- sr-users - kamailio.org

What is a good strategy to build Kamailio <--> pbx
by Johnny Ritzer 16 Aug '20

16 Aug '20

What are your thoughts on architecture build of the following scenarios. The PBX doesn’t support websockets and we want to use sip.js / jsSip. (use Kama as wss:// SIP proxy) PBX has direct PRI/sip Trunks/ DID control/ Voicemail. PBX hosts SIP extensions. UA --> Kamailio --> PBX --------SCENARIO A.) ------- Host SIP[xxxx] extensions on Kama box but same SIP extensions are also on PBX [xxxx] 1. Build some sort of control to register handle kama to pbx …. (but reg webclients) Kama [1100] SIP-PBX [1100] Does kama act as the UA 1100 / registered to PBX What about UA to kamailio (do we do 1100a so UA-kama is 1100a) but kama is maintain REG/status/ but passing that IF client is online ? MWI (voicemail is hosted on PBX) so how does that move through can it. ------- SCENARIO B.) ------ Proxy all/ everything . use route[REGFW] Forward all REG to PBX.. Issues ---- MWI (SIP notify ?) SIP Incoming CALL incoming calls to work. (if UA holds REG to outbound proxy on PBX or ) DTMF (sipINFO) doesn’t work (I see it sending) I noticed I crashed the PBX because all the SIP scanners started pounding my Kamailio box after 2 days when I added route[REGFW] it starting eating up SIP trunk channels on PBX cause started forwarding everything. Like it starting 4241(a)xxxxx.com 4242(a)xxxxx.com . Or might need to look at fail2ban pike module to oget this but still had issues I just don’t know where to look or code to fix on SCENARIO B Snippets on my testing . route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.PBX.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.PBX.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + "kamaproxy"; # $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif --------------- So this led me to extra security should I host SIP extensions on kamailio as an extra security but then what about MWI/inbound then) or do I have 1100 on kama which mitel is 1100 but I pass all info from inbound.

2 1

Kamailio Query
by Amit Pal 15 Aug '20

15 Aug '20

Dear Denial/Team, I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. MY first question: When subscriber kamuser1 and kamuser2 are talking then I want to transfer these two party in freeswitch conference. I am unable to generate re-invite these call . here I was also try with rewritehost(like) but not success. how can we achieve this. My second question : Even MOH not playing when any one party hold the call. I already setup KAMUser1Kamailio(RTPProxy)KAMUser2. I have completely setup the config of rtpproxy and mohqueues. log shows all signal handle by kamailio and RTPProxy but not getting moh. I am attaching my config file for reference. can we do this by mediaproxy with freeswitch ,all thing are configured as document but no success. My third question: Kamailio relay dtmf on sipinfo. How can we get dtmf in between the call and taking actions on it. I am highly thankful for your great support . Thanks Regards

1 0

Kamailio Team Attentions
by Amit Pal 15 Aug '20

15 Aug '20

Dear Team, “I am unable to reach you, my previous mail get bounce. I received bounce mail from sr-users. Below is my previous mail contains.” Last Sent Mail Contents I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. MY first question: When subscriber kamuser1 and kamuser2 are talking then I want to transfer these two party in freeswitch conference. I am unable to generate re-invite these call . here I was also try with rewritehost(like) but not success. how can we achieve this. My second question : Even MOH not playing when any one party hold the call. I already setup KAMUser1Kamailio(RTPProxy)KAMUser2. I have completely setup the config of rtpproxy and mohqueues. log shows all signal handle by kamailio and RTPProxy but not getting moh. I am attaching my config file for reference. can we do this by mediaproxy with freeswitch ,all thing are configured as document but no success. My third question: Kamailio relay dtmf on sipinfo. How can we get dtmf in between the call and taking actions on it. I am highly thankful for your great support . Thanks Regards Neol

1 0

Required Kamailio clarification!!!
by Amit Pal 15 Aug '20

15 Aug '20

Dear Team, I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. MY first question: When subscriber kamuser1 and kamuser2 are talking then I want to transfer these two party in freeswitch conference. I am unable to generate re-invite these call . here I was also try with rewritehost(like) but not success. how can we achieve this. My second question : Even MOH not playing when any one party hold the call. I already setup KAMUser1==>Kamailio(RTPProxy)==>KAMUser2. I have completely setup the config of rtpproxy and mohqueues. log shows all signal handle by kamailio and RTPProxy but not getting moh. I am attaching my config file for reference. can we do this by mediaproxy with freeswitch ,all thing are configured as document but no success. My third question: Kamailio relay dtmf on sipinfo. How can we get dtmf in between the call and taking actions on it. I am highly thankful for your great support . Thanks Regards Neol

1 0

Can the callid be modified/mapped when branching a call?
by Magnus Kelly 14 Aug '20

14 Aug '20

Good evening all, Can I ask what might be a basic question, but I would like to confirm if the CallID field can be modified/mapped when branching a call ? Thanks Magnus

2 2

How to check TLS versions available
by David Cunningham 14 Aug '20

14 Aug '20

Hello, Does anyone know of a method to check what TLS versions are available from Kamailio for clients to use? For example, is TLS 1.0 available, TLS 1.1, etc. Thanks in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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Segfault starting kamailio, function "registered"
by Jon Bonilla (Manwe) 14 Aug '20

14 Aug '20

Hi when I start kamailio in my dev system it fails to start and segfaults. Version: 5.4.0 CFG Line: if (registered("kam_location",$tu)) { Log: Aug 14 10:20:29 pekedev2 kamailio: CRITICAL: <core> [core/cfg.y:3588]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 382, column 34-36: function registered: parameter 2 is not constant#012 Aug 14 10:20:29 pekedev2 kernel: [1501045.504630] kamailio[21276]: segfault at 6e6f6974 ip 00005566607d8887 sp 00007ffd6761b240 error 4 in kamailio[55666079d000+431000] Aug 14 10:20:29 pekedev2 kernel: [1501045.504655] Code: 89 bd 68 ff ff ff 64 48 8b 04 25 28 00 00 00 48 89 45 c8 31 c0 48 83 bd 68 ff ff ff 00 0f 84 f5 09 00 00 48 8b 85 68 ff ff ff <8b> 00 83 f8 01 0f 85 65 09 00 00 48 8b 85 68 ff ff ff 8b 40 0c 85 The coredump is not clear to me. I extracted using coredumpctl (fuck you systemd) and running gdb. Is there a new method now that systemd has taken control of my systems? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

conditional on secfilter seems does not work or how to do it
by PICCORO McKAY Lenz 14 Aug '20

14 Aug '20

I implemented secfilter in a simple way, in first step routing put that conditional for black list check that already works: ``` secf_check_ip(); if ($? == -2) { xlog("L_ALERT", "$rm from $si is blacklisted"); drop(); } ``` BUT NOW i want to change to whitelick checks, so i reviewed the docs and "2" is resulting for whitelist, so then i said "if not whitelisted so block" using "!= 2" as on https://www.kamailio.org/wiki/cookbooks/5.3.x/core#operators BUT SEEMS DOES NOT WORK: that is the code: ``` secf_check_ip(); if ($? != 2) { xlog("L_ALERT", "$rm from $si is not in whitelist, block"); drop(); } ``` when i tested all the calls passed not matter if are in the table or not as whitelist! I want able to call only if are present and whitelisted the ip address.. so i also tested with: ``` secf_check_ip(); if ($? == 0) { xlog("L_ALERT", "$rm from $si is not present, so block"); drop(); } ``` to check if the ip address are presented in the table but call also passed and must not cos i not put never a entry in the table! Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 3

Kamailio drop call with Teams
by sip user 13 Aug '20

13 Aug '20

Hi, I'm connecting Teams with kamailio server. From Kamailio to teams I have no problems, but from teams to Kamailio yes. Drop the call.. With ngrep I see that: INVITE: [image: imagen.png] 200 OK: [image: imagen.png] I never received ACK.. In my configuration: Kamailio.cfg: #!KAMAILIO #!define WITH_TLS event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:FQND:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } request_route{ remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","$fU is trying to call to $rU con valores $tu\n"); $ru = "sip:" + $rU + "@ip asterisk:port"; route(RELAY); } } What I'm doing wrong? I don't understand why not received ACK.. Could anyone help me? Thanks

1 0

Kamailio lookup via Rest Interface
by krishdinesh＠yahoo.com 13 Aug '20

13 Aug '20

Hi, I am trying to do the following and wanted to seek help on how this can be done in Kamailio and if it feasible to do this. If not, whether there are other ways to do this. I would like to do the following .. If there is an incoming call, kamailio should be able to send the incoming calling number to an external application via Rest Interface, the external application will return a response on what to do with this call, it can be simple as sending a call to the called extension or a media server like asterisk for further treatment. I have a kamailio version 5.3.5 running on CentOS 7. I have a few end points registered to kamailio and they are able to call each other Any help would be appreciated. Thank you. Dinesh

1 0

Kamailio v5.4.0 RPM Availability
by Asgaroth 13 Aug '20

13 Aug '20

Hi all, I would just like to check in and find out if/when the Kamailio v5.4.0 rpm's will be available on the OpenSUSE build service. Thanks to all involved for all the hard work in this new release! Thanks There are two major products that came out of Berkeley: LSD and UNIX. We don't believe this to be a coincidence. -- Jeremy S. Anderson 00asgaroth00(a)gmail.com (https://link.getmailspring.com/link/146DE51A-E8C2-4126-B8D5-EA45E62CBA11@ge…)

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smsops module - status report?
by Pafel 13 Aug '20

13 Aug '20

Hello, I am trying to generate sms status report using smsops module but without success. I couldn't find any example on the net. Can somebody help me? According to https://en.wikipedia.org/wiki/GSM_03.40 TP-MTI (tpdu(type) in kamailio) should be "1 0" for status report. Delivery is "0 0" and submit "0 1". According to the provided examples in smsops module: tpdu(type) = 4 is delivery tpdu(type) = 1 is submit So what's the number for status? I could't find the mapping in the code. Only this: // Types of the PDU-Message typedef enum _pdu_message_type { DELIVER = 0x00, SUBMIT = 0x01, COMMAND = 0x02, ANY = 0x03, } pdu_message_type_t; So any hint or example is appreciated. Thanks, Pavel Siderov

1 0

Log-level Kamailio IMS
by Pavithra Mohanraja 13 Aug '20

13 Aug '20

Hi, Is there any way to have global log level changes without restart of kamailio. If there kindly let me know. Thanks, Pavithra

3 6

Mysql lastinsertid
by Kjeld Flarup 12 Aug '20

12 Aug '20

Hello I wonder if there is some way to get the lastinsertid from an "INSERT INTO" sql_query -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

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Best Module for CPS per IP - PIKE or Ratelimit modules ?
by Ameed Jamous 12 Aug '20

12 Aug '20

Dear all, I wanted to confirm, which is the best way to implement CPS limits per source IP in kamilio in 2020? PIKE module or Rate Limit module or both combined. Thanks in advance for your support. -- AJ

3 5

smsops module - status report?
by Pavel Siderov 12 Aug '20

12 Aug '20

Hello, I am trying to generate sms status report using smsops module but without success. I couldn't find any example on the net. Can somebody help me? According to https://en.wikipedia.org/wiki/GSM_03.40 TP-MTI (tpdu(type) in kamailio) should be "1 0" for status report. Delivery is "0 0" and submit "0 1". According to the provided examples in smsops module: tpdu(type) = 4 is delivery tpdu(type) = 1 is submit So what's the number for status? I could't find the mapping in the code. Only this: // Types of the PDU-Message typedef enum _pdu_message_type { DELIVER = 0x00, SUBMIT = 0x01, COMMAND = 0x02, ANY = 0x03, } pdu_message_type_t; So any hint or example is appreciated. Thanks, Pavel Siderov

1 0

Kamalio 5.4.0 should support ka_add_destination function of KeepAlive module ?
by sagar malam 12 Aug '20

12 Aug '20

Hello I want to use ka_add_destination function of KA module but i am getting this error : [root@SBC-4-1] kamailio -c -f /usr/local/etc/kamailio/kamailio.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(48016) ERROR: <core> [core/cfg.y:3451]: yyparse(): cfg. parser: failed to find command ka_add_destination (params 1) 0(48016) CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 1449, column 58: unknown command, missing loadmodule? I know that 5.3.3 version does not support this function but 5.4.0 should as per documentation : https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keepalive.ka… Keepalive module is compiled and loaded in cfg. -- Thanks, Sagar

3 5

Listen IP changes in IMS kamailio
by Pavithra Mohanraja 12 Aug '20

12 Aug '20

Hi, Is it possible to make it work without giving listen IP in pcscf.cfg, icscf.cfg, scscf.cfg files since i am configuring it as kubernetes pods and automating all configurations. Due to this, I am unable to start the kamailio service without giving the proper listen IP. Is there any alternative way. Kindly Help Thanks, Pavithra

1 0

Example config for webRTC-SIP kamailio.cfg for sip.js
by Johnny Ritzer 12 Aug '20

12 Aug '20

Anyone know where to grab an example config for getting web socket-sip working. I tried this one but its failing on kamailio restart it might be some very small thing. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg wss://sip.kamailio-box.com:8080 Client = sip.js (webrtc) SIP server = (mitel) PBX (straight sip)

3 2

Call in Progress Recovery in Redundancy
by amitsharma＠coraltele.com 12 Aug '20

12 Aug '20

Dear Daniel/Team, I had raised one question in "Workshop 3 - Kamailio" at Cluecon 2020(Last Night), i.e. Can Progress Call(Ringing Calls) be recovered in case of redundancy with Kamailio. You were told me that straight way it is not possible but try with hash table once. I had tried following link https://wazo-platform.org/blog/kamailio-ha-dispatcher-and-dmq and able to recover Call in progress within 2-3 nodes. 1. My one question is that either this approach will work in production or not. 2. I have been using Freeswitch for last 6-7 years but "Call in Progress Recovery in Redundancy" is not possible there in "Freeswitch", So I tried Kamailio and got success. My Second question is that can it be possible that Call established on Kamailio and after call set up Kamailio leave that call and handed over it to Freeswitch for further processing(Like Re-homing available in OpenSIPS). This will save years of time that I have invested building features around Freeswitch. Please suggest me the best way possible to achieve this. 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

