- sr-users - kamailio.org

Issues with TOPOS when not relaying an in-dialog request
by Marrold 05 Aug '21

05 Aug '21

Hi, I am currently using TOPOS to make Kamailio behave more like a B2BUA from the clients perspective. It's working well, however I have discovered a scenario where it fails. To aid with some interoperability I am sending (sl_send_reply) 200 OK to an in-dialog SUBSCRIBE request during a call. Once this has happened, the subsequent BYE from the B leg goes to the incorrect (Private) IP. If I disable this SUBSCRIBE, or relay it to the B leg, the BYE goes to the expected IP address. Does anyone know where I am going wrong? I have tried not calling record_route() for these messages but the result is the same. I've attached the ladder diagram of the call below. Note the subscribe is not relayed in this scenario. Thanks! [image: image.png]

4 8

Re: [SR-Users] How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 04 Aug '21

04 Aug '21

> Serdar, > > Have you tried to clean variables before calling new ds_select_domain(), > that are using by dispatcher module failover? > > Like > https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.… <https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…> > and so on? > > But as I got, you're saying, that calling ds_select_domain() with > different setid's in a case of fail, not really fails, but using "old" > available destinations from previous attempt? > > Regards, > Igor Igor, also thanks for your interest. > Hello, > > delete the xavps based on the names you set via modparams xavp_dst and > xavp_ctx. > > Cheers, > Daniel I remove xavp value using "xavp_rm("_dsdst_")", i supposed that my problem was solved but in my other tests, i recognised that xavp_rm removed the first index of "xavp(_dsdst_)" not all previous destinations(indexes). after researching, i found a post at https://lists.kamailio.org/pipermail/sr-users/2020-May/109192.html and i removed all ellements of list as below, if(defined $xavp(_dsdst_)) { while($xavp(_dsdst_[0]) != $null) { xlog("L_INFO", "--- Loaded Dispatchers --- Grp : $xavp(_dsdst_[0]=>grp)\n"); xlog("L_INFO", "--- Loaded Dispatchers --- Uri : $xavp(_dsdst_[0]=>uri)\n"); $xavp(_dsdst_[0]) = $null; } } it worked but i am not sure it is a best solution. Is there another simple way to delete all indexes of xavp? Best regards, Serdar

1 0

Converting Cisco SIP NOTIFY DTMF to SIP INFO (Binary operations)
by Marrold 04 Aug '21

04 Aug '21

Hi, I am using an old Cisco 2811 with an FXS card and trying to get it to convert between POTS and SIP. Mostly for "fun". For the most part it works, however there are some problems with DTMF in the call. Because the end device uses pulse dialling it seems that only the Cisco Proprietary SIP NOTIFY method of sending DTMF is supported. It sends 4 bytes in the message body that indicate the DTMF digit sent. I'd like to convert it to the more standard SIP INFO method. The docs about the encoding are here - https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/1… I looked at the transformations but couldn't find anything relevant. Is it possible to extract the binary from the body and decode it? Thanks Matthew

1 0

which is better for B2BUA
by Mojtaba 04 Aug '21

04 Aug '21

Hello there, According to , https://lists.kamailio.org/pipermail/sr-users/2016-March/092058.html, which talked about B2BUA (just signalling) in Kamailio. As i have experienced working with SEMS, freeswitch and Kamailio while using B2BUA feature, Each of them have pros and cons: 1- The sems is a light sip engine server with several applications (like as sbc) for using b2bua. All incoming and outgoing calls could go to sems server for doing b2bua like this: Incoming<=======>Kamailio<========>Sems<========>Kamailio<=======>outgoing 2- In sems, you could disable rtp realying. It forces sems to work just as b2bua without anchoring RTP 3- Easy to use different active profiles in routing. Just a couple of things there are in SEMS. For example, the sems adds itself (local IP) in Via header, and it couldn't be common in b2bua. like this: . . . INVITE sip:200@cloud.domain.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bK2f53.07a3fd9edaa8c8d609ab2ac6b01a087f.0 *================================>Kamailio public IP* *Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKsPRMNast;rport=5080 ===============================>private loopback sems ip* From: <sip:100@cloud.domain.com ;transport=UDP>;tag=6FFDB493-60EABB3600016ECD-379F9700 To: <sip:200@cloud.domain.com;transport=UDP> CSeq: 10 INVITE Call-ID: Y2M1ODQxNTZmMjdkZWZjN2U5MmMyYjBmN2Y2OGY1ODQ._leg2 Route: <sip:mo@127.0.0.1:5060;lr> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 3.3.25608 r25552 Content-Type: application/sdp Contact: <sip:127.0.0.1:5080;transport=udp> Content-Length: 246 . . . For this reason, Is there a way to avoid this issue? I know it is possible to do this by using other modules like textops and retransformation ? And why in Kamailio, there is no b2bua module to perform all b2bua functionality, yet? Thanks with Best Regards. -- --Mojtaba Esfandiari.S

