- sr-users - kamailio.org

Dispatcher and SUBSCRIBE issue
by Barry Flanagan 08 Apr '21

08 Apr '21

Hi I am using the dispatcher module with Kamailio as a front-end Registrar/LB in front of a cluster of Asterisk boxes. I have come across an issue with SUBSCRIBE messages not getting routed to the correct server in the event that the phone endpoint's dispatch set is changed. It will still go to the old server until such time as the phone is rebooted. The reason this is happening is because the 200 OK in the response from the Asterisk server has Contact set to its own IP:port and subsequent in-dialog reSUBSCRIBE are sent with the original Asterisk ip:port in the RURI. When Kamailio receives these reSUBSCRIBE it does not change the RURI even though ds_select_dst is being called in the [WITHIN_DIALOG] route. My feeling is I have two choices - either alter the Contact in the reply from Asterisk, or, change the RURI manually on receipt of an in-dialog SUBSCRIBE. Has anyone else come across this, or have advice on the best way to handle this? Thanks -- -Barry

2 6

No private identity specified
by Dong Ming (董明) 08 Apr '21

08 Apr '21

Thank you for your reply！ It's the first time to learn ims, I am very sorry for lack of knowledge and inaccurate description of this problem. I sincerely hope that you can give me some advice how to deal with this problem. thank you！发件人: Mojtaba <mespio(a)gmail.com> 发送时间: 2021年4月8日 17:02 收件人: Dong Ming (董明) <dongming(a)twsz.com> 抄送: Daniel-Constantin Mierla <miconda(a)gmail.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> 主题: Re: [SR-Users] sip registration fail Just be notic, being behind nat, cause your request to pcscf is rejected and the same issue (No private identity specified) is appeared in lon info. The public identity (IMPU) and private identity (IMPI) are created with the same pattern by default as shown in your attachment, but the title of the issue is confusing. On Thu, Apr 8, 2021 at 1:08 PM Dong Ming (董明) <dongming(a)twsz.com<mailto:dongming@twsz.com>> wrote: Hello According to the debug information, ================================================ 12(4516) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): SIP Request: 12(4516) DEBUG: <core> [core/parser/msg_parser.c:612]: parse_msg(): method: <REGISTER> 12(4516) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): uri: <sip:scscf.ims.mnc001.mcc460.3gppnetwork.org:6060<http://scscf.ims.mnc001.mcc460.3gppnetwork.org:6060>> 12(4516) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 12(4516) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 12(4516) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [57]; uri=[sip:460010000000001@ims.mnc001.mcc460.3gppnetwork.org<mailto:sip%3A460010000000001@ims.mnc001.mcc460.3gppnetwork.org>] 12(4516) DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body [<sip:460010000000001@ims.mnc001.mcc460.3gppnetwork.org<mailto:sip%3A460010000000001@ims.mnc001.mcc460.3gppnetwork.org>> ], to tag [] 12(4516) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <2> <REGISTER> 12(4516) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK11a4.1097491452b96d42664a6b234646cc60.0>; state=6 12(4516) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 236, <i> = <1>; state=16 12(4516) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 12(4516) DEBUG: <core> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2 12(4516) DEBUG: <core> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via 12(4516) INFO: ims_auth [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_user_name: Failed finding avp (avp_code = 1, vendor_id = 0) 12(4516) ERROR: ims_auth [cxdx_mar.c:151]: async_cdp_callback(): No private identity specified (Authorization: username) ================================================ The above content is found in the log of s-cscf,It seems that the authentication failed. May I ask how to configure this part, or if this part of the code is deleted, will it affect the subsequent process? Thank you very much！发件人: Mojtaba <mespio(a)gmail.com<mailto:mespio@gmail.com>> 发送时间: 2021年4月8日 15:48 收件人: Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> 抄送: Dong Ming (董明) <dongming(a)twsz.com<mailto:dongming@twsz.com>> 主题: Re: [SR-Users] sip registration fail It seems the issue is related to some default configuration in IMS nodes in(pcscf, icscf and scscf). In pcap file, just pcscf node is appeared. please send full signaling flow in whole nodes. Thanks On Thu, Apr 8, 2021 at 11:06 AM Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>> wrote: Hello, On 07.04.21 10:40, Dong Ming (董明) wrote: Hello I have a problem about volte(401 Unauthorized). P-CSCF: 4(4467) INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header present. 4(4467) INFO: ims_registrar_pcscf [sec_agree.c:296]: cscf_get_security_verify(): No sec urity-verify parameters found Can you help me see where the problem is？ Tips：About SQN, the SQN is the same at the beginning of the test (Open5GS, FOHSS), but when it is found that the sip registration fails, the SQN is inconsistent. If this situation is normal? If it is not normal, how to solve it? I am not familiar with the IMS extensions, if nobody else comes with other suggestions, I recommend to set debug=3 in your kamailio configuration file and then look at the syslog, there should be a lot of DEBUG messages that may give more hints about what happens. Also, when sending Kamailio logs, do not make screen shots, just copy and paste them, it is easier to read as well as search in the source code for them by copy&paste. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/ _______________________________________________ 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 -- --Mojtaba Esfandiari.S 本电子邮件（包括任何的附件）为本公司保密文件。本文件仅仅可为以上指定的收件人或公司使用，如果阁下非电子邮件所指定之收件人，那么阁下对该邮件部分或全部的泄漏、阅览、复印、变更、散布或对邮件内容的使用都是被严格禁止的。如果阁下接收了该错误传送的电子邮件，敬请阁下通过回复该邮件的方式立即通知寄件人，同时删除你所接收到的文本。 This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. -- --Mojtaba Esfandiari.S 本电子邮件（包括任何的附件）为本公司保密文件。本文件仅仅可为以上指定的收件人或公司使用，如果阁下非电子邮件所指定之收件人，那么阁下对该邮件部分或全部的泄漏、阅览、复印、变更、散布或对邮件内容的使用都是被严格禁止的。如果阁下接收了该错误传送的电子邮件，敬请阁下通过回复该邮件的方式立即通知寄件人，同时删除你所接收到的文本。 This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

2 1

sip registration fail
by Dong Ming (董明) 08 Apr '21

08 Apr '21

Hello I have a problem about volte(401 Unauthorized). P-CSCF: 4(4467) INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header present. 4(4467) INFO: ims_registrar_pcscf [sec_agree.c:296]: cscf_get_security_verify(): No sec urity-verify parameters found Can you help me see where the problem is？ Tips：About SQN, the SQN is the same at the beginning of the test (Open5GS, FOHSS), but when it is found that the sip registration fails, the SQN is inconsistent. If this situation is normal? If it is not normal, how to solve it? open5gs：v2.2.3 kamailio：5.3 Thanks! 本电子邮件（包括任何的附件）为本公司保密文件。本文件仅仅可为以上指定的收件人或公司使用，如果阁下非电子邮件所指定之收件人，那么阁下对该邮件部分或全部的泄漏、阅览、复印、变更、散布或对邮件内容的使用都是被严格禁止的。如果阁下接收了该错误传送的电子邮件，敬请阁下通过回复该邮件的方式立即通知寄件人，同时删除你所接收到的文本。 This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

3 4

SSL handshake failed : X509 - Certificate verification failed, e.g. CRL, CA or signature check failed
by amitsharma＠coraltele.com 07 Apr '21