3 7

Kamailio configuration ??
by Amit Pal 12 Aug '20

12 Aug '20

Dear Team, I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. Now I want ask is this possible at kamailio? "when subscriber establish their call then kamailio handed over its call control to freeswitch" i.e. when kamailio got "ACK" (call patch signal) of two subscriber then call control will be transferred to freeswitch. Here I am Using Kamailio for its excellent feature recovery of progress call (by DMQ and HTable) in Highly Availability(HA) . Freeswitch doesn't have this feature. But freeswitch we are using as application server. I have configure Kamailio like "KMUser1==>FS==> KAMUser2" . but in that case progress call (KMUser1 Get alert of KMUser2) has not recover on HA. So call will be on kamailio till alerting after connect call will be transferred to Freeswitch. I want to know by which module we can achieve this. It will be great help for me if provided sample . Thanks Amit Pal Coral Telecom Ltd

1 0

ACK no transaction found if re-invite
by Techinnovation 12 Aug '20

12 Aug '20

Hello, Please help. There was issue topic #2436. I have answer: There is some problem with handling Record-Route in client side, maybe not supporting it, because the ACK has no Route header. Is there any hotfix for kamailio ?,As I not sure that customer can change ACK on customers side . ======================== I use kamailio 5.2.7 with Asterisk 16.1.1 . Call Flow: Customer(212.25.64.8) -> Kamailio(48.146.161.152) -> Asterisk(48.146.161.153) ->Kamailio(48.146.161.152) -> Provider(89.210.243.1) I have one issue with call when I receive re-invite in dialog from Provider. My Asterisk not receive ACK from kamailio. I see error on log like "ACK no transaction found" With this I have difference with 30 seconds. Session Customer(212.25.64.8) -> Kamailio(48.146.161.152) -> Asterisk(48.146.161.153) - has 2 seconds Asterisk(48.146.161.153) ->Kamailio(48.146.161.152) -> Provider(89.210.243.1) - has 32 seconds I have this WITHINDLG settings: route[WITHINDLG] { if (has_totag()) { # 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); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } t_relay(); } exit; } } =================================== In kamailio logs I see ACK no transaction found: Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: [core/receive.c:246]: receive_msg(): --- received sip message - reply - call-id: [4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060] - cseq: [12375830 INVITE] Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=270 Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1043]: t_check_msg(): msg (0x7f330aed6050) id=296/1926 global id=295/1926 T start=0xffffffffffffffff Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:920]: t_reply_matching(): t_reply_matching: hash 11128 label 0 branch 0 Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1013]: t_reply_matching(): no matching transaction exists Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1016]: t_reply_matching(): failure to match a transaction Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1113]: t_check_msg(): msg (0x7f330aed6050) id=296/1926 global id=296/1926 T end=(nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_reply.c:2230]: reply_received(): transaction not found - (branch -1) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/msg_translator.c:2290]: generate_res_buf_from_sip_res(): old size: 932, new size: 821 Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/msg_translator.c:2308]: generate_res_buf_from_sip_res(): copied size: orig:127, new: 16, rest: 805 msg= Aug 11 00:06:11 c4-proxy-1 kamailio: SIP/2.0 200 OK Aug 11 00:06:11 c4-proxy-1 kamailio: Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199 Aug 11 00:06:11 c4-proxy-1 kamailio: From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03 Aug 11 00:06:11 c4-proxy-1 kamailio: To: sip:393817097132@48.146.161.153;tag=as73e66fe7 Aug 11 00:06:11 c4-proxy-1 kamailio: Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060 Aug 11 00:06:11 c4-proxy-1 kamailio: CSeq: 12375830 INVITE Aug 11 00:06:11 c4-proxy-1 kamailio: Server: Asterisk PBX 16.1.1 Aug 11 00:06:11 c4-proxy-1 kamailio: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Aug 11 00:06:11 c4-proxy-1 kamailio: Supported: replaces, timer Aug 11 00:06:11 c4-proxy-1 kamailio: Contact: sip:393817097132@48.146.161.153:5060 Aug 11 00:06:11 c4-proxy-1 kamailio: Content-Type: application/sdp Aug 11 00:06:11 c4-proxy-1 kamailio: Content-Length: 270 Aug 11 00:06:11 c4-proxy-1 kamailio: v=0 Aug 11 00:06:11 c4-proxy-1 kamailio: o=root 1995384227 1995384228 IN IP4 48.146.161.153 Aug 11 00:06:11 c4-proxy-1 kamailio: s=Asterisk PBX 16.1.1 Aug 11 00:06:11 c4-proxy-1 kamailio: c=IN IP4 48.146.161.153 Aug 11 00:06:11 c4-proxy-1 kamailio: t=0 0 Aug 11 00:06:11 c4-proxy-1 kamailio: m=audio 18584 RTP/AVP 0 8 18 Aug 11 00:06:11 c4-proxy-1 kamailio: a=rtpmap:0 PCMU/8000 Aug 11 00:06:11 c4-proxy-1 kamailio: a=rtpmap:8 PCMA/8000 Aug 11 00:06:11 c4-proxy-1 kamailio: a=rtpmap:18 G729/8000 Aug 11 00:06:11 c4-proxy-1 kamailio: a=fmtp:18 annexb=no Aug 11 00:06:11 c4-proxy-1 kamailio: a=ptime:20 Aug 11 00:06:11 c4-proxy-1 kamailio: a=maxptime:150 Aug 11 00:06:11 c4-proxy-1 kamailio: a=sendrecv Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:691]: update_sock_struct_from_via(): using via host Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:700]: update_sock_struct_from_via(): check if dns lookup is needed for [89.210.243.1] Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:710]: update_sock_struct_from_via(): address is [89.210.243.1] port 5060 proto 1 Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:830]: do_forward_reply(): reply forwarded (2nd via address: 89.210.243.1 port 5060) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:839]: do_forward_reply(): reply forwarding finished (2nd via address: 89.210.243.1 port 5060) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:442]: receive_msg(): reply-route executed in: 115 usec Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/xavp.c:507]: xavp_destroy_list(): destroying xavp list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:458]: receive_msg(): cleaning up Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/405) [[ACK sip:393817097132@48.146.161.153:5060 SIP/2.0 0D 0A Call-ID: 4a681d8626b801ce1a8185472fda8e86(a)37.148.1]] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:610]: parse_msg(): SIP Request: Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:612]: parse_msg(): method: Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:614]: parse_msg(): uri: sip:393817097132@48.146.161.153:5060 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:152]: get_hdr_field(): cseq : <12375830> Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=as73e66fe7 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:171]: get_hdr_field(): [50]; uri=[sip:393817097132@48.146.161.153] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:174]: get_hdr_field(): to body [sip:393817097132@48.146.161.153], to tag [as73e66fe7] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, = ; state=16 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:500]: parse_headers(): this is the first via Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/receive.c:246]: receive_msg(): --- received sip message - request - call-id: [4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060] - cseq: [12375830 ACK] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=0 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:296]: receive_msg(): preparing to run routing scripts... Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} maxfwd [mf_funcs.c:74]: is_maxfwd_present(): value = 70 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:247]: check_ruri_sip_version(): check_ruri_sip_version entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:270]: check_ruri_sip_version(): check_ruri_sip_version passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:305]: check_required_headers(): check_required_headers entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:313]: check_required_headers(): check_required_headers passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:479]: check_cseq_method(): check_cseq_method entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:512]: check_cseq_method(): check_cseq_method passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:521]: check_cseq_value(): check_cseq_value entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:549]: check_cseq_value(): check_cseq_value passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:558]: check_cl(): check_cl entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:583]: check_cl(): check_cl passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:595]: check_expires_value(): check_expires_value entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:632]: check_expires_value(): no expires header found Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:732]: check_parse_uris(): check_parse_uris entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:736]: check_parse_uris(): parsing ruri Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:750]: check_parse_uris(): looking up From header Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:764]: check_parse_uris(): parsing From header Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=00-07989-012374ff-364731f03 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:790]: check_parse_uris(): parsing From URI Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:808]: check_parse_uris(): looking up To header Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:830]: check_parse_uris(): check_parse_uris(): parsing To URI Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:875]: check_parse_uris(): check_parse_uris passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 1 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} siputils [checks.c:123]: has_totag(): totag found Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} rr [loose.c:114]: find_first_route(): No Route headers found Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} rr [loose.c:961]: loose_route(): There is no Route HF Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1043]: t_check_msg(): msg (0x7f330aed6050) id=300/1925 global id=299/1925 T start=(nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:500]: t_lookup_request(): start searching: hash=11128, isACK=1 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:458]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK-173F80F1-114A19A] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:682]: t_lookup_request(): no transaction found Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1113]: t_check_msg(): msg (0x7f330aed6050) id=300/1925 global id=300/1925 T end=(nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:354]: receive_msg(): request-route executed in: 155 usec Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/xavp.c:507]: xavp_destroy_list(): destroying xavp list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:458]: receive_msg(): cleaning up ============================================================================================================= PCAP trace U 2020/08/11 00:05:51.498554 48.146.161.153:5060 -> 48.146.161.152:5060 #96 INVITE sip:79409098828@212.25.64.8:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. Max-Forwards: 70. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060. Contact: sip:393817097132@48.146.161.153:5060. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 16.1.1. Date: Mon, 10 Aug 2020 21:05:51 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 324. . v=0. o=root 926780413 926780413 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 16980 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ............1_H. U 2020/08/11 00:05:51.499026 48.146.161.152:5060 -> 48.146.161.153:5060 #97 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. Server: kamailio (5.2.7 (x86_64/linux)). Content-Length: 0. . x.....1_........ U 2020/08/11 00:05:51.499105 48.146.161.152:5060 -> 212.25.64.8:5060 #98 INVITE sip:79409098828@212.25.64.8:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK415f.07dfd4858a8d7af10f14a65a40c09040.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. Max-Forwards: 69. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060. Contact: sip:393817097132@48.146.161.153:5060. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 16.1.1. Date: Mon, 10 Aug 2020 21:05:51 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 324. . v=0. o=root 926780413 926780413 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 16980 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. .............1_. U 2020/08/11 00:05:51.523721 212.25.64.8:5060 -> 48.146.161.152:5060 #99 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK415f.07dfd4858a8d7af10f14a65a40c09040.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. . ................ U 2020/08/11 00:05:51.614735 212.25.64.8:5060 -> 48.146.161.152:5060 #104 INVITE sip:79409098828@48.146.161.152:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone. CSeq: 1 INVITE. P-Asserted-Identity: sip:393817097132@10.96.16.6;user=phone. Max-Forwards: 66. Contact: sip:393817097132@212.25.64.8:5060;transport=udp. Min-SE: 90. Session-Expires: 1800. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. User-Agent: Huawei SoftX3000 V600R013. Supported: timer. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222851 36222852 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10892 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. ........1_...$.. U 2020/08/11 00:05:51.615233 48.146.161.152:5060 -> 212.25.64.8:5060 #105 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone. CSeq: 1 INVITE. Server: kamailio (5.2.7 (x86_64/linux)). Content-Length: 0. . ............1_.@ U 2020/08/11 00:05:51.615334 48.146.161.152:5060 -> 48.146.161.153:5060 #106 INVITE sip:79409098828@48.146.161.152:5060;user=phone SIP/2.0. Record-Route: sip:48.146.161.152;lr. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.f74d055cfd186a85dacedcc9f4d282ce.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone. CSeq: 1 INVITE. P-Asserted-Identity: sip:393817097132@10.96.16.6;user=phone. Max-Forwards: 65. Contact: sip:393817097132@212.25.64.8:5060;transport=udp. Min-SE: 90. Session-Expires: 1800. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. User-Agent: Huawei SoftX3000 V600R013. Supported: timer. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222851 36222852 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10892 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. ..........1_*..$ U 2020/08/11 00:05:51.616177 48.146.161.153:5060 -> 48.146.161.152:5060 #107 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.f74d055cfd186a85dacedcc9f4d282ce.0;received=48.146.161.152;rport=5060. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Length: 0. . ................ U 2020/08/11 00:05:51.690457 48.146.161.153:5060 -> 48.146.161.152:5060 #112 INVITE sip:79409098828@89.210.243.1:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Max-Forwards: 70. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. To: sip:79409098828@89.210.243.1:5060. Contact: sip:393817097132@48.146.161.153:5060. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 16.1.1. Date: Mon, 10 Aug 2020 21:05:51 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. P-Asserted-Identity: sip:393817097132@48.146.161.152. Content-Type: application/sdp. Content-Length: 326. . v=0. o=root 1995384227 1995384227 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ........1_K.-)}. U 2020/08/11 00:05:51.690857 48.146.161.152:5060 -> 48.146.161.153:5060 #113 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. To: sip:79409098828@89.210.243.1:5060. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 102 INVITE. Server: kamailio (5.2.7 (x86_64/linux)). Content-Length: 0. . .............1_D U 2020/08/11 00:05:51.690934 48.146.161.152:5060 -> 89.210.243.1:5060 #114 INVITE sip:79409098828@89.210.243.1:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK8b4f.c29b6618fd22db1ce4b8aec0d0dcf809.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Max-Forwards: 69. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. To: sip:79409098828@89.210.243.1:5060. Contact: sip:393817097132@48.146.161.153:5060. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 16.1.1. Date: Mon, 10 Aug 2020 21:05:51 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. P-Asserted-Identity: sip:393817097132@48.146.161.152. Content-Type: application/sdp. Content-Length: 326. . v=0. o=root 1995384227 1995384227 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_...). U 2020/08/11 00:05:51.697205 89.210.243.1:5060 -> 48.146.161.152:5060 #115 SIP/2.0 100 Trying. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060. Via: SIP/2.0/UDP 48.146.161.152;received=48.146.161.152;rport=5060;branch=z9hG4bK8b4f.c29b6618fd22db1ce4b8aec0d0dcf809.0,SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 0. . ................ U 2020/08/11 00:05:51.890363 89.210.243.1:5060 -> 48.146.161.152:5060 #120 SIP/2.0 183 In band info available. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Via: SIP/2.0/UDP 48.146.161.152;received=48.146.161.152;rport=5060;branch=z9hG4bK8b4f.c29b6618fd22db1ce4b8aec0d0dcf809.0,SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 185. . v=0. o=cp10 159709355110 159709355111 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. sip+3+......1_.. U 2020/08/11 00:05:51.890412 89.210.243.1:5060 -> 48.146.161.152:5060 #121 SIP/2.0 200 OK. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. P-Asserted-Identity: sip:79409098828@89.210.243.1;user=phone. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Via: SIP/2.0/UDP 48.146.161.152;received=48.146.161.152;rport=5060;branch=z9hG4bK8b4f.c29b6618fd22db1ce4b8aec0d0dcf809.0,SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 185. . v=0. o=cp10 159709355110 159709355111 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. ........1_...5.. U 2020/08/11 00:05:51.890666 48.146.161.152:5060 -> 48.146.161.153:5060 #122 SIP/2.0 183 In band info available. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 185. . v=0. o=cp10 159709355110 159709355111 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. ..0.....1_.].5.. U 2020/08/11 00:05:51.890854 48.146.161.152:5060 -> 48.146.161.153:5060 #123 SIP/2.0 200 OK. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. P-Asserted-Identity: sip:79409098828@89.210.243.1;user=phone. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 185. . v=0. o=cp10 159709355110 159709355111 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. ............1_iA U 2020/08/11 00:05:51.891240 48.146.161.153:5060 -> 48.146.161.152:5060 #124 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.f74d055cfd186a85dacedcc9f4d282ce.0;received=48.146.161.152;rport=5060. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Type: application/sdp. Require: timer. Content-Length: 324. . v=0. o=root 792201108 792201108 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 17696 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ......P.....1_.. U 2020/08/11 00:05:51.891455 48.146.161.152:5060 -> 212.25.64.8:5060 #125 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Type: application/sdp. Require: timer. Content-Length: 324. . v=0. o=root 792201108 792201108 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 17696 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ......1_..#5X... U 2020/08/11 00:05:51.891522 48.146.161.153:5060 -> 48.146.161.152:5060 #126 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.f74d055cfd186a85dacedcc9f4d282ce.0;received=48.146.161.152;rport=5060. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Type: application/sdp. Require: timer. Content-Length: 324. . v=0. o=root 792201108 792201108 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 17696 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ....H.....1_..%5 U 2020/08/11 00:05:51.891651 48.146.161.152:5060 -> 212.25.64.8:5060 #127 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Type: application/sdp. Require: timer. Content-Length: 324. . v=0. o=root 792201108 792201108 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 17696 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ......8.....1_.. U 2020/08/11 00:05:51.896994 89.210.243.1:5060 -> 48.146.161.152:5060 #128 INVITE sip:393817097132@48.146.161.153:5060 SIP/2.0. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 12375830 INVITE. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. Content-Length: 185. . v=0. o=cp10 159709355110 159709355112 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. .............1_. U 2020/08/11 00:05:51.897340 48.146.161.152:5060 -> 89.210.243.1:5060 #129 SIP/2.0 100 trying -- your call is important to us. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. Server: kamailio (5.2.7 (x86_64/linux)). Content-Length: 0. . .........1_FO}5$ U 2020/08/11 00:05:51.897404 48.146.161.152:5060 -> 48.146.161.153:5060 #130 INVITE sip:393817097132@48.146.161.153:5060 SIP/2.0. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 12375830 INVITE. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 69. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. Content-Length: 185. . v=0. o=cp10 159709355110 159709355112 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. ......1_F..5.... U 2020/08/11 00:05:51.897746 48.146.161.153:5060 -> 48.146.161.152:5060 #131 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Length: 0. . .......(.....1_I U 2020/08/11 00:05:51.897836 48.146.161.153:5060 -> 48.146.161.152:5060 #132 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ..........1_U@.5 U 2020/08/11 00:05:51.898121 48.146.161.152:5060 -> 89.210.243.1:5060 #133 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_..5. U 2020/08/11 00:05:51.903991 89.210.243.1:5060 -> 48.146.161.152:5060 #134 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:05:51.947711 212.25.64.8:5060 -> 48.146.161.152:5060 #146 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK415f.07dfd4858a8d7af10f14a65a40c09040.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060;tag=f5d48kwf-CC-1021. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. Contact: sip:79409098828@212.25.64.8:5060;transport=udp. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222850 36222851 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10052 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. .............1_v U 2020/08/11 00:05:51.947724 212.25.64.8:5060 -> 48.146.161.152:5060 #147 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK415f.07dfd4858a8d7af10f14a65a40c09040.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060;tag=f5d48kwf-CC-1021. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. Session-Expires: 1800;refresher=uac. Require: timer. Contact: sip:79409098828@212.25.64.8:5060;transport=udp. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222850 36222851 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10052 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. ......1_X..8U... U 2020/08/11 00:05:51.947972 48.146.161.152:5060 -> 48.146.161.153:5060 #148 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060;tag=f5d48kwf-CC-1021. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. Session-Expires: 1800;refresher=uac. Require: timer. Contact: sip:79409098828@212.25.64.8:5060;transport=udp. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222850 36222851 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10052 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. .............1_. U 2020/08/11 00:05:51.987695 212.25.64.8:5060 -> 48.146.161.152:5060 #151 ACK sip:79409098828@48.146.161.153:5060 SIP/2.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. CSeq: 1 ACK. Max-Forwards: 69. Content-Length: 0. Route: sip:48.146.161.152;lr. . ..........1_...: U 2020/08/11 00:05:51.988125 48.146.161.152:5060 -> 48.146.161.153:5060 #152 ACK sip:79409098828@48.146.161.153:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.1f078d36629b5ee0771933dbd92008e5.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. CSeq: 1 ACK. Max-Forwards: 68. Content-Length: 0. . ................ U 2020/08/11 00:05:52.397845 48.146.161.153:5060 -> 48.146.161.152:5060 #157 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ..........1_2... U 2020/08/11 00:05:52.398105 48.146.161.152:5060 -> 89.210.243.1:5060 #158 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_A.... U 2020/08/11 00:05:52.403706 89.210.243.1:5060 -> 48.146.161.152:5060 #159 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:05:53.397379 48.146.161.153:5060 -> 48.146.161.152:5060 #168 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ..........1_.... U 2020/08/11 00:05:53.397668 48.146.161.152:5060 -> 89.210.243.1:5060 #169 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_.#... U 2020/08/11 00:05:53.403448 89.210.243.1:5060 -> 48.146.161.152:5060 #170 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:05:54.201755 212.25.64.8:5060 -> 48.146.161.152:5060 #177 BYE sip:79409098828@48.146.161.153:5060 SIP/2.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0cd0000010.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. CSeq: 2 BYE. Reason: Q.850;cause=16;text="normal call clearing". Max-Forwards: 69. Content-Length: 0. Route: sip:48.146.161.152;lr. . .d...........1_. U 2020/08/11 00:05:54.202256 48.146.161.152:5060 -> 48.146.161.153:5060 #178 BYE sip:79409098828@48.146.161.153:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKcbce.1e200d33771522a7baea07b052e62613.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0cd0000010.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. CSeq: 2 BYE. Reason: Q.850;cause=16;text="normal call clearing". Max-Forwards: 68. Content-Length: 0. . ......1_....m... U 2020/08/11 00:05:54.203015 48.146.161.153:5060 -> 48.146.161.152:5060 #179 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKcbce.1e200d33771522a7baea07b052e62613.0;received=48.146.161.152;rport=5060. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0cd0000010.1. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 2 BYE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Length: 0. . .............1_. U 2020/08/11 00:05:54.203229 48.146.161.152:5060 -> 212.25.64.8:5060 #180 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0cd0000010.1. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 2 BYE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Length: 0. . ................ U 2020/08/11 00:05:55.397519 48.146.161.153:5060 -> 48.146.161.152:5060 #189 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. RE.G......1_.... U 2020/08/11 00:05:55.397793 48.146.161.152:5060 -> 89.210.243.1:5060 #190 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .;.......1_..... U 2020/08/11 00:05:55.403541 89.210.243.1:5060 -> 48.146.161.152:5060 #191 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . .....m.JXH._.`.5 I 2020/08/11 00:05:58.447029 37.148.171.161 -> 48.146.161.152 3:1 #198 ....E....w..?...%....DB.......)0OPTIONS sip:193.68.66.131 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK137c.07826ed187328c4cea4207d10acdeae3.0. Via: SIP/2.0/UDP 37.148.171.164:5060;branch=z9hG4bK13044a4f. Max-Forwards: 69. From: "asterisk" sip:asterisk@37.148.171.164;tag=as4239ef1a. To: sip:193.68.66.131. Contact: sip:asterisk@37.148.171.164:5060. Call-ID: 5acd62973c560c4c7544f5c67e551e86@37.148.171.164:5060. CSeq: 102 OPTIONS. User-Agent: Asterisk PBX GIT-master-c83a44c. Date: Mon, 10 Aug 2020 21:05:37 GMT. Allow: INVITE, ACK, CAsip:393817097132 U 2020/08/11 00:05:59.398803 48.146.161.153:5060 -> 48.146.161.152:5060 #203 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. nten......1_.... U 2020/08/11 00:05:59.399150 48.146.161.152:5060 -> 89.210.243.1:5060 #204 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_..".. U 2020/08/11 00:05:59.404928 89.210.243.1:5060 -> 48.146.161.152:5060 #205 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:06:03.398077 48.146.161.153:5060 -> 48.146.161.152:5060 #214 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .70.......1_8T.. U 2020/08/11 00:06:03.398349 48.146.161.152:5060 -> 89.210.243.1:5060 #215 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ..%......1_..... U 2020/08/11 00:06:03.404194 89.210.243.1:5060 -> 48.146.161.152:5060 #216 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . .pBk.^.<.[..P..w U 2020/08/11 00:06:07.398394 48.146.161.153:5060 -> 48.146.161.152:5060 #230 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ster......1_.... U 2020/08/11 00:06:07.398701 48.146.161.152:5060 -> 89.210.243.1:5060 #231 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_..... U 2020/08/11 00:06:07.404413 89.210.243.1:5060 -> 48.146.161.152:5060 #232 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:06:11.397466 48.146.161.153:5060 -> 48.146.161.152:5060 #249 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ster......1_n... U 2020/08/11 00:06:11.397713 48.146.161.152:5060 -> 89.210.243.1:5060 #250 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_j.... U 2020/08/11 00:06:11.403609 89.210.243.1:5060 -> 48.146.161.152:5060 #251 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . gth......1_....b U 2020/08/11 00:06:15.398243 48.146.161.153:5060 -> 48.146.161.152:5060 #266 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. $..;......1_.... U 2020/08/11 00:06:15.398528 48.146.161.152:5060 -> 89.210.243.1:5060 #267 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. t........1_..... U 2020/08/11 00:06:15.404350 89.210.243.1:5060 -> 48.146.161.152:5060 #268 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . .zD8...#:g..!.. U 2020/08/11 00:06:19.397384 48.146.161.153:5060 -> 48.146.161.152:5060 #293 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. L, O......1_lc.. U 2020/08/11 00:06:19.397632 48.146.161.152:5060 -> 89.210.243.1:5060 #294 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_..... U 2020/08/11 00:06:19.403487 89.210.243.1:5060 -> 48.146.161.152:5060 #295 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:06:23.398710 48.146.161.153:5060 -> 48.146.161.152:5060 #313 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ._|r......1_f=.. U 2020/08/11 00:06:23.398998 48.146.161.152:5060 -> 89.210.243.1:5060 #314 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. rQ.......1_..... U 2020/08/11 00:06:23.404721 89.210.243.1:5060 -> 48.146.161.152:5060 #315 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ...@...X.3.....W BR, Alex