6 9

How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 03 Aug '21

03 Aug '21

Hello all, I have a trouble that is related to the way of usage of dispatcher module. I am working with Kamailio 5.3.2 and using dispatcher module as load balancer to route calls to the media gateways. My dispatching routes are below as simply, route[DISPATCH] { if(ds_select_domain("2", "4")) { route(MYRELAY); } else { send_reply("503","Service Unavailable - No MGW"); exit; } } route[MYRELAY] { t_on_failure("MYFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if ($T_reply_code == 408 || $T_reply_code == 503) { if(ds_next_domain()) { route(MYRELAY); } else { send_reply("503","Service Unavailable"); exit; } } else { if(ds_select_domain("6", "4")) { route(MYVMRELAY); } else { send_reply(486,"Busy"); exit; } } } route[MYVMRELAY] { t_on_failure("MYVMFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYVMFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if(ds_next_domain()) { route(MYVMRELAY); } else { send_reply("503","Service Unavailable"); exit; } } I have multiple media gateways(setid=2) and voicemail servers(setid=6). As can be seen from the configuration, after first routing, for transaction reply code except 408 or 503, I am routing call to the voicemail server using dispatcher. In that phase, dispatcher module is remembering previously loaded destinations that come from first ds_select_domain and module try to route these destinations in the case that all voicemail servers are unavailable. So basically, I want to unload all destinations comes from previous ds_select_domain. From the documentation, i tried ds_load_update and ds_load_unset methods but not worked for me. Is there any way to overcome that problem or any right usages of these methods. Thank you, Serdar

2 2

Re: [SR-Users] How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 02 Aug '21

02 Aug '21

Hello Daniel, I removed xavps using with xavp_rm and problem solved. Thanks for your helping Serdar > Hello, > > delete the xavps based on the names you set via modparams xavp_dst and > xavp_ctx. > > Cheers, > Daniel > > On 30.07.21 11:14, Serdar GÜÇLÜER wrote: > >/Hello all, />//>/I have a trouble that is related to the way of usage of dispatcher > module. />//>/I am working with Kamailio 5.3.2 and using dispatcher module as load />/balancer to route calls to the media gateways. />//>/My dispatching routes are below as simply, />//>/route[DISPATCH] { />/ if(ds_select_domain("2", "4")) { />/ route(MYRELAY); />/ } else { />/ send_reply("503","Service Unavailable - No MGW"); />/ exit; />/ } />/} />/route[MYRELAY] { />/ t_on_failure("MYFAILURE"); />/ if(!t_relay()) { />/ sl_reply_error(); />/ } />/ exit; />/} />/failure_route[MYFAILURE] { />/ route(NATMANAGE); />/ revert_uri(); />/ if (t_is_canceled()) { />/ exit; />/ } />/ if ($T_reply_code == 408 || $T_reply_code == 503) { />/ if(ds_next_domain()) { />/ route(MYRELAY); />/ } else { />/ send_reply("503","Service Unavailable"); />/ exit; />/ } />/ } else { />/ if(ds_select_domain("6", "4")) { />/ route(MYVMRELAY); />/ } else { />/ send_reply(486,"Busy"); />/ exit; />/ } />/ } />/} />/route[MYVMRELAY] { />/ t_on_failure("MYVMFAILURE"); />/ if(!t_relay()) { />/ sl_reply_error(); />/ } />/ exit; />/} />/failure_route[MYVMFAILURE] { />/ route(NATMANAGE); />/ revert_uri(); />/ if (t_is_canceled()) { />/ exit; />/ } />/ if(ds_next_domain()) { />/ route(MYVMRELAY); />/ } else { />/ send_reply("503","Service Unavailable"); />/ exit; />/ } />/} />//>//>/I have multiple media gateways(setid=2) and voicemail />/servers(setid=6). As can be seen from the configuration, />/after first routing, for transaction reply code except 408 or 503, I />/am routing call to the voicemail server using dispatcher. />/In that phase, dispatcher module is remembering previously loaded />/destinations that come from first ds_select_domain />/and module try to route these destinations in the case that all />/voicemail servers are unavailable. />/So basically, I want to unload all destinations comes from previous />/ds_select_domain. />//>/From the documentation, i tried ds_load_update and ds_load_unset />/methods but not worked for me. />//>/Is there any way to overcome that problem or any right usages of these />/methods. />//>/Thank you, />//>/Serdar />//>//>/__________________________________________________________ />/Kamailio - Users Mailing List - Non Commercial Discussions />/* sr-users at lists.kamailio.org > <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> />/Important: keep the mailing list in the recipients, do not reply only > to the sender! />/Edit mailing list options or unsubscribe: />/* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> / > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 1

Access via script to location data in the memory
by Social Boh 31 Jul '21

31 Jul '21

Hello List, is there a way to access location data from kamailio script? I mean data I can see with the command: kamctl ul show Regards -- --- I'm SoCIaL, MayBe