07 Apr '21

Hello Team, I am using Linphone with Kamailio and facing *SSL handshake failed : X509 - Certificate verification failed, e.g. CRL, CA or signature check failed* error over *TLS*. Following error is coming Apr 7 15:54:36 irisivdxcloud /usr/sbin/kamailio[9089]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 172.105.49.156 Apr 7 15:54:36 irisivdxcloud /usr/sbin/kamailio[9089]: ERROR: <core> [core/tcp_read.c:1500]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f2c7f85a2e0 r: 0x7f2c7f85a408 (-1) Apr 7 15:54:36 irisivdxcloud /usr/sbin/kamailio[9036]: WARNING: {1 20 REGISTER ~OkQyvT70a} sanity [sanity.c:612]: check_cl(): content length header missing in request Apr 7 15:54:40 irisivdxcloud /usr/sbin/kamailio[9040]: WARNING: {1 20 REGISTER ~OkQyvT70a} sanity [sanity.c:612]: check_cl(): content length header missing in request Apr 7 15:54:43 irisivdxcloud /usr/sbin/kamailio[9071]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Apr 7 15:54:43 irisivdxcloud /usr/sbin/kamailio[9071]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094418:SSL routines:ssl3_read_bytes:tlsv1 alert unknown ca Apr 7 15:54:43 irisivdxcloud /usr/sbin/kamailio[9071]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 49.36.149.254 Apr 7 15:54:43 irisivdxcloud /usr/sbin/kamailio[9071]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 172.105.49.156 Apr 7 15:54:43 irisivdxcloud /usr/sbin/kamailio[9071]: ERROR: <core> [core/tcp_read.c:1500]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f2c7f85a2e0 r: 0x7f2c7f85a408 (-1) Apr 7 15:54:44 irisivdxcloud /usr/sbin/kamailio[9042]: WARNING: {1 20 REGISTER ~OkQyvT70a} sanity [sanity.c:612]: check_cl(): content length header missing in request Apr 7 15:54:44 irisivdxcloud /usr/sbin/kamailio[9072]: INFO: {1 674 REGISTER 583608109(a)192.168.1.105} <script>: REGISTRATION CALLED: 740,Count: 0,Contact: sip:740@117.242.44.49:3668;transport=TCP,Domain: location Please suggest, what is this issue all about. Thanks & Regards, Amit Sharma (Sr. Team Leader) (An ISO 9001:2008 company) Mobile: <tel:9891612004> tel:9891612004 PH: +91 120 2595870 Ext.: <tel:870> tel:870 Email : <mailto:amitsharma@coraltele.com> amitsharma(a)coraltele.com Web : <blocked::http://www.coraltele.com> www.coraltele.com

2 1

fix_nated_sdp function
by Ilie Soltanici 06 Apr '21

06 Apr '21

Hello, Can someone point me to what is wrong with this SDP: ``` v=0 o=- 3826710256 3826710256 IN IP4 100.113.232.22 s=pjmedia b=AS:386 t=0 0 a=X-nat:0 m=audio 4012 RTP/SAVP 96 8 120 121 c=IN IP4 100.113.232.22 b=TIAS:96000 a=rtcp:4013 IN IP4 100.113.232.22 a=sendrecv a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/48000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=ssrc:886437437 cname:61141a4c6ecaa55e a=crypto:1 AES_256_CM_HMAC_SHA1_80 inline:J9gds2sgq3OQccZ4fQoy3Oh9P2sWAXGHpLx+v/Ykh3gFQtAcaD8u5zpygEw2Dw== a=crypto:2 AES_256_CM_HMAC_SHA1_32 inline:yVCi20viRSxW+7hkUDjDBfXX4eAsuNrGj96ByTpDejOww8A0+BwSDjO40U1u3A== a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:9vWxWlHSBcLYv0p+qYTviva7Bc6tUb+fKblHaQ1D a=crypto:4 AES_CM_128_HMAC_SHA1_32 inline:UlLlQR2NB1owPDx7fFyGqnq09AVFVOLB50I3f/FC ``` I'm trying to apply function "fix_nated_sdp" to it, but getting back this error: <no `IP[4|6]' in `a=rtcp' field> By looking at the source code, seems that the nextissip value is 0, and that's why it fails (nextissip value:0/len:14) Any idea why?

2 1

SNMP counter doesn't increase in stateful proxy
by Marat Gareev 06 Apr '21

06 Apr '21

Hello, I'm trying to get the total number of SIP request messages sent out by the proxy with *kamailioSIPSummaryOutRequest* scalar. But the value doesn't increase if I use t_relay() (with forward() counter works). Related thread <https://opensips.org/pipermail/users/2012-December/024045.html>. Is there another SNMP counter which contains a total number of OUT requests? Marat

3 6

ERROR: connect_unix_sock: connect(/var/run/kamailio//kamailio_ctl): No such file or directory [2]
by Fadi Hamdan 06 Apr '21

06 Apr '21

Hi there, When I try to run: kazoo-kamailio status I'm getting this error: ERROR: connect_unix_sock: connect(/var/run/kamailio//kamailio_ctl): No such file or directory [2] Any solution for this error? Cheers, Hamdan, Fadi 24B Carrick Glen Avenue, Flat Bush, Auckland 2019 New Zealand fadi.h(a)msn.com<mailto:%20fadi.h@msn.com> M +64 21 066 0088

3 10

SDP Change
by Nick Meyjes 06 Apr '21

06 Apr '21

Hi, I am trying to send calls to a provider that is switching to Nokia SBCs. They dont like seeing any IP other than our kamailio SBC in the messaging, I changed the contact but I cant find a way to change the owner line in the SDP I have try mangler and SDPOPS to change but neither seem to work on changing or removing the owner line Any ideas? Nick

3 3

How to handle SIP-I(ISUP) with Kamailio
by Waseem Zaidi 06 Apr '21

06 Apr '21

Hi All: I am new to the list, I am struggling to find out Kamailio's capabilities with SIP-I/ISUP. I see that there is SIP-T module in kamailio and ss7ops module, but I need to get a hold on how to start. I would appreciate if someone could guide me. previously, I tried to use OpenSIPS SIP-I to extract ISUP from incoming calls, i was able to extract ISUP parameters in SDP body as xlog("Called Number: $isup_param_str(Called Party Number|Address signal)\r\n"); xlog("Calling Number: $isup_param_str(Calling Party Number|Address signal)\r\n"); xlog("Location is: $isup_param_str(Location Number)\r\n"); Location parameter is appearing as 0x04971400330268 which is wrongly interpreted. its proper interpretation should appear as 4100332086. I also tried below: xlog("Location is: $isup_param_str(Location Number|Address signal)\r\n"); but this subfield is not present. The transformation didn't make any difference: xlog("Location NUM: $(var(isup_body){isup.param, Location Number} Help pleeez ?

2 2

Phase III clinical trials of Kamailio DMQ
by Alex Balashov 01 Apr '21

01 Apr '21

For immediate release: GAINESVILLE, Florida (1 April 2021)--Evariste Systems, in partnership with Posner | Palner Pharmaceuticals, are pleased to announce the eagerly anticipated start of Phase III clinical trials of Kamailio DMQ module replication. In this stage of the investigation of the promising replication scheme, which was developed in record time, DMQ will be tested in a rigourous, double-blind, controlled CI/CD process for large-scale efficacy and continually monitored for safety in shorter and longer terms. The now-concluded second phase established basic safety and Phase 2b entailed a dose-finding study for the optimal number of batched `dmq_usrloc` contacts per kilogram of cloud computing resource. The main cohort of the main sequence of the Phase III trial will enroll 380 randomly selected, statistically representative Floridian SIP registrars and 5700 SIP dialogs, also in the state of Florida. When asked about the disparity in the number of registrars versus dialogs, Alex Balashov of Evariste said, "It is reflective of the small number of SIP registrars in Florida; not to be indelicate, but consider the median age of the population." As previously remarked upon in proceedings of the World SIP Organisation (WSO), the choice of Florida as a venue attracted some questions. We asked Fred Posner of Posner | Palner about the clinical background informing this choice of replication study cohort: "Florida is a good place to find eventual consistency failures; the open climate and reluctance to accept traditional data integrity science would have sealed the deal, but when people started to drink bleach here, I saw an opportunity." Evariste and Posner | Palner are aiming for a US Food & Drug Administration EUA (Emergency Use Authorisation) for DMQ replication as a prophylactic data redundancy and failover measure to ward off the development of moderate to severe viral Sonus Haemorrhagic Fever symptoms. Phase II studies have painted a promising a picture of the DMQ architecture's ability to reduce moderate to severe symptoms and associated 28-day mortality by more than 90%, albeit over a small sample size and in enterprise patients with known pre-existing comorbidities such as Session Border Controller and BroadSoft Multi-System Inflammatory Syndrome. Phase III trials seek to bolster quantums of broader efficacy, including efficacy for localised outbreaks of small to medium business softswitch. Of the control group, Balashov said, "They will receive a standard pipe-to-/dev/null placebo protocol, but with identical '200 OK' confirmation messages. We proudly developed this approach in early 2020 in consultation with the highest levels of the Trump presidential administration; it is called TENP, or 'The Emperor's New Potion.' It is a successor to that administration's previous approved placebo protocol, SHTI - Small Hands, Tremendous Initiatives." Despite speculative autism and packet fragmentation claims by Florida Governor Ron DeSantis, Balashov was quick to provide reassurance about side effects from DMQ in the Phase III trial. "The incidence of severe allergies to plain-text UDP was within standard deviation over the mean for all insecure replication, and there's support for other transports on the way. Anyway, we have a robust adverse effects reporting system in our issue tracker and solid surveillance. Don't let partisan politics dissuade you from allowing request lines with the KDMQ method." Other industry sources have reported that DMQ is not the only replication investigation undertaken by Evariste and Posner | Palner, although it is the only prophylactic suitable for the early infection development and inflammatory phases of a broad class of SBC-complex syndromes typically afflicting service providers. A recent IETF preprint by the same authors has shown high-quality preliminary data suggesting statistically significant improvement in morbidity outcomes from Perimeta and Nokia via a later-stage therapeutic intervention with a monoclonal, monorepo Github cocktail administered intravenously. -- 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 1

