- sr-users - kamailio.org

kamailio it does not start after reboot
by Atux Atux 30 Mar '18

30 Mar '18

i am installing Kamailio using apt in a debian 8 machine by following this guide http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour i managed to create a few users and through Jitsi to talk to each other. once i reboot the machine, then the problems start and Kamailio cannot start. here are a few details from my system. -/etc/apt/sources.list the part for Kamailio #Kamailio deb http://deb.kamailio.org/kamailio51 jessie main deb-src http://deb.kamailio.org/kamailio51 jessie main -Version info Distributor ID: Debian Description: Debian GNU/Linux 8.10 (jessie) Release: 8.10 Codename: jessie -root@debian:~# systemctl start kamailio Job for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. root@debian:~# /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceJob for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. failed! Any ideas, please?

4 6

180 and 183 SIP messages
by Ali Taher 30 Mar '18

30 Mar '18

Hello everyone, I'm working on freeswitch , where I have SIP calls coming from A-Party to my freeswitch and I'm routing it to B-Party. I'm facing an issue , when I'm getting back the 183 message from the supplier(B-Party) and forwarding it to the customer(A-Party); the customer is not hearing the RBT and asking to send 180 rather than 183 . (the supplier is only sending 183 without 180). Is it possible using kamailio to replace this 183 message with 180 message , or even add 180 message while keeping the same 183 message ? Regards, Ali Taher

4 7

kamailio & vmware
by Jean Cérien 29 Mar '18

29 Mar '18

Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts. Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related. Anyone has advice on kamailio on a VM, when it only handles sip ? Rgds J.

7 10

Kamailio fails to start because of jsonrpl
by Dmitry 29 Mar '18

29 Mar '18

Kamailio fails to start and I see the following errors: kamailio[2115]: ERROR: <core> [core/pvapi.c:1085]: pv_parse_spec2(): bad tr in pvar name "jsonrpl" kamailio[2115]: ERROR: <core> [core/pvapi.c:1111]: pv_parse_spec2(): invalid parsing in [$(jsonrpl(body){kz.json, result.NRSETS})] at (4) kamailio[2115]: CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kazoo/kamailio/dispatcher-role.cfg, line 243, colu...ult.NRSETS}) I installed kamailio-json-modules, Kamailio v5.1.2 installed from DEB packets from deb.kamailio.org What is the cause of the error?

2 1

Kamailio to Asterisk Proxy with NAT
by Benjamin Marty 29 Mar '18

29 Mar '18

Hello, I have multiple Asterisk Servers in a private Network. They also have a Public IP via Destination NAT. Now I want to use a Kamailio Proxy in front of them. The Routing I want to do with the Kamailio Dispatcher module. My Question is now how the RTP Media Stream should/can flow. The clients are in different other networks. So P2P Media Stream isn't possible. Should I now run the RTP Stream Client - Asterisk or Client - Kamailio - Asterisk? Is there an Example for an such scenario? Thanks Benjamin

2 3

Missing exit function on KEMI?
by Enrico Bandiera 29 Mar '18

29 Mar '18

Hello, I'm trying to use the KSR.x.exit() function in a KEMI python script but at runtime i get this error: KSR.x.exit() AttributeError: 'module' object has no attribute 'exit' It seems that the exit function is missing even though it is documented. My kamailio version is: kamailio 5.2.0-dev4 (x86_64/linux) (it's a nightly) Any idea on what I can do? Thanks, Enrico.

2 3

Kamailio NAT AWS EC2
by Benjamin Marty 29 Mar '18

29 Mar '18

Hi, I am new to Kamailio and coming a bit from the Asterisk side. My goal is to set up a bunch of users on a Kamailio Instance on AWS EC2 and get them able to call each other and talk to each other over the Internet without P2P media stream. Is rtpproxy the tool that enables that? As far my knowledge is right Kamailio isn't able to do any sort of RTP handling. Regards, Benjamin

2 4

kamailio with media server
by Atux Atux 28 Mar '18

28 Mar '18

Hi. New to the area of Kamailio. i am did install in debian kamailio with rtpproxy and i have created 3 users 1000-10002 (one for each jitsi user) they talk nice between them. I have followed this tutorial http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour and in less than 5 minutes i had my accounts registered. i would like to have a media server so the users could hears announcements and stuff. At the end i would like to have kamailio as a test lab PBX where i could connect my SIP trunk providers and my users to route calls. Is there any guide on how to setup a media server and the services, please?

