, so I >> suppose, ceteris paribus, I'd recommend the first one. >> >> Dispatcher is a great approach! >> >> > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users < >> sr-users(a)lists.kamailio.org> wrote: >> > >> > Hello, >> > >> > We're attempting to add identities to our invites. I believe we want to >> send our invites to the NSS, and it'll reply with a 302'd invite with an >> identity header. We currently use FreeSWITCH for dialplans and media and >> Kamailio as our SBC. >> > >> > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS >> (302) -> Kamailio -> PSTN >> > >> > or would it make more sense to do: FreeSWITCH -> NSS (302) -> >> FreeSWITCH -> Kamailio -> PSTN >> > >> > Assuming we wanted Kamailio -> NSS, is using dispatcher the right >> approach? We currently use it for our two Kamailios and our FreeSWITCHs. >> We're going to have two NSS servers, one for failover. >> > >> > Any thoughts would be greatly appreciated, thanks! >> > >> > -- >> > Anthony Wittig >> > Mango Voice Developer >> > __________________________________________________________ >> > Kamailio - Users Mailing List - Non Commercial Discussions >> > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >> > Important: keep the mailing list in the recipients, do not reply only >> to the sender! >> > Edit mailing list options or unsubscribe: >> >> -- >> Alex Balashov >> Principal Consultant >> Evariste Systems LLC >> Web: https://evaristesys.com >> Tel: +1-706-510-6800 >> >

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STIR/SHAKEN
by Anthony Wittig 24 Jan '24

24 Jan '24

Thanks! This is what I've got so far: 1. Kamailio invite -> NSS 2. NSS 302 -> Kamailio 3. Kamailio ack -> NSS I can see the identity header on the 302 response in my onreply_route. How can I replay the same invite from step #1 with the identity header I get back in step #2? Should I be doing something with a transaction on step #1, maybe start a branch that goes out to NSS and when it comes back resume the transaction (these are all words I've heard smart people say, but I don't fully understand them...). Any pointers would be greatly appreciated, thanks! -- Anthony Wittig Mango Voice Developer ---------- Forwarded message ---------- From: David Villasmil <david.villasmil.work(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Cc: Bcc: Date: Tue, 23 Jan 2024 14:56:52 +0100 Subject: [SR-Users] Re: STIR/SHAKEN Ditto, i do first approach. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users < sr-users(a)lists.kamailio.org> wrote: > These are all valid approaches, depending on preference. Catching 302s and > extracting exactly the desired info might be easier with Kamailio, so I > suppose, ceteris paribus, I'd recommend the first one. > > Dispatcher is a great approach! > > > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users < > sr-users(a)lists.kamailio.org> wrote: > > > > Hello, > > > > We're attempting to add identities to our invites. I believe we want to > send our invites to the NSS, and it'll reply with a 302'd invite with an > identity header. We currently use FreeSWITCH for dialplans and media and > Kamailio as our SBC. > > > > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS > (302) -> Kamailio -> PSTN > > > > or would it make more sense to do: FreeSWITCH -> NSS (302) -> FreeSWITCH > -> Kamailio -> PSTN > > > > Assuming we wanted Kamailio -> NSS, is using dispatcher the right > approach? We currently use it for our two Kamailios and our FreeSWITCHs. > We're going to have two NSS servers, one for failover. > > > > Any thoughts would be greatly appreciated, thanks! > > > > -- > > Anthony Wittig > > Mango Voice Developer > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions > > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply only to > the sender! > > Edit mailing list options or unsubscribe: > > -- > Alex Balashov > Principal Consultant > Evariste Systems LLC > Web: https://evaristesys.com > Tel: +1-706-510-6800 > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: >

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STIR/SHAKEN
by Anthony Wittig 23 Jan '24

23 Jan '24

Hello, We're attempting to add identities to our invites. I believe we want to send our invites to the NSS, and it'll reply with a 302'd invite with an identity header. We currently use FreeSWITCH for dialplans and media and Kamailio as our SBC. Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS (302) -> Kamailio -> PSTN or would it make more sense to do: FreeSWITCH -> NSS (302) -> FreeSWITCH -> Kamailio -> PSTN Assuming we wanted Kamailio -> NSS, is using dispatcher the right approach? We currently use it for our two Kamailios and our FreeSWITCHs. We're going to have two NSS servers, one for failover. Any thoughts would be greatly appreciated, thanks! -- Anthony Wittig Mango Voice Developer

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How to check for branches with a pending reply in MANAGE_REPLY to correctly handle rtpengine?
by Benoit Panizzon 23 Jan '24

23 Jan '24

Hi gang location database contains multiple contacts for an AOR The invite therefore is parallel branched to those multiple contacts. Each reply from each branch is handled by MANAGE_REPLY. rtpengine_manage() is called in MANAGE_REPLY to stop rtpengine processing a failed call. Unfortunately, if any of the contacts replies with an error rtpengine_manage() will stop RTP processing. Is there a way I could check if there are branches with a pending reply from within MANAGE_REPLY so I could call rtpengine_manage() only when the last reply (no pending replies) are present? Hmm, realizing, if I process the last pending error reply and a prior one got a positive reply, this would still kill media. => How do I solve this? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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