Kamailio + Websocket issues
by Vinicius Kwiecien Ruoso 01 Apr '21

01 Apr '21

Hi! I'm using Kamaio in front of multiple Asterisk instances. At this moment it works as a SIP over Websocket proxy, with rtpengine, for browser clients to connect to Asterisk using WebRTC. I do not use the registration module of Kamailio, as each backend Asterisk is independent and handles its own registrations. Everything works great when making calls from the browser, and the routing is correctly executed by Kamailio based on the request SIP domain. We have an internal routing API that it calls to discover which backend Asterisk to route the calls. The issue I have is when a call initiates from that backend Asterisk, trying to reach a contact that is connected in Kamailio via the websocket. The Asterisk sends the message to the proxy, and Kamailio must route it to the corresponding websocket. I've tried a few approaches: - using add_contact_alias + handle_ruri_alias: I have the alias with alias=<ip>~<port>~ws in the contact registration, but for some reason handle_ruri_alias cannot use it - using the Path module on Asterisk, so when registering, the path is recorded and sent back from Asterisk, Kamailio is also not respecting that - Using contact_param_encode and contact_param_encode and contact_param_decode_ruri, but the encoded sip address is always the invalid websocket, like sip:58c0ktrg@5hp0nn5hqqv9.invalid;transport=ws None with success. Any hints on that can be wrong? I can share more detailed information. Greetings, Vinicius

4 11

tcp timeout in kamailio
by Alexander Kudrevatykh 01 Apr '21

01 Apr '21

Hi. Is there an option for tcp timeout in kamailio? In my setup we have tcp peers and kamailio pings them every 5 seconds, but when tcp session stucks kamailio not close this session for 15 minutes. I tried many tcp options kamailio supports, but any of them helps. I think kamailio can use TCP_USER_TIMEOUT on linux to detect such tcp sessions and reestablish failed connections?

2 1

dispatcher algorithm 11, no more than 25 hosts in group for proper distribution
by Ivan Matvieiev 31 Mar '21

31 Mar '21

Hello Community, There is a module dispatcher and sharing algorithm 11 that features the ability to redistribute load when the host is enabled or disabled that is quite valuable for me. In the test lab I faced unexpected behavior: - when I add to group <=25 hosts algorithm behaves as per description. - when a group includes 26 or more hosts, and 100 calls are done for instance, then about 25 go to the first host and 75 are distributed among the rest of hosts. Expected behavior: all hosts of the group having the same rweight value must receive the same amount of calls no mater what number of hosts is in the group. Could you please to help to figure out what is wrong and why when the dispatcher group consists of 26+ hosts dispatching algorithm 11 behaves differently than when group consists of 25 or less of hosts? Some configuration notes: # module configs modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_threshold", 3) modparam("dispatcher", "ds_inactive_threshold", 10) modparam("dispatcher", "ds_probing_mode", 3) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_ping_reply_codes", "501,403,404,400,200") modparam("dispatcher", "ds_ping_from",DS_PING_FROM_PARAM) modparam("dispatcher", "use_default", 0) # dispatching algorithm call if ( ds_is_from_list("101")) { sl_send_reply("100","My calls"); ds_select_dst("100", "11"); return; } # dispatcher list group config 100 sip:10.60.27.123:7000 0 10 rweight=50 100 sip:10.60.27.123:7001 0 10 rweight=50 100 sip:10.60.27.123:7002 0 10 rweight=50 100 sip:10.60.27.123:7003 0 10 rweight=50 100 sip:10.60.27.123:7004 0 10 rweight=50 100 sip:10.60.27.123:7005 0 10 rweight=50 100 sip:10.60.27.123:7006 0 10 rweight=50 100 sip:10.60.27.123:7007 0 10 rweight=50 100 sip:10.60.27.123:7008 0 10 rweight=50 100 sip:10.60.27.123:7009 0 10 rweight=50 100 sip:10.60.27.123:7010 0 10 rweight=50 100 sip:10.60.27.123:7011 0 10 rweight=50 100 sip:10.60.27.123:7012 0 10 rweight=50 100 sip:10.60.27.123:7013 0 10 rweight=50 100 sip:10.60.27.123:7014 0 10 rweight=50 100 sip:10.60.27.123:7015 0 10 rweight=50 100 sip:10.60.27.123:7016 0 10 rweight=50 100 sip:10.60.27.123:7017 0 10 rweight=50 100 sip:10.60.27.123:7018 0 10 rweight=50 100 sip:10.60.27.123:7019 0 10 rweight=50 100 sip:10.60.27.123:7020 0 10 rweight=50 100 sip:10.60.27.123:7021 0 10 rweight=50 100 sip:10.60.27.123:7022 0 10 rweight=50 100 sip:10.60.27.123:7023 0 10 rweight=50 100 sip:10.60.27.123:7024 0 10 rweight=50 Kamailio version: kamailio -v version: kamailio 5.4.4 (x86_64/linux) e16352 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: e16352 compiled on 15:56:46 Feb 15 2021 with gcc 4.8.5 System info CentOS Linux release 7.7.1908 (Core) Linux test-carrier-1.loc 3.10.0-1062.4.1.el7.x86_64 #1 SMP Fri Oct 18 17:15:30 UTC 2019 x86_64 x86_64 x86_64 GNU/Linux

3 3

communication between two Kamailio
by Gustavo Monteiro 31 Mar '21

31 Mar '21

Hello, I need to do the simple communication between two Kamailio, ex: [Side A - K1 (10.0.0.0/24)- UE numbers 500] <----------> [Side B - K2 (192.168.0.0/24) UE numbers 400]. I already looked in the documentation, and I was surprised that I didn't find anything like that. I don't know which module or configuration to use. Thanks Gustavo (BR) Legal Disclaimer : The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message!

2 1

How to Play RBT With Media Server
by amitsharma＠coraltele.com 30 Mar '21

30 Mar '21

Dear Team, I am using SBC as proxy server and I am using Freeswitch for providing RBT/MOH. Please guide how can I play RBT from Freeswitch while using SBC as proxy server. Thanks & Regards, Amit Sharma (Sr. Team Leader) (An ISO 9001:2008 company) Mobile: <tel:9891612004> tel:9891612004 PH: +91 120 2595870 Ext.: <tel:870> tel:870 Email : <mailto:amitsharma@coraltele.com> amitsharma(a)coraltele.com Web : <blocked::http://www.coraltele.com> www.coraltele.com

4 3

Strange behavior with $T_branch_idx in failure_route
by Victor Litvinov 25 Mar '21

25 Mar '21

Hello. We faced an issue that $T_branch_idx called from failure_route returned a strange result. Few words about our setup. We use branch indexes like key in avp when we store via-branch value for rtpengine. route[RTPMANAGE] ... if ( is_request() and is_method("INVITE") and !has_totag() and t_is_branch_route() ) { $avp(viabranch$T_branch_idx) = $branch_str; } $avp(extra_branch_id) = $avp(viabranch$T_branch_idx); $var(rtpkey) = $var(rtpkey) + " via-branch=extra"; ...... rtpengine_manage($var(rtpkey)); Call this route in branch_route and in failure_route. In branch_route T_branch_idx is 0, in failure_route is 1. In documentation: the index (starting with 1 for the first branch) of the branch for which is executed the branch_route[]. If used outside of branch_route[] block, the value is '0'. But in source code behavior is different. (modules/tmx/t_var.c:480): for BRANCH_FAILURE_ROUTE and BRANCH_ROUTE it is idx = tcx->branch_index, but for FAILURE_ROUTE it is idx = t->nr_of_outgoings + nr_branches; Can you describe why for failure_route it is different and what we can do in our case? And can you update documentation for this variable if it is wrong?