3 11

Guidelines on E164 conversions
by Melissa Frasik 30 Jul '21

30 Jul '21

Hi all, I'm running into some issues with E164 conversions and was wondering if there was a decent standardized method for conversions. Conversions using $rU seem to be working great, but $fU is behaving badly. This is what I have so far: xlog("$ci - caller id was $avp(src=>callerid)"); if($avp(src=>callerid) =~ "^\+"){ xlog("$ci - caller id already E164"); }else if($(rU{s.substr,0,$(avp(src=>countrycode){s.len})}) == $avp(src=>countrycode)){ $avp(src=>callerid) = "+" + $avp(src=>callerid); $fU = $avp(src=>callerid); $avp(uacreplacefromdisplay) = $fU; $avp(uacreplacefromuri) = $fu; xlog("$ci - Adding plus to make caller id e164"); }else{ $fU = "+" + $avp(src=>countrycode) + $fU; $avp(uacreplacefromdisplay) = $fU; $avp(uacreplacefromuri) = $fu; xlog("$ci - converting domestic caller id number to $fU ($fu) - $avp(uacreplacefromdisplay) $avp(uacreplacefromuri)"); } Previously, I'd found a suggestion mentioning using uac_replace_from in the route[RELAY] (which is the purpose of the uacreplacefrom avps), but $fU does not appear to be changing its value. Is there something I'm doing incorrectly?

2 4

How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 30 Jul '21

30 Jul '21

Hello all, I have a trouble that is related to the way of usage of dispatcher module. I am working with Kamailio 5.3.2 and using dispatcher module as load balancer to route calls to the media gateways. My dispatching routes are below as simply, route[DISPATCH] { if(ds_select_domain("2", "4")) { route(MYRELAY); } else { send_reply("503","Service Unavailable - No MGW"); exit; } } route[MYRELAY] { t_on_failure("MYFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if ($T_reply_code == 408 || $T_reply_code == 503) { if(ds_next_domain()) { route(MYRELAY); } else { send_reply("503","Service Unavailable"); exit; } } else { if(ds_select_domain("6", "4")) { route(MYVMRELAY); } else { send_reply(486,"Busy"); exit; } } } route[MYVMRELAY] { t_on_failure("MYVMFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYVMFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if(ds_next_domain()) { route(MYVMRELAY); } else { send_reply("503","Service Unavailable"); exit; } } I have multiple media gateways(setid=2) and voicemail servers(setid=6). As can be seen from the configuration, after first routing, for transaction reply code except 408 or 503, I am routing call to the voicemail server using dispatcher. In that phase, dispatcher module is remembering previously loaded destinations that come from first ds_select_domain and module try to route these destinations in the case that all voicemail servers are unavailable. So basically, I want to unload all destinations comes from previous ds_select_domain. From the documentation, i tried ds_load_update and ds_load_unset methods but not worked for me. Is there any way to overcome that problem or any right usages of these methods. Thank you, Serdar

2 1

Setting up uacreg
by Duncan Turnbull 30 Jul '21

30 Jul '21

Hi there I am a new user of Kamailio and we are trying to use it to be as a front end for our asterisk pbx. We are running on Ubuntu 18.04 and Kamailio 5.3.8 with Siremis Rather than proxying the request through to asterisk we are trying to use uacreg to send a login to asterisk. Asterisk will think all the users are appear from the proxy but thats okay. Initially this is just for external users but eventually all phones etc will register via Kamailio and we will have the trunks there (and split them across another kamailio but thats another job) If I add a user to the uacreg then when I register to Kamailio it sends a register request but to the realm in the uacreg table and the matching port Kamailio is running on. Is this because somewhere we have set Kamailio to directly proxy on and we need to turn that off first? This is our uacreg table mysql> select * from uacreg; +----+--------+------------+------------+------------+-----------+-----------+---------------+---------------+----------+--------------------+---------+-------+-----------+--------+ | id | l_uuid | l_username | l_domain | r_username | r_domain | realm | auth_username | auth_password | auth_ha1 | auth_proxy | expires | flags | reg_delay | socket | +----+--------+------------+------------+------------+-----------+-----------+---------------+---------------+----------+--------------------+---------+-------+-----------+--------+ | 1 | testuser | testuser | ourdomain.com | 88 | 10.8.8.20 | 10.8.8.20 | 88 | password | '' | sip:10.8.8.20:5060 | 360 | 0 | 3 | | +----+--------+------------+------------+------------+-----------+-----------+---------------+---------------+----------+--------------------+---------+-------+-----------+--------+ 1 row in set (0.00 sec) All pointer, guides and recommendations will be welcome Thanks very much Cheers Duncan

3 8

Guidance tracking down "qm_free(): BUG: freeing already freed pointer"
by Anthony Joseph Messina 29 Jul '21