3 5

is t_flush_flags() really needed?
by Juha Heinanen 28 Mar '18

28 Mar '18

While testing xflags, i noticed that a regular flag that I set AFTER calling t_newtrans() stays set in onreply_route even when I do not call t_flush_flags(). In a route block: if (!t_newtran()) { ERROR(Failed to create new transaction for $rm <$ru>); exit; }; if (!isflagset(19)) INFO(******** flag 19 is NOT set after t_newtran()); setflag(19); and in onreply_route: if (isflagset(19)) INFO(********* flag 19 is set); else INFO(********* flag 19 is NOT set); Syslog gets: Mar 25 11:24:41 trout /usr/bin/sip-proxy[21362]: INFO: ******** flag 19 is NOT set after t_newtran() Mar 25 11:24:41 trout /usr/bin/sip-proxy[21433]: INFO: ********* flag 19 is set So what is the point of t_flush_flags()? -- Juha

2 3

Terminate dialog from another dialog
by Joel Serrano 27 Mar '18

27 Mar '18

Hi guys, I'm trying to set kamailio to not allow simultaneous calls from the same user, but if a second call comes in, the behavior desired is to process that second call and terminate the previous (already active) one. Say you have call A that creates dialog A. Then you have call B come in, creates dialog B. Can I save some info from dialog (call) A in a htable for example, then check that htable for the needed info from dialog (call) B and terminate dialog (call) A if it exists? I think using dlg_get() and dlg_bye() would do the trick but not 100% sure how they work.. The example in the docs seems exactly the case: dlg_get --> "Search and set current dialog based on Call-ID, From-Tag and To-Tag parameters" if(dlg_get("abcdef", "123", "456")) { dlg_bye("all"); } After the dlg_get() and dlg_bye() what happens to call B? Does the route continue running or does it end? Will it automatically switch back to the current dialog or do you have to manually switch back running dlg_get() again? (Sorry if I'm not explaining myself correctly, let me know if it sounds confusing) Does anyone have experience with these two functions? Thanks, Joel.

2 4

kamailio 5.0.6 try to using dialog module - dont starts dialog counting
by Karsten Horsmann 27 Mar '18

27 Mar '18

Hello, i try to get dialog-infos to see how many calls are running on my kamailio. But i can not figure out what steps are necessary to get this running. loading the module and the dependencies - check. set and dlg_flag via module param - check. MUST i use a DB?! Or what is missing? kamctl stats | grep dialog "dialog:active_dialogs = 0", "dialog:early_dialogs = 0", "dialog:expired_dialogs = 0", "dialog:failed_dialogs = 0", "dialog:processed_dialogs = 0", the dialog snippets (grep out ouf my config): #!define DLG_FLAG 8 loadmodule "dialog.so" modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "rr_param", "did") modparam("dialog", "profiles_with_value", "active_calls; inbound_calls; outbound_calls") modparam("dialog", "send_bye", 1) modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_extra_hdrs", "Hint: inactivity timeout\r\n") # Added to requests generated locally by the module (e.g. BYE) modparam("dialog", "track_cseq_updates", 1) -- Kind Regards *Karsten Horsmann*

4 5

To tags in 407 challenges
by Alex Balashov 27 Mar '18

27 Mar '18

Hi, This may be more a SIP question than a Kamailio question, but I am trying to figure out why 407 challenges from Kamailio contain To tags. I suppose I've not run across it before, and I can't find an RFC-based rationale. A negative reply to an invite transaction without an intermediate early dialog-forming reply (e.g. 183 with To tag) does not form a dialog, nor takes place within a context in which a dialog was created. So what gives? Insights appreciated! -- 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 4