2 1

Using sngrep for visualising encrypted SIP traffic
by Ovidiu Sas 24 Mar '21

24 Mar '21

Hello all, Here's a quick recipe to ease the troubleshooting of encrypted SIP traffic: https://voipembedded.wordpress.com/2021/03/22/troubleshooting-kamailio-encr… Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

3 4

SDPOPs questions
by Gerry | Rigatta 23 Mar '21

23 Mar '21

Hi, The behaviour of the sdpops module is not fully clear to me. Lets say in the INVITE contains in the m= line the payloads 0 99 but has no a= line for codec 0 / PCMU, Would the sdp_keep_codecs_by_name(“PCMU”) keep PCMU ? What about the case that the INVITE contains in the m=line only the payload 99 but shows an additional a=line for codec 0 / PCMU. Would the sdp_keep_codecs_by_name(“PCMU”) keep PCMU ? If yes, would the extra mtype parameter prevent that? e.g. sdp_keep_codecs_by_name(“PCMU”, mtype) What happens when several m= lines are present? Does sdp_keep_codecs_by_name consider all m=lines ? Are the treated separately? What happens to payloads which are not in the hard-coded list in sdpops_data.c ? Can they be used with the sdpops functions? Thanks for your hints. Best Gerry codecsmap_t sdpops_codecsmap_table[] = { { {"PCMU", 4}, {"0", 1} }, { {"GSM", 3}, {"3", 1} }, { {"G723", 4}, {"4", 1} }, { {"DVI4", 4}, {"5,6,16,17", 9} }, { {"LPC", 3}, {"7", 1} }, { {"PCMA", 4}, {"8", 1} }, { {"G722", 4}, {"9", 1} }, { {"L16", 3}, {"10,11", 5} }, { {"QCELP", 5}, {"12", 2} }, { {"CN", 2}, {"13", 5} }, { {"MPA", 3}, {"14", 2} }, { {"G728", 4}, {"15", 2} }, { {"G729", 4}, {"18", 2} }, { {0, 0}, {0, 0} } };

2 1

Dynamic CSEQ Updates
by Tyler Moore 23 Mar '21

23 Mar '21

Hey kamailio team, When are the callbacks in the dialog module for updating $dlg_var(cseq_diff) called? I am trying to dynamically enable / disable cseq updates (some specific carriers need it and some don't) by setting $dlg_var(cseq_diff)=$null prior to sending the request but it seems like the callback happens after relaying. For example I tried various locations: - prior to t_relay() - in onreply_route[] - in failure_route[] - in event_route[tm:local-request] - in event_route[tm:local-response] I also tried setting modparam("dialog", "initial_cbs_inscript", 1) and calling dlg_manage() prior to routing and those didn't seem to be the right callbacks either. Either the callback that updates $dlg_var(cseq_diff) occurs after t_relay() / forward() is called or setting $dlg_var(cseq_diff)=$null has no effect. Is dynamically turning CSEQ updates on / off possible by overwriting $dlg_var(cseq_diff)? Or would this require new logic in the dialog module to be supported? Regards, *Tyler Moore* <https://devopsec.net/> Full Stack Software Engineer <http://www.google.com/search?q=full+stack+software+engineer+job+description> Flyball Labs <http://goflyball.com/> Office: 888-907-2085, ext: 34 <8889072085;ext=34> Cell: 248-909-2769 <2489092769> Email: tmoore(a)goflyball.com [image: FLYBALL TECHNICAL SOLUTIONS, LLC]

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Info: Freezing development for Kamailio v5.5.x series - March 31, 2021
by Daniel-Constantin Mierla 22 Mar '21

22 Mar '21

Hello, this is the notification to let everybody know that the development of Kamailio v5.5.x series will be frozen at the end of March 31, 2021. Should anyone plan to add new features/modules and be part of v5.5.x, then they must be committed to git master branch or a pull request created by that date. Afterwards, the new features will have to be hold for a few weeks till branch 5.5 is created and development branch unfrozen, and they will be part of the next future release v5.6.x (or other version number to be decided later). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

JANSSON Delete from JSON
by Jeremy McNamara 22 Mar '21

22 Mar '21

Hi Folks - I am using JANSSON to store JSON data in an htable. How does one remove an entire element from a JSON? I do not see a specific function mentioning clear or remove exposed in the Kamailio module, but do see various clear/remove methods in the underlying C library. Below does set the value of key foo to null. I would like to remove both the key and value of foo. jansson_set("null", "$var(foo)", "$var(json)", "$var(result)"); Running Kamailio v5.4 Debian 10 -Jeremy

2 4

Kamailio + rtpengine + Freeswitch on public/private network
by Eric Ravelo 20 Mar '21

20 Mar '21

Dear Kamailians, I have Kamailio+rtpengine as a dispatcher in front of FreeSWITCH. Kamailio and FreeSWITCH are on the same private network (in an AWS VPC). Kamailio receives calls on the Public Internet IP address and port, and dispatches to FreeSWITCH on the private network. All calls are coming from a SIP Trunk provider, from the Internet so no NAT on this side. My Hello World scenario is: FreeSWITCH answers the calls, plays a media file and disconnects. Problems: 1) In the INVITE relayed from Kamailio to FreeSWITCH, I can see a Record-Route with the public IP of Kamailio. Hence, the BYE sent by FreeSWITCH is sent to this public IP, rather than going back to the Kamailio path, and the call never gets completed from the caller's point of view. I tried adding record_route() in request_route{} but nothing changes. 2) The SDP part of all the messages coming back and forth from Kamailio and FreeSWITCH all contain the correct private IPs, but still the media file can't be heard on the caller's side. Is RTPengine correctly bridging the RTPs? Kamailio configuration is here: https://gist.github.com/eravelo/b3f026723e27d42354804ef942b34a04 Rtpengine configuration file has: [rtpengine] table = 0 interface = internal/10.20.1.158;external/10.20.1.158!KAMAILIO_PUBLIC_IP listen-ng = 127.0.0.1:2223 timeout = 60 silent-timeout = 3600 tos = 184 port-min = 16384 port-max = 32768 Your help is appreciated. Thank you. Eric R.

2 2

received parameter in route URI not used despite path use_received set to 1
by Easy Rider 19 Mar '21

19 Mar '21

Hi I'd like to make sure I've understood correctly how use_received in path module is supposed to work, because I don't get what I expected to get after reading the documentation I've got a little set up with a UAC behind a NAT registering to a kamailio registrar through a kamailio edge proxy (he UAC is a simple sipp script simulating a UAC behind a NAT) path is enabled on both EP and registrar, and use_received is set to 1 on EP the path, contact and via headers in the REGISTER going from the EP to the registrar are : Via: SIP/2.0/UDP 192.168.26.129;branch=z9hG4bK20da.9f4ab6ad25eae6acdd4bd4d655b320ce.0 Via: SIP/2.0/UDP 10.0.0.5:6060 ;received=192.168.26.1;rport=23456;branch=z9hG4bK-105-1-1 Contact: <sip:1000@10.0.0.5:6060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS" Path: <sip:inside@192.168.26.129;lr;received=sip:192.168.26.1:23456> where 192.168.26.129 is obviously the edge proxy then I place a call to the UAC through registrar the EP receiveds the following INVITE : INVITE sip:1000@10.0.0.5:6060 SIP/2.0 Record-Route: <sip:192.168.26.142;lr> Via: SIP/2.0/UDP 192.168.26.142;branch=z9hG4bKdf25.142eebc72e940d07ad983a858c67f64e.0 Route: <sip:inside@192.168.26.129;lr;received=sip:192.168.26.1:23456> Via: SIP/2.0/UDP 192.168.26.1:34567 ;received=192.168.26.1;rport=34567;branch=z9hG4bK-107-1-1 From: "1001" <sip:1001@my-fake-domain.com>;tag=107-1 To: <sip:1000@my-fake-domain.com;user=phone> CSeq: 1 INVITE Call-ID: 1-107(a)192.168.26.1 Contact: <sip:1001@192.168.26.1:34567> ... The Route header is exactly what is expected, but the edge proxy still sends the request to the request-URI address (10.0.0.5:6060) instead of sending the request to the received address set in the first Route header The path module documentation states : 3.1. use_received (int) If set to 1, the “received” parameter of the first Route URI is evaluated and used as destination-URI if present. kamailio configuration on the EP is almost the default one (just added path support, and modified the REGISTRAR route to forward the request instead on saving the contact, so there is not even a mid registrar on the EP tried with kamailio 5.3.7 and 5.4.4 thanks

3 4

Kamailio how to make UAC challenge on Dispatcher failure route
by Antonio Couto 18 Mar '21