29 Jul '21

I'm seeking guidance on how to track down "qm_free(): BUG: freeing already freed pointer" which occurs only on branched calls. These errors don't crash so I don't get any core dumps. The different log entries below are the result of me selectively disabling sections of the script that apply in branch route, all to no avail. I'm running Kamailio on the current tip of the 5.5 branch (1f9f6fff6e). I'm reviewing https://www.kamailio.org/wiki/tutorials/troubleshooting/memory in the meantime as a place to start. version: kamailio 5.5.1-5.git1f9f6fff6e.fc34 (x86_64/linux) 7abebb 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED 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. CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f04b8f47e90), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f04b8f06c70), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f08e8f9cf80), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afafa60d8), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/sdp/sdp.c: init_p_payload_attr(183) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afae2a018), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afae431e8), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afafa3900), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f5fd4044ef0), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7ffbb2d2e2f0), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f6d1a50b1d8), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc19c165c70), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc19c17adb8), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc19c125790), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4d969c9d48), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc29bdca3d0), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc2c2eccc80), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc2c2ec52c0), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f910c04e240), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa351286b18), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa3512a7ae8), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa0777d9e70), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa0777d7a80), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4f2c92cf20), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4f2c8afa00), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4f2c92aa48), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fe55825eb30), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/sdp/sdp.c: init_p_payload_attr(183) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f6526db8378), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fcd9b465980), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7ff7ca92ff60), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f978a2dc938), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f978a2b7eb8), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa1556c1610), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fd81dfbba78), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fb2750d9988), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fb2750e1b28), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f32f7b45d30), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f35849e1940), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fd3a2c04280), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f5de857a040), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring -- Anthony - https://messinet.com CCED 8BFB 88D9 18F6 9F68 93DC 45C4 A8CE A64E 88E3