Bug in evrexec module?
by Skip Morse 27 Mar '18

27 Mar '18

Hi all, anyone try the evrexec module yet? When I try, Kamailio fails to start and I'm getting an error that it can't find the event route I'm specifying in the modparam, but I started with trying the example from the docs. I think it might be a bug https://www.kamailio.org/docs/modules/5.1.x/modules/evrexec.html Here's what I tried: modparam("evrexec", "exec", "name=evrexec:timer;wait=1000;workers=1;") event_route[evrexec:timer] { $var(x) = 0; while(1) { xlog("$$var(x) is $var(x)\n"); $var(x) = $var(x) + 1; sleep("600"); } } Here's the log: 0(1765) ERROR: evrexec [evrexec_mod.c:232]: evrexec_param(): event route not found: evrexec:timer 0(1765) CRITICAL: <core> [core/cfg.y:3450]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 136, column 70: Can't set module parameter ERROR: bad config file (1 errors) Version info: kamailio -V version: kamailio 5.1.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 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_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 06:41:34 Mar 9 2018 with gcc 4.8.5 Thoughts? Thanks, -Skip

2 2

Removing a cipher
by Wilkins, Steve 27 Mar '18

27 Mar '18

Hello all, I am using Kamailio 5 and am trying to remove the Diffie-Hellman ciphers but have had no luck. I have researched it and have not seen any articles yet where anyone has had any luck. I need to be able to connect from webrtc port 443. Has anyone had any luck with the cipher-list? Thank you, -Steve

1 0

Dispatcher Routing Failure
by Daniel W. Graham 27 Mar '18

27 Mar '18

What constitutes a 'failed request' in dispatcher module, in which a gateway is marked inactive? >From documentation - ds_ping_interval With this parameter you can define the interval for sending a request to a gateway marked as inactive upon a failed request routing to it. ds_probing_threshold If you want to set a gateway into inactive mode, there can be a specific number of failed requests until it will change from "active" to "inactive". Thanks, -dan

2 4

Regular Expression Syntax
by Duarte Rocha 27 Mar '18

27 Mar '18

Greetings, I'm having some trouble with the syntax used in the {re.subst,expression} method. What i want to do is add a prefix to a capture group. I can make it work like that /^[0-9]+/prefix$tU/ but this solution is far from perfect since i may not need the entire $tU. I would like to use it like this /^[0-9]+/prefix\1/ in order to not limit my regex to Kamailio only. When i do it like this the regex result is only "prefix". What's the correct syntax to add a prefix using the capture group ? Thanks in advance