18 Mar '21

I have one Kamailio that makes an outbound register on two Asterisk. I have a dispatcher with that same two Asterisk destinations, on INVITE message I call replace_from("", ) in order to call a internal extension and on failure route I receive 401 and call the process of authorization is OK. The problem is that if the first Asterisk fail and I get 500 on failure route I can not call replace_from() because is not valid on the block. Is there any way to make this type of authorization process after receiving a 500 on failure route of dispatcher ? Below my cfg file. Thanks in advance. Antonio Couto #!KAMAILIO#!ifdef WITH_DEBUG#!define DBGLEVEL 3#!else#!define DBGLEVEL 2#!endif # - flags# FLT_ - per transaction (message) flags# FLB_ - per branch flags#!define FLT_ACC 1#!define FLT_ACCMISSED 2#!define FLT_ACCFAILED 3#!define FLT_NATS 5 #!define FLB_NATB 6#!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0log_prefix="{$mt $hdr(CSeq) $ci} " children=8 auto_aliases=noenable_tls=noenable_sctp=no tcp_accept_no_cl=yes # SIP port listen=udp:192.168.0.201:5060# xhttp port listen=tcp:127.0.0.1:8181 loadmodule "xhttp.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "htable.so" loadmodule "db_text.so" loadmodule "dialog.so" loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 1) # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params -----# default retransmission timeout: 2sec modparam("tm", "fr_timer", 1500)# default invite retransmission timeout after 1xx: 40sec modparam("tm", "fr_inv_timer", 40000) # ----- rr params -----# set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) # account triggers (flags) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:proxy@kamailio.org") modparam("dispatcher", "ds_ping_interval", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_timer_mode", 1) modparam("db_text", "db_mode", 1) modparam("db_text", "emptystring", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "track_cseq_updates", 1) modparam("uac", "restore_dlg", 1) modparam("uac", "reg_db_url", "text:///etc/kamailio/dbtext") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 60) modparam("uac", "reg_contact_addr", "192.168.0.201:5060") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations if (is_method("INVITE")) { route(DISPATCH); } else { sl_send_reply("501","Not Implemented"); exit; } } # Dispatch requests route[DISPATCH] { if ($si == "192.168.0.44" and $sp == 5060) { xlog("L_ALERT","from switch $si:$sp)\n"); $var(groupid) = "2"; } else{ xlog("L_ALERT","from others $si:$sp)\n"); $var(groupid) = "1"; } if(!ds_select_dst($var(groupid), "11")) { send_reply("503", "Service Unavailable"); exit; } if ($du =~ "192.168.0.210") { xlog("L_ALERT", "SCRIPT: uac_replace_from\n"); uac_replace_from("", "sip:4000@192.168.0.201:5060"); } xlog("L_ALERT", "SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { $var(ts) = t_get_status_code(); xlog("L_ALERT","ALERT: failure_route[RTF_DISPATCH]] from $si:$rp status:$var(ts)\n"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { $avp(auser) = "4000"; $avp(apass) = "1234"; $avp(arealm) = "asterisk"; if (uac_auth()) { route(RELAY); } exit; } if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if (ds_next_dst()) { xlog("L_ALERT", "retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { if($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")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) { return; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("500","Server Internal Error"); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # jsonrpc dispatch event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } }

2 2

[Question] About softphone for VoLTE
by shigeru ishida 18 Mar '21

18 Mar '21

Hi all, I have one question. What softphones can I use when trying VoLTE (not VoIP) with Kamailio? I would like to verify VoLTE of private LTE with Open5GS and Kamailio. I'm sorry to ask you a question without much understanding of Kamailio, VoIP and VoLTE. Thank you very much in advance. Best regards, --Shigeru

3 7

Server requirements estimation
by Marat Gareev 17 Mar '21

17 Mar '21

Hello! I'm configuring Kamailio proxy and trying to find instructions or advises for choosing the following options: *Host specific* - Number of cores - RAM size *Kamailio specific* - Number of children to fork for the UDP interfaces - Shared memory size The proxy will not connect to the database, only route INVITEs using the dispatcher. The expected load is 70K requests per hour. Are there any formulas for calculating the parameters?

2 2

sip_trace() in branch_route, kamailio 5.1
by Daniel Donoghue 17 Mar '21

17 Mar '21

Hello, Using kamailio 5.1, I am trying to use sip_trace(<internal-sip-uri>) from a branch route to send a copy of the current sip message to another location. Sending the message works fine but I note that the message being sent is the original incoming request and does not contain what was changed for the particular branch. Is there any way to get sip_trace() to send the content of the message for that specific branch? With every blessing, — Daniel Donoghue

2 1

Kamailio how to make UAC challenge on Dispatcher failure route
by Antonio Couto 17 Mar '21

17 Mar '21

I have one Kamailio that makes an outbound register on two Asterisk. I have a dispatcher with that same two Asterisk destinations, on INVITE message I call replace_from("", ) in order to call a internal extension and on failure route I receive 401 and call the process of authorization is OK. The problem is that if the first Asterisk fail and I get 500 on failure route I can not call replace_from() because is not valid on the block. Is there any way to make this type of authorization process after receiving a 500 on failure route of dispatcher ? Below my cfg file. Thanks in advance. Antonio Couto #!KAMAILIO#!ifdef WITH_DEBUG#!define DBGLEVEL 3#!else#!define DBGLEVEL 2#!endif # - flags# FLT_ - per transaction (message) flags# FLB_ - per branch flags#!define FLT_ACC 1#!define FLT_ACCMISSED 2#!define FLT_ACCFAILED 3#!define FLT_NATS 5 #!define FLB_NATB 6#!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0log_prefix="{$mt $hdr(CSeq) $ci} " children=8 auto_aliases=noenable_tls=noenable_sctp=no tcp_accept_no_cl=yes # SIP port listen=udp:192.168.0.201:5060# xhttp port listen=tcp:127.0.0.1:8181 loadmodule "xhttp.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "htable.so" loadmodule "db_text.so" loadmodule "dialog.so" loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 1) # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params -----# default retransmission timeout: 2sec modparam("tm", "fr_timer", 1500)# default invite retransmission timeout after 1xx: 40sec modparam("tm", "fr_inv_timer", 40000) # ----- rr params -----# set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) # account triggers (flags) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:proxy@kamailio.org") modparam("dispatcher", "ds_ping_interval", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_timer_mode", 1) modparam("db_text", "db_mode", 1) modparam("db_text", "emptystring", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "track_cseq_updates", 1) modparam("uac", "restore_dlg", 1) modparam("uac", "reg_db_url", "text:///etc/kamailio/dbtext") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 60) modparam("uac", "reg_contact_addr", "192.168.0.201:5060") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations if (is_method("INVITE")) { route(DISPATCH); } else { sl_send_reply("501","Not Implemented"); exit; } } # Dispatch requests route[DISPATCH] { if ($si == "192.168.0.44" and $sp == 5060) { xlog("L_ALERT","from switch $si:$sp)\n"); $var(groupid) = "2"; } else{ xlog("L_ALERT","from others $si:$sp)\n"); $var(groupid) = "1"; } if(!ds_select_dst($var(groupid), "11")) { send_reply("503", "Service Unavailable"); exit; } if ($du =~ "192.168.0.210") { xlog("L_ALERT", "SCRIPT: uac_replace_from\n"); uac_replace_from("", "sip:4000@192.168.0.201:5060"); } xlog("L_ALERT", "SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { $var(ts) = t_get_status_code(); xlog("L_ALERT","ALERT: failure_route[RTF_DISPATCH]] from $si:$rp status:$var(ts)\n"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { $avp(auser) = "4000"; $avp(apass) = "1234"; $avp(arealm) = "asterisk"; if (uac_auth()) { route(RELAY); } exit; } if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if (ds_next_dst()) { xlog("L_ALERT", "retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { if($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")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) { return; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("500","Server Internal Error"); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # jsonrpc dispatch event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } }

2 1

kamcmd dlg.end_dlg not working in shell script
by Andrew Chen 16 Mar '21

16 Mar '21

Hi all, I have a script that cleans up hanging dialog and for some reason, executing 'kamcmd dlg.end_dlg' does not seem to work. Below I log events of stuck calls and as you can see it just keeps logging the same call until I manually run the kamcmd either in bash shell prompt or in kamcmd cli: 2021-03-08T19:00:04.347807+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:00:04 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:01:02.531822+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:01:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:02:02.689823+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:02:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:03:02.044960+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:03:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:04:02.436588+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:04:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:05:02.685572+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:05:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:06:02.775561+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:06:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:07:01.867021+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:07:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:08:01.888512+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:08:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:09:02.146887+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:09:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:10:02.649627+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:10:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:11:01.904948+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:11:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:12:01.932267+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:12:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:13:02.428128+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:13:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:14:02.617683+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:14:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:15:01.863099+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:15:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:16:01.650299+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:16:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:17:02.338576+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:17:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:18:01.743163+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:18:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:19:02.257369+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:19:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:20:01.624170+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:20:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:21:02.287343+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:21:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul Thoughts? 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.*