2 3

Change Record Route onreply
by Aidar Kamalov 11 Aug '20

11 Aug '20

Hello, I have kamailio with only one interface. Is it possible to set one RR - lan ip address to INVITEs (to dispatcher list members) and 127.0.0.1 to replies 200OK (to softphone)? I can't use record_route() function in onreply route because: Command cannot be used in the block. -- Aydar A. Kamalov

2 2

dns_naptr_ignore_rfc default value
by Juha Heinanen 10 Aug '20

10 Aug '20

What is the point of dns_naptr_ignore_rfc default value: dns_naptr_ignore_rfc If the DNS lookup should ignore the remote side's protocol preferences, as indicated by the Order field in the NAPTR records and mandated by RFC 2915. dns_naptr_ignore_rfc = yes | no (default yes) In my (and RFC 2915's) opinion the default should be 'no'. -- Juha

2 3

CallerID - multiple ISP Prefix
by Ilie Soltanici 10 Aug '20

10 Aug '20

Hello, We are using a few carries where we are sending calls to, each of them is using a special prefix in the From header in order to manage the callerid. Some of them are not recognizing headers like P-Asserted-Identity or Remote-Party-ID, so for them we have to replace the From header with the corresponding prefix. The way we do - it's by using the uac_replace_from function, which works fine. The problem is when we are sending the same call to the failover trunk, and that trunk is using a different prefix. We have to call again the uac_replace_from function which is appending the new prefix to the existing one, something like: From: <sip:12345@domain.localsip:+12345@domain.local>;tag=as404cb50c I was trying to use uac_restore_from() function and call again uac_replace_from, but the result is the same. Is there any way to call the uac_restore_from function twice for the same call? Thank you.