2 6

BYE not accepted by MS TEAMS
by João Vitor Arruda 29 Jul '21

29 Jul '21

Hello all, I recently built an SBC to integrate Kamailio with MS Teams following Henning Westerholt blob post and all the other awesome tips I was able to find here in the list. Everything is working great. The only missing piece I would like some thoughts on is about why MS Teams is not accepting the BYE sent from Kamailio. It eventually replies with "504 Server Time-out" The below trace shows the entire call. |||||||||||||||||||| ==================== tag: rcv pid: 3381 process: 63 time: 1627070235.086307 date: Fri Jul 23 19:57:15 2021 proto: tls ipv4 srcip: 52.114.76.76 srcport: 4672 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ INVITE sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone;transport=tls SIP/2.0 FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 TO: <sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bK60895c91 RECORD-ROUTE: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> CONTACT: <sip:api-du-b-usea2.pstnhub.microsoft.com:443 ;x-i=a82f1d71-4e21-42a8-8389-2ae26e54610f;x-c=c9b2aa4f658f5788aa7090bef0f5e35e/d/8/b14037a741dd46a79c74dce15912782c> CONTENT-LENGTH: 1103 MIN-SE: 300 SUPPORTED: timer USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.EUNO.0 CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY PRIVACY: id SESSION-EXPIRES: 3600 v=0 o=- 599625 0 IN IP4 127.0.0.1 s=session c=IN IP4 52.113.58.181 b=CT:10000000 t=0 0 m=audio 50122 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118 c=IN IP4 52.113.58.181 a=rtcp:50123 a=ice-ufrag:fTy/ a=ice-pwd:tQKGN0Szi7Q/vTdf6SiXUSpj a=rtcp-mux a=candidate:1 1 UDP 2130706431 52.113.58.181 50122 typ srflx raddr 10.0.33.230 rport 50122 a=candidate:1 2 UDP 2130705918 52.113.58.181 50123 typ srflx raddr 10.0.33.230 rport 50123 a=candidate:2 1 tcp-act 2121006078 52.113.58.181 49152 typ srflx raddr 10.0.33.230 rport 49152 a=candidate:2 2 tcp-act 2121006078 52.113.58.181 49152 typ srflx raddr 10.0.33.230 rport 49152 a=label:main-audio a=mid:1 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:PC00Mgk39GjTYJ+mUSNQg++TU6zJKar4VYDj4/ET|2^31 a=sendrecv a=rtpmap:104 SILK/16000 a=rtpmap:9 G722/8000 a=rtpmap:103 SILK/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=ptime:20 |||||||||||||||||||| ==================== tag: snd pid: 3381 process: 63 time: 1627070235.089106 date: Fri Jul 23 19:57:15 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 100 trying -- your call is important to us FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 TO: <sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bK60895c91;rport=4672 Server: kamailio (5.3.7 (x86_64/linux)) Content-Length: 0 |||||||||||||||||||| ==================== tag: snd pid: 3291 process: 18 time: 1627070235.111825 date: Fri Jul 23 19:57:15 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 t:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e CSEQ: 1 INVITE m:<sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp> User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Accept:application/sdp Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces u:talk,hold,conference,refer l:0 Via: SIP/2.0/TLS 52.114.76.76:5061;rport=4672;branch=z9hG4bK60895c91 Record-Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on>,<sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XBranch: z9hG4bKe7d4.5ae277881edb9c04a7ac65af016e582e.0 |||||||||||||||||||| ==================== tag: snd pid: 3298 process: 21 time: 1627070245.043621 date: Fri Jul 23 19:57:25 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 183 Session Progress FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 t:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e CSEQ: 1 INVITE m:<sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp> User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces u:talk,hold,conference,refer c:application/sdp Content-Disposition:session l:323 Via: SIP/2.0/TLS 52.114.76.76:5061;rport=4672;branch=z9hG4bK60895c91 Record-Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on>,<sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XBranch: z9hG4bKe7d4.5ae277881edb9c04a7ac65af016e582e.0 v=0 o=FreeSWITCH 1627045233 1627045234 IN IP4 SBC_IP_ADDR s=FreeSWITCH c=IN IP4 SBC_IP_ADDR t=0 0 m=audio 49692 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eMGqj0jubamtvLsI/E/YSj8UqNKgUB5DTHGWc7PY a=nortpproxy:yes |||||||||||||||||||| ==================== tag: snd pid: 3293 process: 19 time: 1627070255.938770 date: Fri Jul 23 19:57:35 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 200 OK FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 t:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e CSEQ: 1 INVITE m:<sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp> User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces u:talk,hold,conference,refer c:application/sdp Content-Disposition:session l:323 Via: SIP/2.0/TLS 52.114.76.76:5061;rport=4672;branch=z9hG4bK60895c91 Record-Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on>,<sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XBranch: z9hG4bKe7d4.5ae277881edb9c04a7ac65af016e582e.0 v=0 o=FreeSWITCH 1627045233 1627045234 IN IP4 SBC_IP_ADDR s=FreeSWITCH c=IN IP4 SBC_IP_ADDR t=0 0 m=audio 49692 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eMGqj0jubamtvLsI/E/YSj8UqNKgUB5DTHGWc7PY a=nortpproxy:yes |||||||||||||||||||| ==================== tag: rcv pid: 3370 process: 57 time: 1627070255.957735 date: Fri Jul 23 19:57:35 2021 proto: tls ipv4 srcip: 52.114.76.76 srcport: 4672 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ ACK sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp SIP/2.0 FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 TO: <sip:+5515XXXXXXXXX@SBC-FQDN:5061>;user=phone;tag=e55N8paBUgZac CSEQ: 1 ACK CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKe3747d68 CONTACT: <sip:api-du-b-usea2.pstnhub.microsoft.com:443 ;x-i=a82f1d71-4e21-42a8-8389-2ae26e54610f;x-c=c9b2aa4f658f5788aa7090bef0f5e35e/d/8/b14037a741dd46a79c74dce15912782c> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.EUNO.0 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY Route: <sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> P-SR-XUID: atpsh-60fb1b5d-d35-1 |||||||||||||||||||| ==================== tag: snd pid: 3302 process: 23 time: 1627070275.228363 date: Fri Jul 23 19:57:55 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.75.24 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ BYE sip:52.114.76.76:4672;x-i=a82f1d71-4e21-42a8-8389-2ae26e54610f;x-c=c9b2aa4f658f5788aa7090bef0f5e35e/d/8/b14037a741dd46a79c74dce15912782c SIP/2.0 Record-Route: <sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=e55N8paBUgZac;lr=on> Record-Route: <sip:192.168.1.253:5060;r2=on;ftag=e55N8paBUgZac;lr=on> Via: SIP/2.0/TLS SBC_IP_ADDR:5061;branch=z9hG4bK92fc.fcb147cf8e5e6e6e960d0f2401ad6383.0 v:SIP/2.0/UDP 192.168.1.254;received=192.168.1.254;rport=5060;branch=z9hG4bKgvXrQrcNjNj3m f:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac t:"Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 i:c9b2aa4f658f5788aa7090bef0f5e35e CSeq:38981345 BYE User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces Reason:Q.850;cause=16;text="NORMAL_CLEARING" Max-Forwards: 64 l:0 Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XUID: atpsh-60fb1b5d-d35-1 |||||||||||||||||||| ==================== tag: rcv pid: 3385 process: 65 time: 1627070300.423499 date: Fri Jul 23 19:58:20 2021 proto: tls ipv4 srcip: 52.114.75.24 srcport: 5061 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 504 Server Time-out FROM: <sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac TO: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 CSEQ: 38981345 BYE CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e VIA: SIP/2.0/TLS SBC_IP_ADDR:5061;branch=z9hG4bK92fc.fcb147cf8e5e6e6e960d0f2401ad6383.0 CONTENT-LENGTH: 0 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY SERVER: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.EUWE.3 Via: SIP/2.0/UDP 192.168.1.254;received=192.168.1.254;rport=5060;branch=z9hG4bKgvXrQrcNjNj3m Record-Route: <sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=e55N8paBUgZac;lr=on>,<sip:192.168.1.253:5060 ;r2=on;ftag=e55N8paBUgZac;lr=on> P-SR-XBranch: z9hG4bK92fc.fcb147cf8e5e6e6e960d0f2401ad6383.0