2 1

topos_redis and session-timer timeouts
by Daniel Tryba 27 Mar '18

27 Mar '18

I was testing topos with a mysql backend and all was fine. But since the purpose of this machine doesn't require a complete mysqlserver I tried the redis backend. With redis the calls will drop after some timeout (somewhere < 450 seconds (session timer set to 900)). Config: # topos modparam("ndb_redis", "server", "name=srv2;addr=127.0.0.1;port=6379;db=2") modparam("topos", "storage", "redis") modparam("topos_redis", "serverid", "srv2") #modparam("topos", "storage", "db") #modparam("topos", "db_url", DBURL) modparam("topos", "mask_callid", 0) modparam("topos", "sanity_checks", 0) modparam("topos", "branch_expire", 180) modparam("topos", "dialog_expire", 10800) modparam("topos", "clean_interval", 60) The call will end on the first session-time firing with a "404 Not here" Debug contains: Mar 27 12:20:56 kam kamailio[11649]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:20:56 kam kamailio[11649]: DEBUG: ndb_redis [redis_client.c:878]: redisc_exec_argv(): rsrv->ctxRedis = 0x55a7c8da8500 Mar 27 12:20:56 kam kamailio[11649]: DEBUG: topos_redis [topos_redis_storage.c:980]: tps_redis_load_dialog(): hmap with key [d:z:atpsh-5aba19b2-2d7f-2] not found This records was inserted/updated: Mar 27 12:17:10 kam kamailio[11647]: DEBUG: topos_redis [topos_redis_storage.c:253]: tps_redis_insert_dialog(): inserted dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 26 Mar 27 12:17:10 kam kamailio[11647]: DEBUG: topos_redis [topos_redis_storage.c:282]: tps_redis_insert_dialog(): expire set on dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 3 Mar 27 12:17:11 kam kamailio[11648]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:17:11 kam kamailio[11648]: DEBUG: topos_redis [topos_redis_storage.c:1293]: tps_redis_update_dialog(): updated dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 6 Mar 27 12:17:11 kam kamailio[11648]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:17:11 kam kamailio[11648]: DEBUG: topos_redis [topos_redis_storage.c:1293]: tps_redis_update_dialog(): updated dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 6 Mar 27 12:17:11 kam kamailio[11650]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:17:11 kam kamailio[11650]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:17:11 kam kamailio[11650]: DEBUG: topos_redis [topos_redis_storage.c:1381]: tps_redis_end_dialog(): updated on end the dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 6 Mar 27 12:17:11 kam kamailio[11650]: DEBUG: topos_redis [topos_redis_storage.c:1412]: tps_redis_end_dialog(): expire set on branch record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 3 It looks like the last update/expire sets the timeout to 180s, so the not found is reasonably expected. But why is it that these records aren't deleted in the mysql backend? 0000 00 00 03 04 00 06 00 00 00 00 00 00 00 00 08 00 ................ 0010 45 00 00 6d 0a 52 40 00 40 06 32 37 7f 00 00 01 E..m.R@.@.27.... 0020 7f 00 00 01 b1 0c 18 eb c4 68 f0 3b 93 17 20 b0 .........h.;.. . 0030 80 18 03 d6 fe 61 00 00 01 01 08 0a 00 08 50 67 .....a........Pg 0040 00 08 50 67 2a 33 0d 0a 24 36 0d 0a 45 58 50 49 ..Pg*3..$6..EXPI 0050 52 45 0d 0a 24 32 35 0d 0a 64 3a 7a 3a 61 74 70 RE..$25..d:z:atp 0060 73 68 2d 35 61 62 61 31 39 62 32 2d 32 64 37 66 sh-5aba19b2-2d7f 0070 2d 32 0d 0a 24 33 0d 0a 31 38 30 0d 0a -2..$3..180..

1 2

Kamailio Call monitoring
by Tomi Hakkarainen 27 Mar '18

27 Mar '18

Hi you all, I have set up Kamailio and RTPProxy in front of Asterisk. Currently successfully able to route calls from two differrent SIP-Trunk providers terminated to Kamailio and RTPProxy. I would like to get a view of calls currently running through Kamailio? I should be able to see some details about the call like the calling called numbers and the time call has been active. I’m not sure if Siremis would provide such view. Also looked Homer and also not sure it has something like that. Still those provide much more like for troubleshooting. If anyone has idea how that would be possible to accomplish or know that there already is such implementation I would be interested. Tomi

3 3

Kamailio send 2 Invite request for Client behind NAT
by amit Kumar 27 Mar '18

27 Mar '18

Hi All, Kamailio send 2 Invite request for Client behind NAT. due to this reason for one request,Client respond with 486 Busy here. we only want to send 1 INVITE request. Following are the AOR for client at kamailio.if field(ip port) in CONTACT and RECEIVED differ then kamailio send 2 INVITE request. Please advice if there is any way to send only ONE INVITE*.* AOR:: 123 Contact:: sip:123@*192.168.230.124:45672 <http://192.168.230.124:45672>* Q=1 Expires:: 894 Callid:: 003A8EC4-12351(a)192.168.230.124 Cseq:: 2 User-agent:: SIPPER for 3CX Phone Received:: sip:*204.192.206.208 <http://204.192.206.208>: 45672 * State:: CS_NEW Flags:: 0 Cflag:: 192 Socket:: udp:172.12.120.88:5060 Methods:: 4294967295 Ruid:: uloc-5aba112d-7fce-5 Reg-Id:: 0 Last-Keepalive:: 1522144583 Last-Modified:: 1522144583 thanks, amit

1 0

Kamailio NAT AWS EC2
by Ilie Soltanici 27 Mar '18

27 Mar '18

Hi, First of all, check if SIP Traffic is going through your SIP Server, not directly between Endpoints. To do this check if Record-route parameter is set in configuration file: record_route(); https://www.kamailio.org/docs/modules/5.1.x/modules/rr.html Also, be sure that in SDP body you have External IP of your Kamailio/RTPproxy. If Kamailio is behind NAT you may need to use fix_nated_sdp function from nathelper module: https://www.kamailio.org/docs/modules/5.1.x/modules/nathelper.html#nathelpe… Regards,