3 3

Preprocessor ERROR! on upgrading to 5.4.0
by sagar malam 08 Aug '20

08 Aug '20

Hello , I was using Kamailio 5.3.3 . After upgrading to 5.4.0 i am getting below error with same cfg file which worked well with 5.3.3. 0(102523) ERROR: <core> [core/ppcfg.c:235]: pp_ifdef_level_error(): different number of preprocessor directives: 2 more #!if[n]def as #!endif What does this mean? Any clue to solve this? -- Thanks, Sagar

2 2

In memory ISP Info
by Ilie Soltanici 07 Aug '20

07 Aug '20

Hello, What will be the best way to save ISP Information in Kamailio memory? I have the following information for each ISP into the DB: *ISP Name | ISP IP | ISP Username | ISP Password | ISP Prefix* ISP1 | 1.1.1.1 | user | pass | 1234 This info it's not changed/updated too often, so I want to avoid querying the DB on every call and get this info directly from the memory. I was thinking to use htable for that, so I created a htable as an array for that and loaded all the information there manually, something like that: *kamcmd htable.sets isp ISP1[address] 1.1.1.1* * kamcmd htable.sets isp ISP1[username] user* * kamcmd htable.sets isp ISP1[password] pass* * kamcmd htable.sets isp ISP1[prefix] 1234 * *$sht( isp => ISP1[address]) == "1.1.1.1"* * $sht( isp => ISP1[username]) == "user" * * $sht( isp => ISP1[password]) == "pass" * * $sht( isp => ISP1[prefix]) == "1234" * So far so good, I'm able to get this information from the htable and use it whenever I need it. Now, the problem comes when I want to load all this information automatically from the DB. How can I setup htable module to get the array key (address, username...) from the database automatically on server restart, and append this array_key to the column value? I've tried this way: modparam("htable", "htable", "isp=>size=8;dbtable=isp;cols='trunk_name[address],trunk_address'") But it's not working unfortunately, getting back this error: You have an error in your SQL syntax; check the manual that corresponds to your MariaDB server version for the right syntax to use near '[address]' If I'm setting this way: modparam("htable", "htable", "isp=>size=8;dbtable=isp;cols='trunk_name,trunk_address'") and setting trunk_name value to: ISP1[address], then it's working fine, but in this case I will have to create by one column for each array key, which may be confusing and it's a lot of unnecessarily information into the DB. Is it possible to do that, or better to trigger a script on each restart which will load this information from the DB into htable? Thanks

2 2

Using wildcard certificates for Kamailio server
by Leonid Fainshtein 07 Aug '20

07 Aug '20

Hello, Is it permitted to use the wildcard TLS certificates for Kamailio server? In reality, it works (tested with v.5.4) but the RFC-5922 disables the wildcard certificates usage: "Implementations MUST match the values in their entirety: Implementations MUST NOT match suffixes. For example, "foo.example.com" does not match "example.com". Implementations MUST NOT match any form of wildcard, such as a leading "." or "*." with any other DNS label or sequence of labels. For example, "*.example.com" matches only "*.example.com" but not "foo.example.com". Similarly, ".example.com" matches only ".example.com", and does not match "foo.example.com". (Ref.:https://tools.ietf.org/html/rfc5922#section-7.2) To be honest, I don't understand why this restriction is good for... Is somebody aware of a newer RFC that removes this limitation? Best regards, Leonid Fainshtein

4 4

Deletion of Prometheus metrics
by Marat Gareev 07 Aug '20

07 Aug '20

Hello, I have confuse about this timeout: https://www.kamailio.org/docs/modules/devel/modules/xhttp_prom.html#xhttp_p… Is there reason for forced deletion of unused metrics? Let's imagine a situation: Zabbix reads metrics at a certain interval, unused metrics are deleted if they are not used for a period of time (xhttp_prom_timeout). I drew 2 cases when Zabbix will receive the same value, but we cannot say unambiguously whether these are new values or not. [image: xhttp_prom_timeout.jpg] Regards, Marat

2 4

Route to PSTN no works
by sip user 07 Aug '20

07 Aug '20

Hi, I have kamailio connect to Teams, and works form Asterisk -> Teams calls. For Teams -> Asterisk calls I'd worked using extension and register Asterisk with that extension. But I'd like to use direct routing with IP. In kamailio.cfg I activate define WITH_PSTN. I configured the IP and PORT for my PSTN. I'm using the default route[PSTN]: route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined xlog("L_INFO","PSTN ACTIVADO"); if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")){ xlog("L_INFO", "Error en el formato numerico!!"); return; } # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + #if (starts_with("$rU", "00")) { # strip(2); # prefix("+"); #} if (strempty($sel(cfg_get.pstn.gw_port))) { #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } else { xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } And in my request_route: remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { if($src_ip != "IP ASTERISK"){ xlog("L_INFO", "***********ROUTE PSTN***********"); route(PSTN); } else { xlog("L_INFO","LLamada desde $si con puerto $sp"); record_route_preset("FQND:5061;transport=tls", "IP KAMAILIO:5060"); add_rr_param(";r2=on"); route(DISPATCH); route(RELAY); } } But never see that the call go to PSTN route.. I'd made any wrong?? Thanks

3 2

Info: 2 Kamailio presentations soon live at ClueCon - 1:30pm CST (20:30 Berlin)
by Daniel-Constantin Mierla 06 Aug '20

06 Aug '20

Hello, quick info for those that want to watch the 2 presentations about Kamailio at ClueCon Deconstructed, starting in about 1 hour 40 minutes from now (or 13:30CST, 20:30Berlin/Paris/Madrid/...). One will be done by me and the 2nd by Fred Posner. A few more details are available at: - https://www.kamailio.org/w/2020/07/cluecon-deconstructed-2020/ The sessions should be live streamed on youtube - should appear in the freeswitch channel there at the time of start: - https://www.youtube.com/c/FreeSWITCH/ Of you can join the video conference room for a more interactive participation -- broadcast room listed at: - https://cluecon.cantina.video/#/home Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 2

Audio on Calls Stopping After 15 Minutes
by admin＠conradit.com 06 Aug '20

06 Aug '20

All, hi: I'm a Kamailio newbie! I've managed to set up a proxy server with the minimal configuration file found here: https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamail…. The issue that I'm having is that the audio on calls simply stop precisely after 15 minutes. It's always precisely at 15 minutes. The calls continue connected but without any audio. A summary about my set up: 1. This is an Amazon AWS EC2 server so it doesn't have a public IP directly attached to the interface. This makes me believe that the issue might be related to NAT. 2. I use Flowroute as the SIP trunk for these tests. 3. My phone system, for testing purposes, works on UDP port 6060 an the RTP ports are 20,000 to 30,000. 4. I am not using, for the time being, TLS nor SRTP. 5. My phone system is inside the same subnet as the Kamailio server. Questions: 1) Am I missing the NATHELPER module and if so how could we combine it to the minimal proxy configuration described above? Assuming that my Kamailio server uses local IP 10.80.10.5 and the PBX 10.80.10.6, both on UDP port 6060 for SIP. 2) May I simply combine the example file for nathelper.cfg as per this URL: https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/nathel… to the proxy example files, change IP addresses, ports, etc.? If not, how would one go about adding the NATHELPER module to the minimal proxy configuration? 3) Could it be some other issue that is causing the 15-minute interruption on audio? Thank you so much for your help!

2 1

Use-case does Kamailio support PBX - webrtc (web sockets)
by Johnny Ritzer 06 Aug '20

06 Aug '20

We have a PBX that doesn’t support websockets . But we want a PWA webaapp ctxSIP that uses sip.js to work . Due to its not SIP generic do we setup like this PBX <----> Kamailio <---->webapp Im looking for developers for android build via react native or ionic framework but not sure what solution/archtectiure Question. Does Kamailio register the SIP user/pass on the Kamailio server meaning it keeps it registered. And thus the ctxSIP client app just handles the pass thru for that connection. What about RTP do I need a separate handler for that. Tha nks

2 1

Info: Kamailio Getting Started Workshop at ClueCon - Live in 1 hour
by Daniel-Constantin Mierla 05 Aug '20

05 Aug '20

Hello, in about 1 hour I will do a quick (1 hour long) workshop at ClueCon deconstructed. It can be watched live on youtube -- should appear in their youtube channel: * https://www.youtube.com/channel/UC0i2xbniNHJQNfPPs_v4T0Q Or if you want to join the live video conference, which allows you to ask questions and interact with the participants, use an webrtc capable browser and go to: * https://cluecon.cantina.video/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 2

kamailio and rtpengine_manage() / rtpengine_delete()
by Karsten Horsmann 05 Aug '20

05 Aug '20

Hi list, I use rtpengine 7.2.1 branch with kamailio 5.2.x and 5.3.x in different setups. Due to some issues with my 5.2.x config I saw the "Total timed-out sessions via TIMEOUT" grow but "Total regular terminated sessions never" goes up from 0. That was an indicator for an non-existing rtpengine_manage() or rtpengine_delete() at the end of dialogs. Okay. Now I want to observe that a bit more. I use parallel forking to webrtc clients, that i guess will be the "more Offers" than Answers. Any hints on what numbers are important? rtpengine-ctl -ip 127.0.0.1:4444 list totals Total statistics (does not include current running sessions): Uptime of rtpengine :27726 seconds Total managed sessions :3435 Total rejected sessions :0 Total timed-out sessions via TIMEOUT :0 Total timed-out sessions via SILENT_TIMEOUT :0 Total timed-out sessions via FINAL_TIMEOUT :0 Total timed-out sessions via OFFER_TIMEOUT :0 Total regular terminated sessions :3435 Total forced terminated sessions :0 Total relayed packets :69127802 Total relayed packet errors :5063 Total number of streams with no relayed packets :128 Total number of 1-way streams :8 Average call duration :233.321876 Graphite interval statistics (last reported values to graphite): Total calls duration :0.000000 Min managed sessions :0 Max managed sessions :0 Min/Max/Avg offer processing delay :0.000000/0.000000/0.000000 sec Min/Max/Avg answer processing delay :0.000000/0.000000/0.000000 sec Min/Max/Avg delete processing delay :0.000000/0.000000/0.000000 sec Min/Max/Avg offer requests per second :0/0/0 per sec Min/Max/Avg answer requests per second :0/0/0 per sec Min/Max/Avg delete requests per second :0/0/0 per sec Control statistics: Proxy | Offer | Answer | Delete | Ping | List | Query | StartRec | StopRec | Errors | BlkDTMF | UnblkDTMF 127.0.0.1 | 3896 | 3757 | 3499 | 0 | 0 | 0 | 0 | 0 | 0 | 0 | 0 Kind Regards -- Mit freundlichen Grüßen *Karsten Horsmann*

2 1

Redundancy of Progress Call using Kamailio
by Amit Pal 04 Aug '20

04 Aug '20

Hi Team , Hope all is fine. My requirement : I have a customer who wants high availability for ringing calls and calls that in process of being set up. Can you suggest an alternate mechanism to achieve replication of transaction state. Can you guide how it can be achieved in Kamailio or is there a mechanism to get commercial support to achieve the desired objective. I am waiting your response. Thanks Regards Amit Pal

2 3

Kamailio Kubernetes AWS NLB UDP
by Ewgeny Berladin 04 Aug '20

04 Aug '20

Hi ! Does anyone have success Kamailio deployments via service on AWS EKS with UDP port exposing ? apiVersion: v1 kind: Service metadata: name: sip-proxy annotations: service.beta.kubernetes.io/aws-load-balancer-type: "nlb" spec: selector: app: kamailio-sip externalTrafficPolicy: Local type: LoadBalancer ports: - name: sip-up protocol: UDP port: 5060 As result we have this ERROR: ---- ------ ---- ---- ------- Normal EnsuringLoadBalancer 3m20s (x7 over 8m35s) service-controller Ensuring load balancer Warning SyncLoadBalancerFailed 3m20s (x7 over 8m35s) service-controller Error syncing load balancer: failed to ensure load balancer: Only TCP LoadBalancer is supported for AWS ELB As I know this is a BUG in AWS EKS. Maybe there’s some workarounds ? Kubernetes v 1.17 Thanks

2 1

ACC events missing to-tag when doing t_reply
by Julien Chavanton 04 Aug '20

04 Aug '20

Hi, considering this example : modparam("acc_json", "acc_missed_flag", 6) failure_route[test] { setflag(6); // MISSED_ACC, the failure will be recorded by ACC t_flush_flags(); // This will set the flags in the newly created transaction t_reply("500", "Service Unavailable"); } This does generate an ACC event, however the to-tag is not populated, my guess this is because the TM callback is done before the to-tag is generated. I wonder if there is a way to populate the to-tag immediately ? I will dig it further, but maybe there is a trick I am missing