2 2

Accessing Transformations in KEMI
by Ben Kaufman 28 Jul '21

28 Jul '21

Sorry to bother on this, but is it possible to access Kamailio transformations in KEMI, and if so, how? In Kamailio's native scripting language I could do: $var(pai_user) = $(hdr(P-Asserted-Identity){tobody.user}); In KEMI, I can easily get the full header like this: KSR.hdr.get("P-Asserted-Identity") But I don't see how to access the transformations. Am I missing it? Ben Kaufman

3 6

Password for pi_framework.xml - Xhttp_pi module
by Abdirahman Osman 27 Jul '21

27 Jul '21

Hello everyone, In the xhttp_pi file of pi_framework.xml,is there a way I can pass the password to the xhttp_pi module as a pre-processing define? <db_url id="postgres">postgres://kamailiorw:kamailiorw@localhost /kamailiorw</db_url> Thank you! Abdirahman

2 1

Info: Kamailio World Online - Sep 1-2, 2021 - Celebrating 20 Years Of Development
by Daniel-Constantin Mierla 26 Jul '21

26 Jul '21

Hello, given the current situation across the world, no physical Kamailio World Conference is planned in 2021. However, this year the project celebrates 20 years of development -- according to commit history, the first one was done on the 3rd of September, 2021: > ❯ git log --reverse > commit 512dcd9838f5aeb0bfccd02cfc156067b7f02b71 > Author: Andrei Pelinescu-Onciul <andrei(a)iptel.org> > Date: Mon Sep 3 21:27:11 2001 +0000 > Initial revision To mark this notable milestone, we are going to organize an online event during Sep 1-2, 2021, similar to the one from 2020 -- the website for the event is available at: * https://www.kamailioworld.com/k09-online/ Mark the date in your calendar, more details about the content of the event will be published in the near future! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

New user checking on SIP proxying use case
by Nick Hurlburt 22 Jul '21

22 Jul '21

Hello all, My name is Nick Hurlburt, and work with a nonprofit tech organization that provides a software platform for child helplines around the world. I am a new user to Kamailio, and I'm interested in figuring out if my use case is appropriate, getting some tips on how to go about it, and possibly enlisting some expert help. Our service is built on top of Twilio, but we are working in some countries where Twilio doesn't have phone numbers. So in those areas, we are working with local telecoms on ways to integrate using a SIP connection. In one particular example, the telecom company can send SIP INVITEs but can only address them to a static IP. However, Twilio needs to receive the invite addressed to a fully-qualified domain name (eg, example-setup.sip.twilio.com). The idea arose to use Kamailio as a proxy to accept the static IP-addressed request and then rewrite the request to pass it along to a domain name. Then once the connection is established between the two endpoints, the media can flow directly between them. I have a software engineering background but still coming up to speed when it comes to VoIP technology. I'd be interested in: - does this sound like a good use case for Kamailio? - is there any specific documentation you could point me to for this use case? (I have been looking at the Kamailio documentation, but a quick pointer could save a lot of time in digging through it) - if anyone from the community would be open to helping us develop a prototype/proof-of-concept instance of this, especially if you're open to contributing to a nonprofit on a volunteer or discounted rate, I'd love to talk more Thank you, Nick

4 4

Kamailio/rtpengine not populating mos avp stats?
by Anthony Joseph Messina 22 Jul '21

22 Jul '21

I'm using Kamailio compiled from the current 5.5 tip (1f9f6fff6e), and rtpengine compiled from the current master (6e160da4). While rtpengine is working quite well, within the past month or so, I've noticed that Kamailio isn't populating the mos* related AVPs. I see continued upstream rtpengine work stream handling, etc. -- should I wait or try to debug? Thanks -A -- Anthony - https://messinet.com CCED 8BFB 88D9 18F6 9F68 93DC 45C4 A8CE A64E 88E3

3 6

How to get Stir/Shaken certificates
by ryan embgrets 22 Jul '21

22 Jul '21

Greetings. Does someone know where one can get stir/shaken certificates? Are they same as letsencrypt or do I need to obtain them from some other authority. If someone could provide me with a link to that authority it will be very helpful. Thanks again for your help and time. Ryan

4 5

Question about MS Teams integration
by Володимир Іванець 22 Jul '21

22 Jul '21

Hello all! Sorry, this is not a direct Kamailio question. I would like to ask for some help from those who already did this. I'm trying to integrate my testKamailio installation with MS Teams. I added the "mydomain.com" to the Domains section (I kept the " mydomain.onmicrosoft.com"). When I tried to create SBC with a " kamailio-dev2.mydomain.com" FQDN I got this error message: We* can't use the "kamailio-dev2.mydomain.com <http://kamailio-dev2.mydomain.com>" domain as it hasn't be set up in the organization. Try again. If you continue to have problems, contact Microsoft customer support.* There were no issues creating a dummy SBC with a " kamailio-dev2.mydomain.onmicrosoft.com" FQDN. Does anyone know how to set up a domain in the organization from MS perspective? Thank you very much! Regards, Volodymyr Ivanets.