5 15

dmq support for all transports - udp, tcp, tls, sctp
by Daniel-Constantin Mierla 16 Mar '21

16 Mar '21

Hello, (cross-posting - notification for developers, but also looking for testing support in the users community). So far dmq had support only for UDP, suitable to use for replication over trusted networks (local private network, or vpn). There were couple of discussions in the past about supporting other transport like tcp or tls. Earlier today I pushed several commits to dmq module trying to remove the limitation on UDP, and support the rest of the base protocols available in Kamailio (TCP, TLS and SCTP). Internally, a significant behaviour change is that the first (startup) peer notification is now done in the SIP worker one process, instead of the main process, in order to have all transports initialized. I could not spot any relevant side effect that can happen based on quick analysis, because the code with UDP only support was sending out the KDMQ requests, the corresponding replies could come back later. Anyhow, if any other dmq developer thinks of something not being quite right, reply to the sr-dev mailing list to discuss further. For users community: it would be appreciated if people using dmq can test with other transports (TLS, TCP, ...) and report any problems on sr-users mailing list or open an issue on bug tracker. Interesting test scenarios would be with replication of large amount of data or high rate of replicated requests per second. My testing got to the level of connecting to peers via TLS, trying to ensure the code is no longer limiting to UDP. I will do more, but I thought of giving a notification in early phase and get some feedback in order to avoid advancing too much in a wrong direction. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Dialog using Recod Route, with an asterisk
by Nicki Otte 15 Mar '21

15 Mar '21

Hi everyone I have a question regarding the dialog module. It uses the RecordRoute header, which seems all fine. The situation: I have a setup, where the Kamailio proxy, which is running with the dialog module, is talking to an Asterisk. Now the logic in the Asterisk, is to place a new call, and return in to the Kamailio proxy, which sends it to the PSTN. By the logic of the asterisk, it doesn’t persist with the RecordRoute headers. It starts from scratch with a new callId. Here lies the problem. Because the RecordRoute header isn’t given, then a new “dialog” is made, instead of using the one that is established. Here comes my question. Is it against the standards to make the Asterisk persist with these headers, or am I looking at the wrong solution? If not, I do see the power of the dialog module, when a call bounces between UAS’, but not when an asterisk is in the setup. Another solution could be to use a custom x- header for the dialog id, but it doesn’t look like that is a possibility. Have any of you run into this situation, and might have a solution? Thanks in advance // Nicki Bo Otte

2 2

Re: [SR-Users] Trouble with Kamailio/RTPEngine and NAT/Firewall
by PJ Slauta 15 Mar '21

15 Mar '21

Hi Jeremy, We are currently using that configuration. -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of sr-users-request(a)lists.kamailio.org Sent: Sunday, March 14, 2021 7:00 AM To: sr-users(a)lists.kamailio.org Subject: sr-users Digest, Vol 190, Issue 13 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Flists.kama… or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: iOS CallKit and tsilo (Jeremy McNamara) 2. Re: Trouble with Kamailio/RTPEngine and NAT/Firewall (Jeremy McNamara) ---------------------------------------------------------------------- Message: 1 Date: Sat, 13 Mar 2021 12:43:55 -0500 From: Jeremy McNamara <mcnamara.jeremy(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] iOS CallKit and tsilo Message-ID: <CAO0X5JSeyYHB8MhWrxd9khdQNbSigfGVOVZBw43=z0rVwhi-GA(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi Igor - Are you following the example from Frederico? We have implemented his idea for our iOS and Android apps... Make sure to send over the SIP CallID in the push notification. That way if a CANCEL does get sent (on JOIN/RESUME) the app can react to it correctly. https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.kamail…

1 0

iOS CallKit and tsilo
by Igor Olhovskiy 14 Mar '21

14 Mar '21

Hello! Is there any way to "store" already finished transactions in tsilo? Idea is to deliver, for example, canceled calls to the phone, when call already was answered on other device, but push notification arrive later? Major problem here, that there how it's working on iOS. On iOS phone first show you calling screen, than - app is waking and after app will register and receive invite with tsilo, it updates calling screen with CallerID and other info. But if call was canceled before, calling screen is shown, but app not receiving INVITE, so, call screen is just there for some timeout (for Linphone, for ex, it's 20 sec). Right now I've manage to do it via external SIPP call, that emulates "fake missed call", but maybe there is other way to "store" already dead transactions for some time? PS: Unfortunately, can't solve this on mobile app level. -- Regards, Igor

4 10

Trouble with Kamailio/RTPEngine and NAT/Firewall
by PJ Slauta 13 Mar '21

13 Mar '21

Hi, I have a Kamailio/RTPEngine setup with Asterisk on a couple of EC2 instances. There is a public IP and it's setup to accept any ports from our IP addresses. If I DMZ my machine behind my router, I get audio, otherwise I don't get any audio, but the SIP portion of the call connects properly. It isn't specific to my network either, it's the same for my partner on his network. I'm assuming that NAT is getting in the way, although I'm not 100% if it's that or if it's a problem opening ports. I have tried with SIP/RTP Passthrough enabled without success. Can anyone provide some guidance here? Thanks, PJ Slauta

2 1

Open separate TCP streams to the same destination
by Sebastian Damm 12 Mar '21

12 Mar '21

Hi, I have a peer that requests separate TCP connections for each registration to their server. Those registrations run through the same Kamailio instance on my side. Now what I tried is to use tcpops and the tcp_set_otcpid function, supplying a previously nonexistant TCP connection ID before relaying the register request. The command itself returns true, however, it does not do anything. The content of $conid of the reply is something totally different. So I guess, this function only works for already existing connections. Is there another way to explicitly tell Kamailio to open up another connection to a peer even though a connection to that peer is already open? Thanks in advance. Regards, Sebastian

3 3

Comparing if header has 0 value in body
by Jurijs Ivolga 11 Mar '21

11 Mar '21

Hi, I'm running Kamailio 5.3.5 I have following header "X-myheader: 0" Nevertheless it seems this is not working for me: if ($hdr(X-myheader)==0) { ..... }; What I'm doing wrong? Jurijs

3 4

Best way to ignore HTTP requests
by Ben Kaufman 11 Mar '21

11 Mar '21

I've set up a server listening on TCP recently, and notice that I'm receiving intermittent, random HTTP requests from the internet. While it would probably be a good idea to enforce a firewall rule to only allow known hosts to communicate, what would be the best way within Kamailio to ignore http requests? Would just checking $rP work? Regards, Ben Kaufman ben.kaufman(a)altigen.com<mailto:ben.kaufman@altigen.com> Director of Cloud Operations AltiGen Communications, Inc.

3 4

Kamailio how to make UAC challenge on Dispatcher failure route
by Antonio Couto 10 Mar '21