2 2

Kamailio Dockerization
by Edward Romanenco 04 Aug '20

04 Aug '20

Hi, I am working on a project involving Kamailio dockerezation, which is meant to run alongside Freeswitch and RTPEngine containers, on the basis of a Docker-Compose file which is launched on top of a CentOS 7.7 host system. I was able to create and run the containers successfully, they are starting and listening to the correct ports, but for some unexplained reason - the incoming SIP traffic is not getting picked up by Kamailio. I can easily trace the traffic from the host, but when SSHing the container and running a test from within, no traffic goes by. I've used netcat to generate plain UDP traffic to the container, and it was logged into the Kamailio log files, but real-life traffic doesn't seem to work. I've tried moving to host mode (from bridge), but it didn't make any difference. All required firewall rules were opened obviously, I've also tried shutting the firewall off completely but it didn't help. Does anyone experienced anything similar while running Kamailio in Dockers, and could provide me a go-through on what steps did he take to fix it? EXCERPT FROM MY DOCKERFILE # Getting Kamailio source code from GIT RUN mkdir -p /usr/local/src/kamailio-5.3 WORKDIR /usr/local/src/kamailio-5.3 RUN git clone --depth 1 --no-single-branch https://github.com/kamailio/kamailio WORKDIR /usr/local/src/kamailio-5.3/kamailio RUN git checkout -b 5.3.2 # Compile the source code and install Kamailio RUN make include_modules="phonenum db_mysql xmlrpc http_async_client jansson auth_db nathelper websocket tls outbound topoh http_client" cfg && \ make all && make install # Default setting is to run Kamailio as user “kamailio” and group “kamailio” RUN adduser --quiet --system --group --disabled-password \ --shell /bin/false --gecos "Kamailio" \ --home /var/run/kamailio kamailio # To use init.d script for starting/stopping the Kamailio server COPY Init/kamailio /etc/init.d/ RUN chmod 755 /etc/init.d/kamailio COPY Default/kamailio /etc/default/ COPY kamailio.service /etc/systemd/system/ RUN mkdir -p /var/run/kamailio RUN chown kamailio:kamailio /var/run/kamailio COMPOSE (BRIDGE NETWORK VERSION) kamailioegress: build: kamailio_egress image: kamailioegress:latest container_name: kamailioegress restart: always environment: - DATABASE=kamailioe - SIP_DOMAIN=XXX - DBHOST=kamailiodb - DBROOTUSER=root - DBROOTPASS=XXX - PUBLIC_IPV4=XXX depends_on: - Kamailio-Base - kmdb - freeswitch - rtpengine expose: - "5060/udp" - "5060/tcp" ports: - "XXX:5060:5060/udp" networks: private-net: ipv4_address: "172.18.0.30" deploy: mode: replicated replicas: 1 restart_policy: condition: always delay: 5s max_attempts: 3 window: 120s networks: private-net: driver: bridge ipam: config: - subnet: 172.18.0.0/16 driver_opts: com.docker.network.bridge.name: wrtcpriv public-net: external: name: host COMPOSE (HOST MODE VERSION) kamailioegress: build: kamailio_egress image: kamailioegress:latest container_name: kamailioegress network_mode: host restart: always environment: - DATABASE=kamailioe - SIP_DOMAIN=XXX - DBHOST=172.18.0.10 - DBROOTUSER=root - DBROOTPASS=XXX - PUBLIC_IPV4=XXX - EGPORT=5060 - LINTE=ens224 - LINTI=ens192 - RTPENGINE=localhost depends_on: - Kamailio-Base - kmdb - freeswitch - rtpengine expose: - "5060/udp" ports: - "213.8.76.13:5060:5060/udp" deploy: mode: replicated replicas: 1 restart_policy: condition: always delay: 5s max_attempts: 3 window: 120s CONFIG FILE /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:0.0.0.0:LPORT advertise PUBLIC_IP:LPORT KAMAILIO-LOCALE #!define DBURL "mysql://root:XXX@DBHOST/kamailioe" #!substdef "!MY_DBURL!mysql://root:XXX@DBHOST/kamailioe!g" #!substdef "!RTPENGINE!MY_RTPENGINE!g" #!substdef "!SIP_DOMAIN!MY_SIP_DOMAIN!g" #!substdef "!PUBLIC_IP!MY_PUBLIC_IP!g" #!substdef "!PRIVATE_IP!MY_PRIVATE_IP!g" #!substdef "!LPORT!MY_LPORT!g" #!substdef "!LINT!MY_LINT!g" #!substdef "!HOMER_IP!10.1.0.100!g" #!substdef "!API_URL!http://localhost:3000/v1/mock!g" #!substdef "/CCODES/972|380/" #!substdef "/NUM_TRANSLATE_OUT_RE/\\\\+?(CCODES)([0-9]+)/" #!substdef "/NUM_TRANSLATE_IN_RE/0([0-9]+)/" FIREWALL RULES -bash-4.2# firewall-cmd --list-all public (active) target: default icmp-block-inversion: no interfaces: ens192 ens224 sources: 192.168.1.39 services: dhcpv6-client http https sip ssh ports: 9323/tcp 9323/udp protocols: masquerade: no forward-ports: source-ports: icmp-blocks: rich rules: rule family="ipv4" destination address="XXX" port port="5060" protocol="udp" accept rule family="ipv4" destination address="XXX" port port="5060" protocol="tcp" accept NETSTAT OUTPUT Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name udp 0 0 XXX:5060 0.0.0.0:* 22479/kamailio Edward

4 7

No more updates on Official Precise repositories
by Victor Seva 04 Aug '20

04 Aug '20

Hello, Ubuntu Precise is EOL since April 28, 2017 [0] and in order to support newer Ubuntu releases our deb build system will no longer produce newer packages for precise. Repositories will still be there but don't expect any updates. Cheers, Victor [0] https://wiki.ubuntu.com/Releases

2 3

[Help] Kamailio not receive bye requests
by Duy Phan 04 Aug '20

04 Aug '20

Hi guy! I use ec2 ubuntu on aws. But kamailio not receive bye requests. But when I install Kmailio in vultr is OK. can you help me?

4 3

search_append_body?
by Steve Wilkins 04 Aug '20

04 Aug '20

Hello, Does search_append_body require an "\r\n" in the attribute that is being added? If I don't use a "\r\n", it appears to look correct in the trace, however one provider says the attribute that was added was just a continuation of the prior attribute, which Wireshark does not show to be the case. Thank you,

2 1

topoh problem with bye from trunk
by mohsen khashei 04 Aug '20

04 Aug '20

Hi all. I use topoh module for topology hiding and this is my config for topoh: modparam("topoh", "mask_key", "my key") modparam("topoh", "mask_callid", 1) modparam("topoh", "callid_prefix", "!!:") modparam("topoh", "sanity_checks", 1) modparam("topoh", "mask_ip", "x.x.x.x") modparam("topoh", "uri_prefix_checks", 1) modparam("topoh", "uparam_name", "line") modparam("topoh", "uparam_prefix", "sr-") modparam("topoh", "vparam_name", "branch") modparam("topoh", "vparam_prefix", "z9hG4bKsr-") and also i use: modparam("dialog", "lreq_callee_headers", "TH: dlh\r\n") for dialog module. my calls come from asterisk and routed to the carrier(sip trunk) and vice versa, when i make a call from asterisk every thing works fine until the call hanged up from the carrier. at that modemt the kamailio send an 481 call/leg transaction does not exist. if call dropped bye the asterisk user everything works fine. how can i solve this?

3 3

Kamailio-IMS - Public Network Issue
by Pavithra Mohanraja 04 Aug '20

04 Aug '20

Hi, I have deployed kamailio-ims-5.3.5 as docker containers. I had private IP for containers. So when i try to establish a call from the public network , The via header is taken as container Private IP even if i use advertise address as public IP. So ACK is not reaching the pcscf from ue. Any idea can be provided for this issue. Kindly help. Thanks, Pavithra

2 1

Setting up MSFT Teams SBC <> SIP trunk but forwarding ACK and CANCEL messages
by Daniel van der Tang 02 Aug '20

02 Aug '20

Hello, I'm setting up a Kamailio MSFT Teams SBC that is connected to our SIP trunk provider. This server is directly connected to the Internet and not behind a NAT routing. It is successfully processing invites from both sides (MSFT Teams and SIP trunk). Unfortunately the ACKs and CANCEL messages are not relaying to Teams. In the sipdump i can't find these messages. Does anyone see what I am doing wrong? Below you can find my config #!KAMAILIO ####### Defined Values ######### #!define MULTIDOMAIN 0 # - 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 #!define FROM_TEAMS 11 #!define FROM_PBX 12 ######## Define Modules ########### #!define WITH_RTPENGINE #!define WITH_TLS #!define WITH_SIPDUMP #!define WITH_DISPATCH ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=2 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes alias=SBC_FQDN /* listen addresses */ listen=udp:PUBLIC_IP:5060 advertise SBC_FQDN:5060 #!ifdef WITH_TLS listen=tls:PUBLIC_IP:5061 advertise SBC_FQDN:5061 #!endif server_header= "Server: ABC SBC" user_agent_header= "User-Agent: ABC SBC" ###### TLS Enable ###### #!ifdef WITH_TLS enable_tls=yes #tcp_connect_timeout=1000 tcp_accept_no_cl=yes tcp_async = yes tcp_connection_lifetime=600 /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ ####### Modules Section ######## /* set paths to location of modules */ 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 "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_SIPDUMP loadmodule "sipdump.so" #!endif #!ifdef WITH_DISPATCH loadmodule "dispatcher.so" #!endif # ----------------- setting module-specific parameters --------------- #!ifdef WITH_RTPENGINE # ----- rtpengine params -----+ modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:12222") #!endif #!ifdef WITH_TLS # ----- tls params -----+ modparam("tls", "xavp_cfg", "tls") modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "connection_timeout", 10) modparam("tls", "ssl_release_buffers", 1) modparam("tls", "send_close_notify", 1) modparam("tls", "session_cache", 0) #!endif #!ifdef WITH_SIPDUMP modparam("sipdump", "enable", 1) #!endif #!ifdef WITH_DISPATCH #---------- dispatch modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 300) #!endif # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); #check who is the sender route(INITCHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # update $du to set the destination address for proxying #$du = "sip:" + $rd + ":9"; route(RELAY); exit; } route[INITCHECK] { if(from_uri =~ ".*microsoft.com") { setflag(FROM_TEAMS); $du = "sip:" + "PBX_IP"; route(HANDLE_RTP_FROM_TEAMS); } else if(from_uri =~ ".*" + "PBX_IP") { setflag(FROM_PBX); $du="sip:sip.pstnhub.microsoft.com;transport=tls"; route(HANDLE_RTP_FROM_PBX); } else { exit; } } #Manage RTP & transcoding comming from Teams to PBX route[HANDLE_RTP_FROM_TEAMS] { if (has_body("application/sdp")) { t_on_reply("PBX_REPLY_TO_TEAMS"); rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection ICE=remove"); record_route(); t_relay_to_udp("PBX_IP","5060"); } } #Manage RTP & transcoding comming from PBX to Teams route[HANDLE_RTP_FROM_PBX] { if (has_body("application/sdp")) { t_on_reply("TEAMS_REPLY_TO_PBX"); rtpengine_manage("SRTP codec-mask=all ICE=force codec-transcode=PCMA replace-origin replace-session-connection"); record_route_preset("SBC_FQDN:5061;transport=tls"); add_rr_param(";r2=on"); $rd = "sip.pstnhub.microsoft.com"; $td = "SBC_FQDN"; $fd = "SBC_FQDN"; #Set TLS SNI (server name & server id) $xavp(tls=>server_name) = "SBC_FQDN"; $xavp(tls=>server_id) = "SBC_FQDN"; t_relay(); } } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|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")) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; #Teams reINVITEs if(isflagset(FROM_TEAMS)) { t_relay_to_udp("PBX_IP","5060"); exit; } # 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 #set coresponding cert on transactions if($fd == "SBC_FQDN") { $xavp(tls=>server_name) = "SBC_FQDN"; $xavp(tls=>server_id) = "SBC_FQDN"; } } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); } #PBX On Reply onreply_route[PBX_REPLY_TO_TEAMS] { if (has_body("application/sdp")) { rtpengine_manage("SRTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=PUBLIC_IP"); } } #From Teams On Reply onreply_route[TEAMS_REPLY_TO_PBX] { if (has_body("application/sdp")) { rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=PUBLIC_IP"); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { if (t_is_canceled()) exit; } event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:SBC_FQDN:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); }

6 5

Manipulating SDP IP for Inbound Calls
by Edward Romanenco 02 Aug '20

02 Aug '20

Hey guys, I am working on a project involving Kamailio dockerezation, which is meant to run alongside Freeswitch and RTPEngine containers, on the basis of a Docker-Compose file which is launched on top of a CentOS 7.7 host system. Anyway, I would love to know if there is any way to manipulate / mask the IP addresses that are being appended to a status 183 response for an incoming invite. For some reason which I am trying to figure out in parallel, Freeswitch uses the local network bridge subnet instead of the defined external RTP IPs, and I was wondering - Can I manipulate them using Kamailio? I know that Mangler module can do it for outbound calls, but can I do the same for inbound? v=0. o=FreeSWITCH 1595974788 1595974789 IN IP4 172.18.0.40. s=FreeSWITCH. c=IN IP4 172.18.0.40. t=0 0. m=audio 45878 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16 Edward

2 1

lua_dostring sr vs KSR
by Julien Chavanton 31 Jul '20

31 Jul '20

lua_dostring("sr.pv.seti(\"$$var(var1)\", $avp(var1) < $avp(var3) and 1 or 0)"); results in error : kamailio app lual_dostring(): error from Lua: attempt to index global 'sr' (a nil value) replacing sr with KSR is working : lua_dostring("KSR.pv.seti(\"$$var(var1)\", $avp(var1) < $avp(var3) and 1 or 0)"); Shall we update the docs for https://kamailio.org/docs/modules/devel/modules/app_lua.html#app_lua.f.lua_… ?