1 0

NOTIFY message without XML part
by denislav.tsonev＠musala.com 27 Mar '18

27 Mar '18

Hello, I use kamailio 5.1.1 with default configuration for PRESENCE: Route: #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif and modparams: #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif I use and GRANSTREAM phones. When I make subscribe from soft sip client (Micro sip) to the grandstreams, all NOTIFY messages are without XML. Subscriptions between two soft sip clients like Zoiper and Micro SIP works perfectly. Is this related to GRANDSTREAM phones - configuration or so... or is necessary to modify kamailio configuration and if Yes, please provide some hitns how to change kamailio config? Thanks in advance. Best regards. Denislav Raychev Tsonev | Integration and Infrastructure Engineer | Musala Soft JSC www.musala.com | t: +359 2 969 58 21 | m: +359 878 270 965 | f: +359 2 969 58 22 World Trade Center, 36 Dragan Tsankov blvd., Sofia 1057, Bulgaria

1 0

Strange auth_challenge behaivor on 407
by Yuriy Gorlichenko 27 Mar '18

27 Mar '18

Hi I'm using version: kamailio 5.1.2 I have strange issue with auth_challenge kamialio works as NAT handler. When Register comes kamailio handles nat and save original ip:port at the received field in location table I found that when local port in contact is different than external port reply 407 on INVITE comes to externalip (from received) : internalport(from contact) And never reaches destination for exmaple Register comes from 192.168.1.10:50555 Kamailio saves at the Received 12.1.1.12:1030 but when invite comes form 12.1.1.12:1030 kamailio sends reply to 12.1.1.12:50555 i don't think that this is an issue in kamailio source code but i can find where i can configure it

2 2

How should i configure kamailio connection freepbx?
by 星昊通技术 27 Mar '18

27 Mar '18

hello all: I install kamailio 5.1 and siremis 5.1, I want to configure kamailio connection freepbx, how should i configure kamailio vai siremis GUI? Thanks. ------------------ Best regards! alex Hiastar 深圳星昊通科技 Sangoma in APAC, 潮流网络IP话机，IPPBX Asterisk cards, SBC, NetBorder VOIP Gateway, Transcoding, 潮流网络IP话机 QQ：2933996052 website:www.hiastar.com|www.freepbx.cn, contact number: 13574109204 关注微信公众帐号：Asterisk-CN

2 1

Help on setting UP SIP PRoxy
by Mohammed Ayub 26 Mar '18

26 Mar '18

HI, Myself Ayub from Phonology, Bangalore, India. We are trying to set up an SIP proxy server using Kamailio with Siremis as web interface. Have installed and configured kamailio and siremis but need help in config if sip domain , users and etc... Please revert. Regards Ayub

2 1

Can't get route[AUTH] working as expected.
by Aqs Younas 25 Mar '18

25 Mar '18

Greetings list. I can see that I was able to bypass the default route[AUTH] if I send an invite containing from_uri which is not local but requested line containing a local user. llisten=udp:172.16.40.10:5060 route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } Below INVITE get passed above auth route. INVITE sip:60129879190@172.16.40.10 SIP/2.0 Via: SIP/2.0/UDP 139.5.177.91:5060;branch=z9hG4bK31edc7f4;rport Max-Forwards: 70 From: <sip:0128888877@139.5.177.99>;tag=as2274e806 To: <sip:60129879190@172.16.40.10> Contact: <sip:0128888877@139.5.177.91:5060> Call-ID: 7b6d32bc6c679bb23eb248b955c0ac8b@139.5.177.91:5060 CSeq: 102 INVITE User-Agent: FPBX-13.0.194.2(13.17.0) Date: Fri, 23 Mar 2018 09:33:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 321 v=0 o=root 237494576 237494576 IN IP4 139.5.177.99 s=Asterisk PBX 13.17.0 c=IN IP4 139.5.177.99 t=0 0 m=audio 15332 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv >From INVITE and route[AUTH] I can see why it is being passed. But should not it by default authenticate every request if IP address is not allowed in permission module. Br, Aqs.

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