10 Mar '21

I have one Kamailio that makes an outbound register on two Asterisk. I have a dispatcher with that same two Asterisk destinations, on INVITE message I call replace_from("", ) in order to call a internal extension and on failure route I receive 401 and call the process of authorization is OK. The problem is that if the first Asterisk fail and I get 500 on failure route I can not call replace_from() because is not valid on the block. Is there any way to make this type of authorization process after receiving a 500 on failure route of dispatcher ? Below my cfg file. Thanks in advance. Antonio Couto #!KAMAILIO#!ifdef WITH_DEBUG#!define DBGLEVEL 3#!else#!define DBGLEVEL 2#!endif # - flags# FLT_ - per transaction (message) flags# FLB_ - per branch flags#!define FLT_ACC 1#!define FLT_ACCMISSED 2#!define FLT_ACCFAILED 3#!define FLT_NATS 5 #!define FLB_NATB 6#!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0log_prefix="{$mt $hdr(CSeq) $ci} " children=8 auto_aliases=noenable_tls=noenable_sctp=no tcp_accept_no_cl=yes # SIP port listen=udp:192.168.0.201:5060# xhttp port listen=tcp:127.0.0.1:8181 loadmodule "xhttp.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "htable.so" loadmodule "db_text.so" loadmodule "dialog.so" loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 1) # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params -----# default retransmission timeout: 2sec modparam("tm", "fr_timer", 1500)# default invite retransmission timeout after 1xx: 40sec modparam("tm", "fr_inv_timer", 40000) # ----- rr params -----# set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) # account triggers (flags) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:proxy@kamailio.org") modparam("dispatcher", "ds_ping_interval", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_timer_mode", 1) modparam("db_text", "db_mode", 1) modparam("db_text", "emptystring", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "track_cseq_updates", 1) modparam("uac", "restore_dlg", 1) modparam("uac", "reg_db_url", "text:///etc/kamailio/dbtext") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 60) modparam("uac", "reg_contact_addr", "192.168.0.201:5060") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations if (is_method("INVITE")) { route(DISPATCH); } else { sl_send_reply("501","Not Implemented"); exit; } } # Dispatch requests route[DISPATCH] { if ($si == "192.168.0.44" and $sp == 5060) { xlog("L_ALERT","from switch $si:$sp)\n"); $var(groupid) = "2"; } else{ xlog("L_ALERT","from others $si:$sp)\n"); $var(groupid) = "1"; } if(!ds_select_dst($var(groupid), "11")) { send_reply("503", "Service Unavailable"); exit; } if ($du =~ "192.168.0.210") { xlog("L_ALERT", "SCRIPT: uac_replace_from\n"); uac_replace_from("", "sip:4000@192.168.0.201:5060"); } xlog("L_ALERT", "SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { $var(ts) = t_get_status_code(); xlog("L_ALERT","ALERT: failure_route[RTF_DISPATCH]] from $si:$rp status:$var(ts)\n"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { $avp(auser) = "4000"; $avp(apass) = "1234"; $avp(arealm) = "asterisk"; if (uac_auth()) { route(RELAY); } exit; } if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if (ds_next_dst()) { xlog("L_ALERT", "retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { if($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")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) { return; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("500","Server Internal Error"); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # jsonrpc dispatch event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } }

1 1

limits in hashtable
by Fabian Borot 10 Mar '21

10 Mar '21

Hi I have a hashtable with about 2500 elements in it. Some of the key_names are the same and in the config file and I traverse them using the index. The problem that I am having is that not all the values are loaded into memory. Lets say I have 5 entries with key_name and key_values like this: 1.2.34:5060:10.10.10 --> 0001232555690 1.2.34:5060:10.10.10 --> 0004562555691 1.2.34:5060:10.10.10 --> 0007892555692 1.2.34:5060:10.10.10 --> 0003212555693 1.2.34:5060:10.10.10 --> 0006542555694 when I do this kamcmd htable.get mytable 1.2.34:5060:10.10.10[0] kamcmd htable.get mytable 1.2.34:5060:10.10.10[1] kamcmd htable.get mytable 1.2.34:5060:10.10.10[2] kamcmd htable.get mytable 1.2.34:5060:10.10.10[3] kamcmd htable.get mytable 1.2.34:5060:10.10.10[4] I am supposed to get their values on the console but sometimes I can not get the 3rd or 4th value , instead I receive this error error: 500 - Key name doesn't exist in htable. What I have been doing so far is to delete extra/old keys from the physical hashtable and load the values again and then it works for a few days. But there is a process that adds entries to the hashtable and at some point it breaks again and I have to delete again old/unused ones. I can not use 'expire' here because the time it takes to become "old" is unknown/variable. This is how the table is defined..2^16 is more than enough to hold 2500 values. What am I missing here? modparam("htable", "htable", "a=>size=16;autoexpire=0;dbtable=htable;") I have taken pcap captures between the kamailio server and the db and I see all the values being transferred over the network when the kamailio is loading/refreshing the values from the db. Is there another limit somewhere? Also when I try to dump the values I get 'reply too big'. modparam("ctl", "binrpc_max_body_size", 10240) modparam("ctl", "binrpc_struct_max_body_size",10240) ./kamcmd htable.dump a ERROR: reply too big any assistance is highly appreciated fborot

3 2

Can't show single location via kamctl
by Jon Bonilla (Manwe) 09 Mar '21

09 Mar '21

Hi all Using version 5.4.x here. I save the registrations with use_domain 0 but I do use the domain module to do some checks. If I execute "kamctl ul" show I can see the registrations but if I execute "kamctl ul show username" I get "ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN" I understand I should use domain in case I was using it in the usrloc module but I'm not. How can I tell kamctl not to ask for domain? cheers, Jonm -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 4

Route - loose_route
by Ilie Soltanici 09 Mar '21

09 Mar '21

Hello, I have a strange issue here with the Route header not been respected by the Kamailio server. I'm getting the INVITE from the SBC-1, adding my own RR header (SBC-2), and send it to the SBC-3 This is the Invite I'm sending outside to the SBC-3 Proxy. ``` INVITE sip:+123456789@192.168.1.1:5060 SIP/2.0 Record-Route: <sip:SBC-2:5067;transport=tls;r2=on;ftag=as2b53df6f;lr> Record-Route: <sip:SBC-1;ftag=as2b53df6f;lr> ``` I'm getting back 200OK with mirrored RR, the call is established successfully. But, the problem is with the BYE message coming from the SBC-3: ``` BYE sip:+123456789@192.168.1.2:5060 SIP/2.0 ROUTE: <sip:SBC-2:5067;transport=tls;r2=on;ftag=as2b53df6f;lr>,<sip:SBC-1;ftag=as2b53df6f;lr> ``` There are 2 routes within a single header. Afaik, loose_function should remove his own Route header (SBC-2) and forward the message to the next Route (SBC-1), or to the URI if there are not other Routes. But, in my case, for some reason Kamailio on SBC-2 is sending the request directly to the URI, bypassing the SBC-1. In the logs I'm getting the following errors: *rr_do_force_send_socket(): no socket found to match second RR (sip:SBC-1)* *rr_do_force_send_socket(): second RR uri is not myself (sip:SBC-1)* To enforce the socket I was trying to use 'sn=interface', then I'm not getting anymore this error - but still, Kamailio sends the message directly to the URI, instead of SBC-1. Is there anything wrong with my config, or I am missing something with the Route headers? Kamailio Version: version: kamailio 5.4.4 (x86_64/linux) bdf4e0 Thank you.

2 1

Lookup issue with websockets contacts
by moisesynfante 09 Mar '21

09 Mar '21

I have configured Kamailio for websockets support and I'm having issues whenever I try to lookup a WS registered uac on the location table, it never finds the contact even though the contact is present when I dump the list via kamcmd. This issue only happens with WS, I tested using another user agent with a different protocol and it works fine, not sure if this is related to the fact that WS RURI doesn't really contain the AoR username. The AoR is something like 104-p4f578, but the contact's address stored is something like: sip:f0ck9vph@sjm8i3la4i7n.invalid;transport=ws; -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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TLS certificate unknown
by Daniel Hermann N'don 09 Mar '21

09 Mar '21

Hi guys, I am a beginner in kamailio technology. I want to use kamailio in secure mode with TLS. But when I try to connect via TLS, I get this error in terminal: Mar 08 12:29:36 KAMAILIO5 /usr/sbin/kamailio[1627]: ERROR: <core> [core/tcp_read.c:1512]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f6d71a82800 r: 0x7f6d71a828e8 (-1) Mar 08 12:29:44 KAMAILIO5 /usr/sbin/kamailio[1629]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094416:SSL routines:ssl3_read_bytes:sslv3 alert certificate unknown tls.cfg : [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /etc/certs/kamailio/kamailio-key.pem certificate = /etc/certs/kamailio/kamailio-cert.pem ca_list = /etc/certs/demoCA/kamailio-cert.pem I followed this tutorial to generate the ssl keys: http://www.kamailio.net/dokuwiki/doku.php/tls:create-certificates Can someone please help me?