2 4

Issue with kamctl after upgrading to 5.3.4
by Igor Potjevlesh 31 Jul '20

31 Jul '20

Hello there! I'm having trouble with kamctl since I upgraded to 5.3.4. The command is working fine with root user but not with another user. The problem seems to be that the command try to access to /var/run/kamailio/kamailio_rpc.fifo even if the file exists. I guess it's a permission problem. But I was thinking that FIFO is no more used. Only jsonrpcs. How can I fix that? Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

3 11

publishing hash values for download files of releases
by Daniel-Constantin Mierla 30 Jul '20

30 Jul '20

Hello, being discussed during the last devel meetings, I published the md5, sha1 and sha256 hash values for the tarballs with sources and i386 binaries we make available for download on kamailio.org on each release -- e.g., for 5.4.0: * https://www.kamailio.org/pub/kamailio/5.4.0/src/ * https://www.kamailio.org/pub/kamailio/5.4.0/bin/ Before making a more official announcement about it and adding to the download/install docs, I want to discuss a little bit here and get to the right solution to publish these hash values. For the moment I put them in a single file, adding -checksums.txt to the tarball name, listing inside all 3 hashes as computed by md5sum, sha1sum and sha256sum. That because I couldn't decide alone if there is sort of a standard on how to do it. Couple of projects I checked they just list the hash values on the html page with the link to download file. Others have dedicated files per hashing type, named like MD5SUMS, SHA1SUMS and SHA256SUMS, containing hash values for all downloadable files in the folder. Then, asterisk projects publishes 3 files, asterisk-VERSION.{md5,sha1,sha256}, corresponding to the tar.gz file they made available. Freeswitch publishes more than one archive file type, so it makes available files like freeswitch-VERSION.EXT.{md5,sha1,sha256}, where EXT can be tar.gz, tar.xz, zip ... My questions now. What kind of files with hash values people here are used with? Any variants that tends to be (or become the standard)? Any tools you are aware of for automatically checking the integrity with one of these specific hash files (like, if I have the tarball and the hashes file in the same folder and run it, it gives the ok/not-ok, without me having to do md5/sha1/sha256 manually and check "by eye" the values)? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 2

Kamailio v5.4.0 Released - new major version is out
by Daniel-Constantin Mierla 29 Jul '20

29 Jul '20

Kamailio v5.4.0 is out – it comes with 5 new modules and a considerable set of improvements touching again more than 100 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-4-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.4.0 brings more flexibility and optimizations across many components, enhancements to load balancer, stateless active calls tracking, STIR/SHAKEN support, Kafka connector, plenty of new variables and transformations, ability do print JSON-structured log messages ... just to list only a few here. Enjoy Kamailio v5.4.0! Thank you for flying Kamailio! Stay safe and healthy! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

The plan for releasing Kamailio v5.4.0
by Daniel-Constantin Mierla 29 Jul '20

29 Jul '20

Hello, we are considering to release the next major stable release, respectively 5.4.0, on Wednesday, July 29, 2020. The branch 5.4 was already created, several doc resources were already added (wiki pages with what is new, core/variables/transformations/rpcs/stats cookbooks, alphabetic indexes, ...), at least debian/ubuntu packages should be built nightly, ... If anyone is testing an upgrade from branch 5.3 to 5.4 and encounters changes that should be done in Kamailio config, add notes about at: * https://www.kamailio.org/wiki/install/upgrade/5.3.x-to-5.4.0 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 2

PHONENUM Module on Centos7
by Duarte Rocha 28 Jul '20

28 Jul '20

Greetings, I've trying to use the module PHONENUM on my Kamailio on CentOS 7. However, getting the required libphonenumber library hasn't been possible. Is there any CentOs7 repository for this ? What's the best way to work with this module on CentOS ? Best Regards,

2 1

Kamailio IMS with custom AVPs
by Gaston Siffert 28 Jul '20

28 Jul '20

Hello all, I am investigating on how to integrate few IMS modules in the Platform I am working on. At the moment the main drawback for me is to find a way to export data from the Diameter Answers. In the RFC 7155, it’s written that the AA-Answer could have a list of customisable fields. While looking at the implementation of the Rx_AAR (IMS QOS) in Kamailio, the module does not seems to export any of the fields from the AA-Answer except the return code. The same applies for the implementation of I_perform_location_information_request (IMS ICSCF) in Kamailio. Do you have any recommendation on how to export the other AVPs from the Diameter Answers ? Instead of IMS QOS and IMS ICSCF, I could use diameter_request_async (IMS Diameter Server) to make my own AAR and LIR, but is it the recommended practice ? https://tools.ietf.org/html/rfc7155#section-3.2 <https://tools.ietf.org/html/rfc7155#section-3.2> https://kamailio.org/docs/modules/5.0.x/modules/ims_qos.html#idp18760852 <https://kamailio.org/docs/modules/5.0.x/modules/ims_qos.html#idp18760852> https://kamailio.org/docs/modules/4.4.x/modules/ims_icscf.html#idp39432196 <https://kamailio.org/docs/modules/4.4.x/modules/ims_icscf.html#idp39432196> https://www.kamailio.org/docs/modules/devel/modules/ims_diameter_server.htm… <https://www.kamailio.org/docs/modules/devel/modules/ims_diameter_server.htm…> Best regards, Gaston Siffert

1 0

PHONENUM Module on Centos7
by Ilie Soltanici 28 Jul '20

28 Jul '20

Hello, If you want to use c++ library, then you can try to compile it by yourself: yum install cmake boost boost-devel boost-doc gtest gtest-devel protobuf protobuf-devel libicu libicu-devel $ cd libphonenumber/cpp $ mkdir build $ cd build $ cmake .. $ make $ ./libphonenumber_test cp -r libphonenumber.so* /usr/lib64/ cp -r libgeocoding.so* /usr/lib64/

1 0

ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction
by Denys Pozniak 28 Jul '20

28 Jul '20

Hello! Sometimes presence server stops sending SIP NOTIFY events with syslog output below (after Kamailio restarting everything is restored): Jul 27 17:07:28 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:28 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:29 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:29 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Where is a possible source of the problem? Kamailio 5.1.2 with MySQL modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "expires_offset", 0) modparam("presence", "max_expires", 1800) modparam("presence", "db_update_period", 30) modparam("presence", "clean_period", 180) modparam("presence", "send_fast_notify", 1) modparam("presence", "pres_htable_size", 32) modparam("presence", "subs_htable_size", 32) modparam("presence", "publ_cache", 0) modparam("presence", "notifier_processes", 0) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "integrated_xcap_server", 0) modparam("presence_xml", "disable_presence", 1) modparam("presence_xml", "disable_winfo", 1) modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence_dialoginfo", "force_dummy_dialog", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_match_mode", 1) modparam("kazoo", "node_hostname", NODE_HOSTNAME) modparam("kazoo", "amqp_connection", "AMQP_URL") modparam("kazoo", "pua_mode", 1) modparam("kazoo", "db_url", DBURL) modparam("kazoo", "amqp_consumer_processes", 10) modparam("kazoo", "amqp_max_channels", 50) -- BR, Denys Pozniak

2 1

Kamailio CentOS 7 repos broken
by Patrick Wakano 28 Jul '20

28 Jul '20

Hi list, Hope you are all good! I've been trying to install kamailio 5.2.7 from the centos 7 repo, but it is failing in both repos..... With the http://rpm.kamailio.org/centos/ repo it fails with: Error: Package: kamailio-sipcapture-daemon-config-5.2.7-0.el7.centos.x86_64 (kamailio-5.2) Requires: kamailio-sipcapture = 5.2.7 With the http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… repo it fails with: kamailio-5.2.7-7.el7.centos.x8 FAILED =================== ] 579 kB/s | 12 MB 00:00:04 ETA https://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.2.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.2.7-7.el7.centos.x86_64: [Errno 256] No more mirrors to try. Would someone be able to have a look at that? Thank you! Patrick Wakano

2 1

Registrar - registered match_received
by Ilie Soltanici 27 Jul '20

27 Jul '20

Hello, Yes, i have it there but the variable name was mismatching with the module variable name: $xavp(regcfg=>match_received) = $su; modparam("registrar", "xavp_cfg", "reg") It's all good now. Thanks for your help.

1 0

Registrar - registered match_received
by Ilie Soltanici 27 Jul '20

27 Jul '20

Hello, Trying to check if an extension it's already registered with the same received path. if (registered("location","$fu",2)) { sl_send_reply("200","OK"); exit; }; But, it's not working for me, getting true even if the received path it's different: This is what I'm getting from Kamailio: [root@dev ilie.soltanici]# /usr/local/sbin/kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 1024 AoRs: { Info: { AoR: 101 HashID: -1698832128 Contacts: { Contact: { Address: sip:101@192.168.1.10:12767;ob Expires: 2452 Q: 1.000000 Call-ID: 3fa6c9f7db404e8e93a9664c3634ddf7 CSeq: 31114 User-Agent: MicroSIP/3.19.31 Received: sip: 192.168.1.10:12767 Path: [not set] State: CS_SYNC Flags: 0 CFlags: 12288 Socket: udp: 192.168.1.1:5060 Methods: 8159 Ruid: uloc-b-5f1eea5d-57b0-1 Instance: [not set] Reg-Id: 0 Server-Id: 11 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1595862742 Last-Modified: 1595861637 } } This is the SIP Trace: 2020/07/27 16:17:12.214151 192.168.1.10:12610 -> 192.168.1.1:5060 OPTIONS sip:dev.local SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-91-1-0 From: "Test Account" <sip:101@dev.local>;tag=91SIPpTag001 To: sip:dev.local Call-ID: 1-91(a)192.168.1.10 User-Agent: SIPP CSeq: 1 OPTIONS Content-Length: 0 As you can see the source port it's different 12767/12610, but the registered function is still getting true. Thank you.

2 1

Install 5.3 on Ubuntu 20.04 With MySQL 8
by Mark Boyce 27 Jul '20

27 Jul '20

Hi all This is installing 5.3 for Bionic (18.04) on Focal (20.04) but as I know you’re in the throws of building 5.4 thought I'd throw this out there in case it changes anything; Installing kamailio from; deb http://deb.kamailio.org/kamailio53 bionic main The following packages have unmet dependencies: kamailio : Depends: libreadline7 (>= 6.0) but it is not installable kamailio-mysql-modules : Depends: libmysqlclient20 (>= 5.7.11) but it is not installable kamailio-redis-modules : Depends: libhiredis0.13 (>= 0.13.1) but it is not installable libreadline7 -> libreadline8 (v 8.0-4) libmysqlclient20 -> libmysqlclient21 (v 8.0.21-1ubuntu20.04) libhiredis0.13 -> libhiredis-dev (v 0.14.0-6). Ubuntu Server 20.04 LTS (focal) MySQL 8 (From MySQL Repo) Redis 6 from latest stable Best regards Mark -- Mark Boyce Dark Origins Ltd

2 6

Using GeoIP2 AS Lookup?
by Mark Boyce 27 Jul '20

27 Jul '20

Hi all Just looking at the latest GeoIP2 MaxMind databases (now requires registration, but still free) and noticed that they also include the AS (ISP) lookup one in the free offering. Wondering if this is another way to facilitate better security for users on dynamic IP. Typically working from home these days. So, rather than just limiting an end device to a country we could limit it to a particular ISP within that country. Has anyone tried this? Have I missed a reason why this wouldn’t help? Admin overhead not worth it? Thoughts? Best regards Mark -- Mark Boyce Dark Origins Ltd

2 5

kamailio 5.3.5 - none local route header
by Jonathan Hunter 27 Jul '20

27 Jul '20

Hi Guys, Sorry if this is too generic or silly a question but I am testing kamailio 5.3.5 within Azure and its behind NAT. I am having an issue with passing an ACK from our internal side back out to the carrier side and it appears having run loose_route a none local route-header is consistently being removed by kamailio at this point. ie Route: <sip:CARRIER_IP;lr> Is this normal or configuration related? As I am unsure within what scenario I would remove a route header which doesnt contain the local Proxy IP. Please can someone advise? Many thanks Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

2 3

Redis error: Server closed the connection / nbd_redis / kamailio 5.3.5
by Karsten Horsmann 27 Jul '20

27 Jul '20

Hi everyone, I get some strange errors from nbd_redis but my setup is working. kamailio 5.3.5 on CentOS 7.7.1908. Redis-Server is an haproxy frontend that pushed the stuff to an redis-sentinel cluster (with and really old version redis-3.2.1-2.el7.remi.x86_64). Any ideas on that? Jul 22 13:42:52 siptrunk5 kamailio[112377]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 13:43:34 siptrunk5 kamailio[112359]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 13:43:34 siptrunk5 kamailio[112378]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 13:44:33 siptrunk5 kamailio[112363]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 13:45:03 siptrunk5 kamailio[112362]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 14:00:01 siptrunk5 kamailio[112381]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 14:00:01 siptrunk5 kamailio[112365]: ERROR: {24519 CANCEL 59644df6-68ac-4dd6-9fa5-1a9c2eda3b59} ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection # grep redis /etc/kamailio/kamailio.cfg loadmodule "ndb_redis.so" loadmodule "topos_redis.so" modparam("ndb_redis", "server", "name=CFG_REDIS_NAME;addr=CFG_REDIS_ADDR;port=6379;db=CFG_REDIS_DB") modparam("ndb_redis", "connect_timeout", 1500) # default 1000ms/1sec modparam("ndb_redis", "cmd_timeout", 1500) # default 1000ms/1sec modparam("topos", "storage", "redis") modparam("topos_redis", "serverid", "CFG_REDIS_NAME") -- Cheers *Karsten Horsmann*

3 2

failed to lock the memory pages (disable swap) WARNING
by Joel Serrano 27 Jul '20

27 Jul '20

Hi all, Can anyone give me some info on what this warning means: Jul 7 17:02:29 csbc02 csbc[16006]: WARNING: <core> [core/daemonize.c:596]: mem_lock_pages(): failed to lock the memory pages (disable swap): Cannot allocate memory [12] Yes, if I set mlock_pages=no I don't see the warning, but I'm trying to understand why can't it allocate with mlock_pages=yes. Joel.