4 9

serial forking kamailio - twilio
by Filippo Graziola 21 Jul '21

21 Jul '21

Hi all, I have a question for my kamailio configuration which works like this: - INVITE arrives to kamailio - HTTP async request to API endpoint - Resume, use of rtjson module for parallel forking and loop to kamailio itself - One of the invite is "transformed" in serial forking with rtjson module and it is sent with a delay to TWILIO SIP Domain (landline/mobile phone numbers) Twilio is giving me some troubles because it answers with a 200 OK but the call it is still in establishing/ringing on the "landline" side and as expected when reaching the timeout (no answer from phone number on TWILIO side) kamailio is receiving a BYE. Now the question, how to handle next number to call through Twilio? I was thinking of using rtjson_next_route() when handling the in-dialog BYE but as it is not a failure route so it does not work, or I'm doing it wrong. Thanks in advance for any help/hint. Filippo

2 2

Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record
by Henning Westerholt 20 Jul '21

20 Jul '21

Hello, I have not tested it, but the dialog database will probably only help in a restart scenario with regarding to dialogs, at least with a standard cfg. Regarding the example below, this would mean starting the primary Kamailio again, then the cdrs should be generated during the hangup from the primary. You can test also to restart the secondary Kamailio, it should then also read the dialogs from the replicated table and generate cdrs at hangup. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Social Boh <social(a)bohboh.info> Sent: Tuesday, July 20, 2021 2:02 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record Hello, with dialog db_mode 1 and Primary/Primary MariaDB Replication If I switch Off primary Kamailio during a call and move the call to Replica and then hangup, no entry on acc_cdrs table. acc table is ok. Regards --- I'm SoCIaL, MayBe El 20/07/2021 a las 4:09 a. m., Henning Westerholt escribió: Hello, Usage of DB to store dialogs. See https://kamailio.org/docs/modules/devel/modules/dialog.html#dialog.p.db_mode – in particular mode 1 and 3. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org><mailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Social Boh Sent: Tuesday, July 20, 2021 12:55 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org><mailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record Hello what means "DB loading" ? Regards --- I'm SoCIaL, MayBe El 19/07/2021 a las 9:25 a. m., Henning Westerholt escribió: Hello, so, you are saying that in a DMQ replicated case the acc CDR generation is not working in failover cases. I just looked briefly to the code; others might have more details. But I think this is because the acc module do not register to a callback function for dialog loading on startup for the DMQ replication. This is done for DB loading, but seems to be not implemented yet for DMQ synchronisation. You could open a feature request on our github tracker for this. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org><mailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Social Boh Sent: Monday, July 19, 2021 1:58 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org><mailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record Hi, I'm using 5.4.6 I switch off Kamailio on the first server and keepalived move the virtual IP on the second server so First Kamailio is off. This during the call. Before switch off Kamailio and during the call I can see data dialog on both server but in the LOG /var/log/messages in my case: server 1 INVITE server 2 BYE acc_cdrs table empty on both servers. acc table first server INVITE; acc table second server BYE Regards --- I'm SoCIaL, MayBe El 18/07/2021 a las 2:33 p. m., M Arqum CH escribió: Hi SoCIal, which Kamailio version you are using. did you able to get the dialog information using the dialog command. https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_l… when you received bye on secondary server, is that server recognized the bye. On Sun, Jul 18, 2021 at 3:38 AM Social Boh <social(a)bohboh.info<mailto:social@bohboh.info>> wrote: Hello, if during a call I stop Primary Kamailio the Secondary Kamailio take de Virtual IP but don't save call data: In the Primary database, acc table, I have INVITE data but not BYE data; no data in acc_cdrs for the call. In the Secondary Kamailio, acc table, I have BYE data but not INVITE data. in the file LOG no data about call cdr. Maybe a solution is write call data whene the call end? Regards --- I'm SoCIaL, MayBe El 17/07/2021 a las 11:34 a. m., M Arqum CH escribió: Make sure that your table columns matched the field's you are trying to insert, please check the syslog for acc module log, acc module print the cdr in log facility too, if acc module facing any issue while writing cdr in db, it will throw error in logs. Hope this will help you. On Sat, Jul 17, 2021, 7:33 PM Social Boh <social(a)bohboh.info<mailto:social@bohboh.info>> wrote: Hello, i tested with DMQ module and happens the same. DMQ module loaded before dialog module and on dialog module parameters: modparam("dialog", "enable_dmq", 1) During a call I can see on both Kamailio servers the dialog data but when switch off the active kamailio server and the second kamailio server take the control and hangup the call, no data acc_cdrs table. Regards --- I'm SoCIaL, MayBe El 17/07/2021 a las 1:24 a. m., M Arqum CH escribió: Hi Social, Replication of dialog using dmq will solve your problem, you need to load dmq module before loading dialog module and enable replication of dialogs in dialog module, please check dialog module conf option for more details. Cheers On Fri, Jul 16, 2021, 7:32 PM Social Boh <social(a)bohboh.info<mailto:social@bohboh.info>> wrote: Im' testinf Kamailio HA with Keepalived. I have a problem with acc_cdrs table in this situation: A call B and when the two users are in the call I switch off the active Kamailio. Keepalived pass the Virtual IP to the secondary Kamailio and audio still work because I'm using a remote RTPENGINE server for both Kamailio. The problem is when the call end. On the acc table i can see INVITE and BYE entries but on the acc_cdrs table nothing. Second problem the dialog still present on the table dialog in a Primary/Primary replication. Is there a way to solve it? Thank you Regards -- --- I'm SoCIaL, MayBe __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