2 1

client TLS session reuse problem
by Rupesh Kumar 09 Mar '21

09 Mar '21

Hi, I am running kamailio 5.2.6 on centos 8 and openssl 1.1.1c and connecting and using it as a proxy sip endpoints to a legacy PBX over TLS. The connection to the backend PBX is over TLS 1.2 . Whenever kamailio initiates a TLS connection to the PBX, it uses session ID and a random session id . The server side has a bug and cannot handle the TLS session resuse apparently because of some bug/issue in caching the TLS sessions. The renegotiation and session_cache is by default turned off and I also explicitly set to 0 via modparam but kamailio would always send the session ID in the initial client hello and this is causing us trouble. Any help would be greatly appreciated. https://kamailio.org/docs/modules/5.2.x/modules/tls.html#tls.p.renegotiation Regards, RK

2 2

Core dumps/crash handlers
by Alex Balashov 09 Mar '21

09 Mar '21

For memory bugs like this ... [mem/f_malloc.c:599]: fm_free(): BUG: fm_free: bad pointer 0x1 (out of memory block!), called from tm: h_table.c: free_cell_helper(185) - aborting ... I'd love to see a core dump, but unfortunately the SIGCHLD from the SIGABRT-terminated child is intercepted and leads to a graceful shutdown: [main.c:743]: handle_sigs(): child process 16037 exited by a signal 6 Is there any way Kamailio can be set to dump core when this happens? -- 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 1

monitor uac registration status
by Davide Zanichelli 05 Mar '21

05 Mar '21

Hello is there a way to monitor (write to db) uacreg status ? Thanks You Davide

3 2

Kamailio and Asterisk in K8s
by H Yavari 05 Mar '21

05 Mar '21

Hi to all, I am going to run multiple Asterisk instances (Pods) in my Kubernetes cluster. I have a stateless Kamailio sip proxy also. We know that K8s can't handle port range for its services, So we have to use host network for Asterisk deployment. But in this mode, Asterisk pods can't connect to other pods through local IPs and you have to add a second network interface to your pods. One solution is using something like Multus that provides multi-home pods. But it has some issue in host-mode! Also, Using media proxy like RTPEngine in front of Asterisk instances doesn't make a change. You have to run RTPEngine in the host mode or just run it on a VM with multiple NIC.What is the practical solution for this scenario? Thank you. BRHossein

3 4

Re: [SR-Users] with 302 response dispatcher list .
by Devang Dhandhalya 05 Mar '21

05 Mar '21

On Thu, Mar 4, 2021 at 6:22 PM Devang Dhandhalya < devang.dhandhalya(a)ecosmob.com> wrote: > Thank you sir for giving this much method . > > but in my case i want to take Destination address from dispatcher.list > file only . and from there when every call coming i need to send response > 302 and in that contact header i need to add those destination by group id > . so any XAVP variable use in my case or any other way to solve this issue > ? > > Thanks in Advance > Devang > > On Thu, Mar 4, 2021 at 6:09 PM Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> then maybe you can keep the list of addresses in database and use >> sql_query() (from sqlops) to retrieve them and build the headers using >> $dbr(...). >> >> Or if you want caching, see the htable module. >> >> If you do not want to use a database system, see db_text to use with >> htable or just read file content with corex module. >> >> Further flexibility can be added by using other embedded scripting >> languages, e.g., app_lua. >> >> Cheers, >> Daniel >> On 04.03.21 12:31, Devang Dhandhalya wrote: >> >> Hello >> >> Right now i am not using dispatcher for routing purpose . but i add >> destination in dispatcher.list and i need to set all those list of >> destination addresses grouped by id in 302 response in contact header >> field . >> . >> Thanks in advanced >> Devang >> >> On Thu, Mar 4, 2021 at 4:43 PM Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> do you need dispatcher for providing something else? Or just to keep the >>> list of addresses grouped by id? There can be other variants if dispatcher >>> is not needed for something else. >>> >>> Cheers, >>> Daniel >>> On 04.03.21 09:19, Devang Dhandhalya wrote: >>> >>> my scenario is : i have dispatcher list >>> >>> # dispatcher destination sets (groups) >>> # >>> >>> # line format >>> # setid(int) destination(sip uri) flags(int,opt) priority(int,opt) >>> attributes(str,opt) >>> >>> # proxies >>> 2 sip:172.19.56.21:5060 0 1 >>> 2 sip:172.19.56.22:5060 0 2 >>> >>> # gateways >>> 1 sip:192.168.0.11:5060 0 1 >>> 1 sip:192.168.0.22:5060 0 2 >>> 1 sip:192.168.0.33:5060 0 3 >>> >>> when call coming from first time client to kamailio . kamailio just need >>> to send 302 response to the client with all destination ip of group id 1 >>> priority wise in its contact header field . in easy way >>> >>> client ------------------> kamailio >>> 302 33(3) 22(2) 11(1) >>> <--------------- >>> >>> and when call coming second time from client to kamailio . kamailio need >>> to send group id 2 all destination ip in 302 response in its contact header >>> field . >>> >>> client ------------------> kamailio >>> 302 22(2) 21(1) >>> <--------------- >>> >>> can any one suggest me how to do this thing from kamailio side ? use of >>> append_branch is good option but how to configure this scenario . >>> >>> Thanks in advance >>> Devang >>> >>> *Disclaimer* >>> In addition to generic Disclaimer which you have agreed on our website, >>> any views or opinions presented in this email are solely those of the >>> originator and do not necessarily represent those of the Company or its >>> sister concerns. Any liability (in negligence, contract or otherwise) >>> arising from any third party taking any action, or refraining from taking >>> any action on the basis of any of the information contained in this email >>> is hereby excluded. >>> >>> *Confidentiality* >>> This communication (including any attachment/s) is intended only for the >>> use of the addressee(s) and contains information that is PRIVILEGED AND >>> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >>> of this communication is prohibited. Please inform originator if you have >>> received it in error. >>> >>> *Caution for viruses, malware etc.* >>> This communication, including any attachments, may not be free of >>> viruses, trojans, similar or new contaminants/malware, interceptions or >>> interference, and may not be compatible with your systems. You shall carry >>> out virus/malware scanning on your own before opening any attachment to >>> this e-mail. The sender of this e-mail and Company including its sister >>> concerns shall not be liable for any damage that may incur to you as a >>> result of viruses, incompleteness of this message, a delay in receipt of >>> this message or any other computer problems. >>> >>> _______________________________________________ >>> 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 >>> >>> >> *Disclaimer* >> In addition to generic Disclaimer which you have agreed on our website, >> any views or opinions presented in this email are solely those of the >> originator and do not necessarily represent those of the Company or its >> sister concerns. Any liability (in negligence, contract or otherwise) >> arising from any third party taking any action, or refraining from taking >> any action on the basis of any of the information contained in this email >> is hereby excluded. >> >> *Confidentiality* >> This communication (including any attachment/s) is intended only for the >> use of the addressee(s) and contains information that is PRIVILEGED AND >> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >> of this communication is prohibited. Please inform originator if you have >> received it in error. >> >> *Caution for viruses, malware etc.* >> This communication, including any attachments, may not be free of >> viruses, trojans, similar or new contaminants/malware, interceptions or >> interference, and may not be compatible with your systems. You shall carry >> out virus/malware scanning on your own before opening any attachment to >> this e-mail. The sender of this e-mail and Company including its sister >> concerns shall not be liable for any damage that may incur to you as a >> result of viruses, incompleteness of this message, a delay in receipt of >> this message or any other computer problems. >> >> -- >> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Funding: https://www.paypal.me/dcmierla >> >> -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 2

RTPEngine control protocol via TCP
by Alex Balashov 04 Mar '21

04 Mar '21

Hi, Recent versions of RTPEngine (9.x+) implemented support for speaking the "NG" wire protocol over TCP, via the `--listen-tcp-ng` option. Does the Kamailio `rtpengine` module support this, and/or are there any plans to? We've run into a few scenarios where very large responses from RTPEngine can't make it back to Kamailio over the UDP protocol due to fragmentation problems in unique network settings like tunnels. -- 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/

3 2

with 302 response dispatcher list .
by Devang Dhandhalya 04 Mar '21

04 Mar '21

my scenario is : i have dispatcher list # dispatcher destination sets (groups) # # line format # setid(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt) # proxies 2 sip:172.19.56.21:5060 0 1 2 sip:172.19.56.22:5060 0 2 # gateways 1 sip:192.168.0.11:5060 0 1 1 sip:192.168.0.22:5060 0 2 1 sip:192.168.0.33:5060 0 3 when call coming from first time client to kamailio . kamailio just need to send 302 response to the client with all destination ip of group id 1 priority wise in its contact header field . in easy way client ------------------> kamailio 302 33(3) 22(2) 11(1) <--------------- and when call coming second time from client to kamailio . kamailio need to send group id 2 all destination ip in 302 response in its contact header field . client ------------------> kamailio 302 22(2) 21(1) <--------------- can any one suggest me how to do this thing from kamailio side ? use of append_branch is good option but how to configure this scenario . Thanks in advance Devang -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 3

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sr-users