4 4

Kamailio Installation Steps
by Abdul Rasheed 26 Jul '20

26 Jul '20

Hello All , I am trying to setup Kamailio on Debian . May i know which document should be followed for installing the latest version of Kamailio on Debian 10 Buster. ? Thanks and Regards Abdul Rasheed

2 2

Dialplan: using $avp("key") on subst rule with regexp capturing group
by João Vitor Arruda 24 Jul '20

24 Jul '20

Hello, I'm trying to use the dialplan with $avp("key") on match_exp and subst_exp but i'm getting an error message when there is a regex capturing group on subst_exp to be used on repl_exp This is the config used: modparam("dialplan", "match_dynamic", 1) $avp(s:PSTNAC) = "0"; Dialplan entry: dpid = 1 match_op = 1 match_exp = ^$avp(s:PSTNAC) match_len = 0 subst_exp = ^$avp(s:PSTNAC)(.*) repl_exp = 9\1 After executing dp_translate while the dialplan is properly matched the following error message occurs and the repl_exp didn't work as expected: ERROR: dialplan [dp_repl.c:422]: rule_translate(): illegal access to 1-th subexpr of subst expr (max 0) I've tried some other dialplan entries like: dpid = 1 match_op = 1 match_exp = ^$(avp(s:PSTNAC)[*]) match_len = 0 subst_exp = ^$(avp(s:PSTNAC)[*])(.*) repl_exp = 9\1 But that resulted in the same error message and no replacement either. Any suggestion on what I'm doing wrong here? Thanks, Joao Arruda

1 2

how to achive destination number based routing before falling back to dispatcher-module
by Christoph Russow 24 Jul '20

24 Jul '20

Hi all, im fairly new to Kamailio and currently setting up our new sip-server systems so, hello everyone! right now i have a Kamailio SBC which routes all calls using the dispatcher module to our internally setup application servers (freeswitch). i just got the idea that we might want specific routing based on destination phone number in addition to simple loadbalancing. can anyone point me in the right direction where i should look for something i can place before the dispatcher routing to route calls based on some entries in a database like destination number XYZ goes to server A, number ABC goes to server Z and so on. Or is there a way to select the dispatcher group id from the database based on destination number? best regards Christoph

4 9

HEP siptrace HEPv3 truncated and ignored by heplify-server (Homer 7)
by José Lopes 24 Jul '20

24 Jul '20

Hello all, I am using the siptrace module on Kamailio 5.3.5 and I am trying to monitor the SIP over websockets traffic with Homer 7.8.1 (heplify-server). I notice a strange behavior, some packets are not presented on Homer. I checked heplify-server and i notice the next warning: heplify-server: 2020/07/22 18:41:09.425513 decoder.go:103: WARN HEP packet length is 8192 but should be 11663 So I think the packets are truncated by 8192 (packet size), so they are ignored by heplify-server. I have done a tcpdump on the Homer machine and have the next output, where I can see some packets with size greater than 8192. 19:58:29.788927 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, bad length 11658 > 1472 19:58:29.793539 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 626 19:58:29.810733 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 579 19:58:29.811121 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 579 19:58:29.838824 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, bad length 11953 > 1472 19:58:29.852286 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 473 19:58:29.865388 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, bad length 3521 > 1472 19:58:29.875311 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 562 Can you guide me to identify where the packets are truncated? Os melhores cumprimentos / Best regards, *José Lopes* Research and Development Technician Phone: +351 256 370 980 Email: jose.lopes(a)itcenter.com

2 2

new in upcoming Kamailio 5.4.0
by Henning Westerholt 23 Jul '20

23 Jul '20

Hello, I've updated the "New in version 5.4.0" document in our wiki. It is again an impressive list! * In this development period (started October 2019) almost 1500 commits have been added to the repository * 5 new modules have been added (pv_headers, kafka, secsipid, systemdops, dlgs) * Over 70 modules have been extended, many of them substantially (dispatcher, pv, tls, presence etc..) * Countless bugfixes and documentation extensions/fixes have been added as well * Many refactorings, extensions and architectural improvements were done in Kamailio core, tooling and modules You'll find the full details at the usual place: https://www.kamailio.org/wiki/features/new-in-5.4.x Many thanks to everybody that contributed to the Kamailio project. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

5 15

TAS recommendation
by BALL SUN 23 Jul '20

23 Jul '20

Hi We recently setup Kamailio as the IMS core in our test lab, and we would like to test scenarios with TAS, is there any recommendation? Does Kamailio able to setup as a SIP AS? if yes, any online instruction on how we can make it works? Thanks RBK

1 0

Dispatcher does not go to 2nd server
by Aristeidis Tsitras 22 Jul '20

22 Jul '20

I have a Kamailio and 2 asterisk servers. All users are created in both of the asterisk servers. I am forwarding the registration to asterisk. The problem is that it is always used on only one server from the list. Even if one goes to shutdown, then there is not any registration sent to the available server. Even if *some *of the extensions can be seen registered in both of the asterisk's, if the secondary goes down, then there are no services for the phones. I am attaching the kamailio.cfg. My dispatch list is: 1 sip:192.168.0.100:5080 0 0 maxload=20 2 sip:192.168.0.101:5080 0 0 maxload=20 In both of the asterisk servers i am using sip.conf to create users and s sip trunk for the Kamailio. Nothing special about it. I am looking to find what is wrong with my config and i cannot loadbalance/failover to the asterisk servers.

6 12

Siremis - PassThru Authentication
by Adam - UC-Reseller 21 Jul '20

21 Jul '20

I have recently installed Siremis And have set up a domain pointing to it, how do I configure it so registration requests to this domain are passed thru to the actual PBX Adam

2 1

P-Asserted-Identity in SIP INVITE from Kamailio IMS to PSTN (MGCF)
by BALL SUN 21 Jul '20

21 Jul '20

Hi We have a question. We are preparing a setup to bridge the SIP call from Kamailio IMS to PSTN, but we are puzzling how we can present the P-Asserted-Identity in the SIP INVITE. By default, the PAI is in sip:user@domain, but we believed we should use<tel> format in the PAI when bridge to PSTN. question is 1. should we have two PAI headers?, or 2. should we paste <tel> format in PAI User Part and domain in PAI Host Part?, or 3. Should we only use one PAI header with tel format? In our test setup, we are using FHoSS, and the IMPU sip:user@domain have below entry in "IMPUs from Implicit-Set" - sip:user@domain - tel:number Question is how we can extract the number from the HSS and presented in PAI? - RBK

1 1

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would like tohave: -kamailio (5.3 from deb repo) in front to load balance. Keep registrations. IP: 192.168.1.100 -Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to Kamailio in sip.conf -Asterisk-2 (16 LTS). IP: 192.168.1.102. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the Kamailio setup and users can register on Kamailio following the basic setup. I would like advice on setup and configuration, to integrate Kamailio with Asterisk boxes Some advice please? Sincerely yours, John

2 2

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would have: -kamailio in front to load balance. Keep registrations. IP: 192.168.1.100 -Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to Kamailio in sip.conf -Asterisk-2 (16. IP: 192.168.1.102. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3. 0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the Kamailio setup and users can register on Kamailio following the basic setup. I would like advice on setup and configuration, to integrate Kamailio with Asterisk boxes

1 0

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would have: -kamailio in front to load balance. Keep registrations. IP: 192.168.1.218 -Asterisk _1. IP: 192.169.1.232.It will serve as media server for IVR, VM, etc -Asterisk-2. IP: 192.168.1.233. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the I would like advice on setup and configuration,

1 0

Kamailio & Freeswitch as media and SBC
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup Kamailio with Freeswitch in a Debian machine as in the http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc, but Kamailio fails to start and gives a ton of errors. Before of the alterations Kamailio and Siremis start correctly and i can easily create users through Siremis. After the changes to include Freeswitch's blocks, it fails to start. All run in the same machine with IP 192.168.1.213. I am attaching kamailio.cfg (named as test1.txt) if someone could spot the error, please. Also i am attaching the kamailio.cfg(Original1.txt) when it was working fine with Siremis.

3 8

SIP Security handling in Kamailio IMS
by BALL SUN 20 Jul '20

20 Jul '20

Hi During the testing with Kamailio IMS, we found that SUBSCRIBE and NOTIFY message flow between UE and P-CSCF is shown below. SUBSCRIBE and its response: 192.168.1.102:9101 (port-c) -> 192.168.2.66:6101 (port-s) NOTIFY: 192.168.2.66:5101 (port-c) -> 192.168.1.102:9100 (port-s) based on the sip security, is it the expected behavior that they are using port-c and port-s? The reason why we asked is because we found that the NOTIFY did not properly propagate to the UE due to different port addresses. Does anyone have this experience? and how we can resolve it? - RBK

2 2

the list with sip compact header names
by Daniel-Constantin Mierla 17 Jul '20

17 Jul '20

Hello, anyone knowing by heart or a place listing all compact header names? I found: - https://www.cs.columbia.edu/sip/compact.html but it says: Last updated 01/10/2008 Maybe newer RFC introduced other compact header names. Or, in case you are aware of some other compact names not listed there, just say it here, as I am working a bit on updating the parser on this scope. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 2

CommCon 2020 Online - Now presentation about Kamailio
by Daniel-Constantin Mierla 16 Jul '20

16 Jul '20

Hello, being a bit late with this announcement, as it started a couple of minutes ago, but if anyone wants to watch the live presentation about Kamailio by Fred Posner at CommCon 2020 Online, here is the link: - https://www.youtube.com/watch?v=FGZF7dYdRng 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

Reset Prometheus counter once
by Marat Gareev 16 Jul '20

16 Jul '20

Hello. Are there any ways to reset the Prometheus counter once? I can do something like this, but counter will be reset on each request loadmodule "xhttp_prom.so" > modparam("xhttp_prom", "prom_counter", "name=mycounter;") > route { > prom_counter_reset("mycounter"); > } Otherwise, I will not receive the counter until a special and rare event occurs. Regards, Marat

3 4

RPC Socket stops answering after a few time.
by David VILLAUME 16 Jul '20

16 Jul '20

Hello, I’m having problem with RPC socket that stops answering after approximatively one or two hours. Version is 5.2.7 My principal host with exact same version and configuration has no problem, only the backup node (Keepalived + DMQ) encounters the problem. I could catch this log about the problem, but didn’t find any reason for this : Jun 18 10:53:33 sbc-kam-partner-b kamailio[83170]: ERROR: ctl [binrpc_run.c:600]: rpc_send(): ERROR: binrprc: rpc_send: send failed My configuration about rpc : loadmodule "jsonrpcs.so" modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") modparam("ctl", "binrpc_max_body_size", 1024) Do you have any idea to troubleshoot this ? Regards, David

2 13

Routing specific number from kamailio to asterisk
by Hendra 834 16 Jul '20

16 Jul '20

Hi, Im new in VoIP, especially Kamailio. I want to connect my kamailio with asterisk by routing the call coming from kamailio to asterisk for specific number, also from asterisk to kamailio. For example, if kamailio client call ext number 0123, the will be routed to asterisk, if kamailio client call 0233 the call will not be routed to external proxy. And also how can I receive call form asterisk to kamailio ? Is there any real example, ? because I dont understand what to change in documentation example. Thanks

2 1

Is it possible to load both dialog and ims_dialog modules in the same configuration?
by Sumesh Soman 16 Jul '20

16 Jul '20

Hi, Is it possible to load and use both dialog and ims_dialog modules in the same server ? Even though ims_dialog docs says ims_dialog extends dialog module, I do see that some of the dialog module features are not present in ims_dialog. Example, wait_ack , track_cseq_updates are not present in ims_dialog. I have requirements to use ims_dialog mainly for ims_charging module and dialog module with wait_ack and track_cseq_updates parameters are enabled. -- Regards, Sumesh S.

1 0

5.4 compiler warning
by Juha Heinanen 14 Jul '20

14 Jul '20

Got this when building 5.4; CC (gcc) [sip-proxy] core/dset.o core/dset.c: In function 'uri_trim_rcv_alias': core/dset.c:1042:6: warning: variable 'proto' set but not used [-Wunused-but-set-variable] int proto; ^~~~~ -- Juha

2 2

Re: [SR-Users] xHTTP_PROM with older Kamailio
by Marat Gareev 13 Jul '20

13 Jul '20

Hello, Karsten, Thanks for the link. I will definitely examine it.

1 0

No of registartions
by Pavithra Mohanraja 13 Jul '20

13 Jul '20

Hi, Is there any way to query number of registrations happened in kamailio IMS. Kindly reply . Thanks, Pavithra

2 1

subsequent ds_select_dst with different groups
by David VILLAUME 13 Jul '20

13 Jul '20

Hello, Is it possible to use the following logic : Route() ds_select_dst (1,2) failure_route(1) ds_next_dst(first_dispatcher_ secondgateway) failure_route(newdispatcher_group) ds_select_dst (2,2) failure_route(secondispatcher_secondgateway) ds_next_dst() Thanks in advance. Regards, David

4 5

Compact SIP
by Abdoul Osséni 13 Jul '20

13 Jul '20

Hello, I use Kamailio and Asterisk. SIP Compact header is activated by ASTERISK ( https://www.cs.columbia.edu/sip/compact.html). I am trying to disable the SIP Compact header on KAMAILIO. Is it possible? Regards Abdoul OSSENI