Kamailio KeepAlived HA acc_cdrs table record
by Social Boh 20 Jul '21

20 Jul '21

Im' testinf Kamailio HA with Keepalived. I have a problem with acc_cdrs table in this situation: A call B and when the two users are in the call I switch off the active Kamailio. Keepalived pass the Virtual IP to the secondary Kamailio and audio still work because I'm using a remote RTPENGINE server for both Kamailio. The problem is when the call end. On the acc table i can see INVITE and BYE entries but on the acc_cdrs table nothing. Second problem the dialog still present on the table dialog in a Primary/Primary replication. Is there a way to solve it? Thank you Regards -- --- I'm SoCIaL, MayBe

3 11

Kamailio python_exec getting randomly TypeError
by Voip support 19 Jul '21

19 Jul '21

Dear Kamailio users, I am trying to use a python script using the module app_python3 to do some billing calculations. The only argument i want to pass to the python script is duration which i calculate in event_route[dialog:start] { $dlg_var('call_start') = $TS; event_route[dialog:end] { $var('v_duration') = $TS - $dlg_var('call_start'); So my $var('v_duration') has the duration of the call in seconds. Then i execute from my kamailio.cfg: python_exec("calculateCost","$var('v_duration')"); The script works randomly - sometimes all is good and sometimes it throw error : app_python3 [python_support.c:156]: python_handle_exception(): apy_exec: calculateCost(4): Unhandled exception in the Python code: TypeError: an integer is required (got type NoneType) The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio-python.py", line 282, in calculateCost duration = int(duration) It looks like the kamailio log show calculateCost(4) so argument 4 is passed to python function but once it executes it appear like there is no string/integer but NoneType argument. I cannot figure this out why this randomly happens - maybe it's an app_python3 module issue? Looking to hear back from you! BR, Tom

2 1

please help to configure tls in kamailio for webrtc client like simpl5
by ThanhTruong 19 Jul '21

19 Jul '21

Dear, I am trying to make a test to see how can webrtc work in kamailio and tls. I was try some link : 1 <https://github.com/havfo/WEBRTC-to-SIP>. https://github.com/havfo/WEBRTC-to-SIP <https://github.com/havfo/WEBRTC-to-SIP> 2. https://telecom.altanai.com/2018/09/04/kamailio-webrtc-sip-server/ <https://telecom.altanai.com/2018/09/04/kamailio-webrtc-sip-server/> And Generate CA from the link: https://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates <https://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates> Then, I tried to login with simpl5 and i got the message back Disconnected: Failed to connet to the server I do not know how can i fix that issue, look like TLS configure has issue with that. the log is showing something like Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094416:SSL routines:ssl3_read_bytes:sslv3 alert certificate unknown Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 27.65.214.194 Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 172.31.44.170 Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: <core> [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f4233c60fc0 r: 0x7f4233c610e8 (-1) My tls.cfg is : [server:default] method = SSLv23 verify_certificate = yes require_certificate = yes private_key = /etc/certs/my.domain/key.pem certificate = /etc/certs/my.domain/cert.pem ca_list = /etc/certs/demoCA/cert.pem [client:default] verify_certificate = yes require_certificate = yes Please help me to register sipml5 or how can i create CA for TLS in kamailio to use with webrtc. Thank all.

4 14

Active Profile for sipreply in SEMS
by Mojtaba 17 Jul '21

17 Jul '21

Hello, Does anybody know how i could change white filter in siprelpy in sems? In Kamailio, I got "200 OK" from UE, then add a new header (SBC: mt-reply) and send it to sems server. The sems server remove it: [#7f410fbfb700/4560] [inplaceHeaderFilter, HeaderFilter.cpp:226] DEBUG: erasing header 'sbc' by whitelist I really don't want to remove it. Actually if I knew how i active profile for sip replying in sems, I would be able to solve this. Thanks with regards. -- --Mojtaba Esfandiari.S

2 2

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users