3 4

Re: [SR-Users] xHTTP_PROM with older Kamailio
by Marat Gareev 12 Jul '20

12 Jul '20

I'm sorry. I mean *xhttp_prom.c* file instead of *xhttp_mod.c*. >

1 0

xHTTP_PROM with older Kamailio
by Marat Gareev 11 Jul '20

11 Jul '20

Hello, I have a customized Kamailio (kamailio 5.1.1 (x86_64/linux)), xHTTP_PROM <https://www.kamailio.org/docs/modules/devel/modules/xhttp_prom.html> was released in the 5.3.0 <https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/> version. I'm trying to import the latest version of xhttp_prom to older Kamailio, but something goes wrong. This is simple config for test: log_stderror=yes > listen=5060 > debug=2 > mpath="/home/devel/build_dir/build/lib64/kamailio/modules" > loadmodule "tm.so" > loadmodule "sl.so" > loadmodule "xlog.so" > loadmodule "pv.so" > disable_tcp=no > tcp_accept_no_cl=yes > loadmodule "xhttp.so" > loadmodule "xhttp_prom.so" > modparam("xhttp_prom", "xhttp_prom_stats", "all") > route { > sl_send_reply("200", "OK"); > exit; > } > event_route[xhttp:request] { > if (prom_check_uri()) { > prom_dispatch(); > } else { > xhttp_reply("404", "Not Found", "text/plain", "Wrong URL $hu\n"); > } > } then I send HTTP request $ curl http://localhost:5060/metrics but get error: > 6(110968) CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 11 > 0(110962) ALERT: <core> [main.c:746]: handle_sigs(): child process 110967 > exited by a signal 11 > 0(110962) ALERT: <core> [main.c:749]: handle_sigs(): core was generated > 0(110962) INFO: <core> [main.c:771]: handle_sigs(): terminating due to > SIGCHLD note: for http://localhost:5060/test request I get the expected message "Wrong URL /test". Backtrace for killed process > (gdb) c > Continuing. > Program received signal SIGSEGV, Segmentation fault. > 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > 224 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) > (gdb) bt > #0 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > #1 0x00007fefff3ff0e2 in futex_get (lock=0x0) at > ../../core/mem/../futexlock.h:99 > #2 0x00007fefff4130bf in prom_metric_list_print (ctx=0x7fefff628da0 > <ctx>) at prom_metric.c:1319 > #3 0x00007fefff416468 in prom_stats_get (ctx=0x7fefff628da0 <ctx>, > stat=0x7fefff6281f0 <xhttp_prom_stats>) at prom.c:178 > #4 0x00007fefff3d4d10 in prom_send (ctx=0x7fefff628da0 <ctx>) at > xhttp_prom.c:361 > #5 0x00007fefff3d5faf in ki_xhttp_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:418 > #6 0x00007fefff3d5feb in w_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:428 > #7 0x0000000000461dbe in do_action (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1067 > #8 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1565 > #9 0x0000000000461d2d in do_action (h=0x7ffecb730e00, a=0x7ff0007ca628, > msg=0x7ffecb730ee0) at core/action.c:1058 > #10 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c76f8, > msg=0x7ffecb730ee0) at core/action.c:1565 > #11 0x00007fefff62f649 in xhttp_process_request (orig_msg=0x7ff0007ebbd8, > new_buf=0x7ff0007cc280 "GET /metrics HTTP/1.1\r\nVia: SIP/2.0/TCP > 127.0.0.1:37000\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", new_len=119) at xhttp_mod.c:298 > #12 0x00007fefff630d78 in xhttp_handler (msg=0x7ff0007ebbd8) at > xhttp_mod.c:385 > #13 0x0000000000520141 in nonsip_msg_run_hooks (msg=0x7ff0007ebbd8) at > core/nonsip_hooks.c:111 > #14 0x000000000058cef1 in receive_msg (buf=0x7feffee65bb0 "GET /metrics > HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: localhost:5060\r\nAccept: > */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0) > at core/receive.c:216 > #15 0x000000000061233f in receive_tcp_msg (tcpbuf=0x7feffee65bb0 "GET > /metrics HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0, > con=0x7feffee658b8) at core/tcp_read.c:1379 > #16 0x000000000061479e in tcp_read_req (con=0x7feffee658b8, > bytes_read=0x7ffecb731c7c, read_flags=0x7ffecb731c78) at > core/tcp_read.c:1611 > #17 0x00000000006174e9 in handle_io (fm=0x7ff0007d9720, events=1, idx=-1) > at core/tcp_read.c:1785 > #18 0x00000000006069b4 in io_wait_loop_epoll (h=0xaa76c0 <io_w>, t=2, > repeat=0) at core/io_wait.h:1065 > #19 0x00000000006194ff in tcp_receive_loop (unix_sock=14) at > core/tcp_read.c:1955 > #20 0x00000000004f1a43 in tcp_init_children () at core/tcp_main.c:4853 > #21 0x0000000000423f99 in main_loop () at main.c:1716 > #22 0x000000000042a5ee in main (argc=9, argv=0x7ffecb7323b8) at main.c:2646 > (gdb) > (gdb) p 'prom_metric.c'::prom_lock > $4 = (gen_lock_t *) 0x0 If I understood correctly, the problem is in an uninitialized variable static gen_lock_t *prom_lock = NULL; /* Lock to protect Prometheus metrics. */ So, is it possible to import xHTTP_PROM to older Kamailio? I will be glad for any advice. Thanks.

1 0

loose_route failing with topos ?
by Mark Boyce 10 Jul '20

10 Jul '20

Hi all Just enabled topos with redis backend (topos/topos_redis/ndb_redis) The interesting bits of the calls are going from kamailio (5.3.5) to Asterisk with topos enabled for that leg All seems to be ok until the far end (Asterisk) sends a BYE. At which point kamailo passes message to WITHINDLG, in loose_route comes up false (-1) and the BYE gets missed, falling through to the “404 Not here” at the end of WITHINDLG. I’m guessing that loose_route should be able to “see inside” topos, or should I be throwing loose_route away when using topos? Happy to upgrade to 5.4 and do diags if it appears to be a bug. Best regards Mark -- Mark Boyce Dark Origins Ltd

3 4

rtcp-mux is missing in the OK for WebRTC-Calls
by Benjamin Flügel | vio:networks 10 Jul '20

10 Jul '20

Hey guys, I'm trying to configure a Kamailio to work with a browser softphone based on SIPJS using WebRTC. So far it works great on Firefox but have a specific problem with chrome, when I want to make call from the softphone to another extension. After anwsering the call Chrome/the softphone sends a BYE immediately, because this line "a=rtcp-mux" is missing in the OK. The Kamailio is a proxy. Behind the Kamailio there is an Asterisk, which is responsible for the pbx-features. Those are my rtpengine Flags for the Invite: rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux DTLS=off SDES-on ICE=remove RTP/AVP And those are the flags for the response, in this case the OK: rtpengine_manage: replace-origin replace-session-connection rtcp-mux-offer rtcp-mux-accept generate-mid DTLS=off SDES-on ICE=force RTP/SAVPF direction=internal direction=external loop-protect It seems that the Kamailio ignores ths "rtcp-mux-offer rtcp-mux-accept" in the response. Can you help me get it to work? Here is the SIP-Dialog. The call is from extension 201 to the extension 2. INVITE: INVITE sip:2@mydomain SIP/2.0 Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;branch=z9hG4bK5104892 To: <sip:2@mydomain.com> From: <sip:201@mydomain.com>;tag=aqplr71k05 CSeq: 2 INVITE Call-ID: sde09v49f8gqtt59oddn Max-Forwards: 70 Proxy-Authorization: Digest algorithm=MD5, username="201", realm="mydomain.com", nonce="sfdsdfdsfsdfdsfsdfaseqww", uri="sip:2@mydomain.com", response="81rewtrega23423r" Contact: <sip:kf6iduon@g4n0lpfirgpv.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIPJS Content-Type: application/sdp Content-Length: 1916 v=0 o=- 5826889093965459811 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO m=audio 54274 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 mycomputerIP a=rtcp:9 IN IP4 0.0.0.0 a=candidate:1577908739 1 udp 2113937151 efd6d297-d186-4c07-87d8-933e5846a82d.local 54274 typ host generation 0 network-cost 999 a=candidate:842163049 1 udp 1677729535 mycomputerIP 54274 typ srflx raddr 0.0.0.0 rport 0 generation 0 network-cost 999 a=ice-ufrag:Ktxl a=ice-pwd:x5xDUb41GeOXAcHNQlra4yUN a=ice-options:trickle a=fingerprint:sha-256 2C:7C:32:AF:34:A3:9D:AE:C7:FD:92:68:DD:D8:AB:82:DB:F0:32:51:14:97:20:60:66:5C:2F:CF:B7:98:B8:A8 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:2376769177 cname:RgPkYyxDrEJTpmO2 a=ssrc:2376769177 msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=ssrc:2376769177 mslabel:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO a=ssrc:2376769177 label:dc187be3-4cd0-4129-98e4-f804cd8d2c94 RINGING: SIP/2.0 180 Ringing Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internalIP;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:externalIP;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Length: 0 OK: SIP/2.0 200 OK Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internal-ip;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:external-ip;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Type: application/sdp Content-Length: 806 v=0 o=root 882840402 882840402 IN IP4 externalIP s=Asterisk PBX 11.22.0 c=IN IP4 externalIP t=0 0 a=rtpengine:4d633e3022f5 m=audio 16416 RTP/SAVPF 111 9 8 0 126 a=maxptime:60 a=mid:0 a=rtpmap:111 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0 a=fmtp:126 0-16 a=sendrecv a=rtcp:16417 a=setup:active a=fingerprint:sha-1 B1:67:4B:B8:47:89:E8:49:CD:DD:F8:FF:41:5C:83:72:D9:DE:4D:45 a=ptime:20 a=ice-ufrag:WBZN8m7c a=ice-pwd:mKVR6Jv0G0pvrTv7OfEm2wPW9Q a=ice-options:trickle a=candidate:3kddB3zBUV2n2jt0 1 UDP 2130706431 externalIP 16416 typ host a=candidate:3kddB3zBUV2n2jt0 2 UDP 2130706430 externalIP 16417 typ host a=end-of-candidates I hope, you can help me with this. If you have further question, I will try to answer them as best I can. Greetings, Ben

2 1

Development is open for future release of v5.5.x series
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 was created, therefore the master branch is open for adding new features, to be part of future release series v5.5.x. Any bug fix committed to master that applies to 5.4.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.4 or 5.3. Expect that v5.4.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.5.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime in the first part of 2021. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Branch 5.4 has been created
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 has been created, to be used for releasing v5.4.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.4 cd kamailio-5.34 git checkout -b 5.4 origin/5.4 Pushing commits in this branch: git push origin 5.4:5.4 Note that 5.4 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.4 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.4 will be out, respectively Kamailio v5.4.0. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Drop local generated requests
by Patrick Wakano 10 Jul '20

10 Jul '20

Hello list, Hope you all doing fine! I have a case where I want to drop a local generated request but it looks like you can't drop the request from the tm:local-request route. I found this ticket https://github.com/kamailio/kamailio/issues/980, but I could not find somewhere in the docs or emails definitely saying that drop() is not supported from tm:local-request.... And in case we definitely can't drop the request in there, any ideas on how to avoid a local generated request of going out? I am specifically interested in dropping Registers from the uacreg module under certain situations, but I don't want to disable it or delete it from the DB.... Thank you, Kind regards, Patrick Wakano

2 4

Help on dumping running config in Kamailio
by BALL SUN 09 Jul '20

09 Jul '20

Hi I need an urgent help, I accidentally pipe the output to the kamailio config, and now I lost the copy of my latest setting, is there a chance that I can dump the running config back to file? Thanks PLEASE HELP RBK

2 1

Branching 5.4 in git repository
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, so far it looks pretty calm during the testing phase, some static analysis didn't reveal any new major issues (well, this is done also during development phase). Therefore, if nothing else is proposed or shows up, I am considering to create the git branch 5.4 next week on Thursday -- this branch will hold the code for 5.4.x release series. After branching 5.4, the master branch will be open again for new features. The 5.4.0 enters the release candidate phase, likely to have the first release out in 1-2 weeks after branching. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

Maintenance work on kamailio.org server - Wed, July 8, 2020
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, on Wednesday, July 8, 2020, there will be some work on Kamailio project infrastructure. Therefore there could be short intervals of downtime for online resources like the web server, wiki, documentation and downloads portals, mailing lists or the GIT repository mirror. Once the maintenance work is finished, we will post an update. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 2

30X redirect in interconnects - better alternative?
by Gerry | Rigatta.com 08 Jul '20

08 Jul '20

Hi, I would like to use Kamailio for load balancing incoming carrier traffic. We do currently IP authentication and call logic in Yate boxes. Ideally I would like to distribute calls with 30X redirects with the Kamailio dispatcher so that IP authentication and all logic can stay in the Yate boxes. However I have doubts that 30X redirects are generally accepted in interconnects. What is your experience with this? What is the possible alternative to redirects if one wants to keep IP authentication and call logic in the boxes behind the Kamailio SIP router? E.g. how can one reliably check the carrier source IPs behind Kamailio? Custom headers injected by Kamailio? Of cause I can check source IPs with a database lookup in Kamailio but I try to avoid that as this makes the setup much more complicated and error prone. Thank you for your ideas. Gerry

3 2

Re: [SR-Users] sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Update: Disabling the topoh module on the proxy which produces the error seems to stop the failure from manifesting. I'll try using topos_redis instead, but should this be treated as a bug? BR, George On Wed, 8 Jul 2020 at 12:37, George Diamantopoulos <georgediam(a)gmail.com> wrote: > Hello again, > > Indeed $mb seems to contain garbage: > > SCRIPT_MB: ACK <BC><EA><8F> SIP/2.0#015#012Via: SIP/2.0/UDP > 172.30.154.189;TH=dcv;branch=z9hG4bK629b.6af9302cd78dc58dffe817e60124f4ed.0#015#012Route: > <sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@172.30.155.1 > ;lr;received=sip:2.2.2.2:32768 > ;ob;r2=on>,<sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@3.3.3.3 > ;lr;received=sip:2.2.2.2:32768;ob;r2=on>#015#012Max-Forwards: > 68#015#012From: "Anonymous" <sip:unknown@voip.domain.com>;tag=as4bc9e324#015#012To: > <sip:voip-test-user-02@voip.domain.com>;tag=jw7z5s0zvc#015#012Call-ID: > 0138c6370346d1dd7f1b47f604b01f92(a)voip.domain.com#015#012CSeq > <http://0138c6370346d1dd7f1b47f604b01f92@voip.domain.com#015%23012CSeq>: > 102 ACK#015#012Content-Length: 0#015#012TH: dch#015#012#015#012 > > How can this be possible? Capturing traffic on wire shows the RURI I > pasted in my original message and there are no script operations on the > RURI before sanity_check() (message buffer above is printed just before > sanity_check() is run in REQINIT). > > BR, > George > > On Wed, 8 Jul 2020 at 11:18, George Diamantopoulos <georgediam(a)gmail.com> > wrote: > >> I'll post the message buffer ASAP, but in the meantime I don't see how >> config operations could affect the RURI. Here's everything that's happening >> until the sanity check involved: >> >> request_route { >> >> if(is_method("KDMQ")) { >> dmq_handle_message(); >> } >> >> # no connect for sending replies >> set_reply_no_connect(); >> >> if($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")) { >> force_rport(); >> sl_send_reply("483","Too Many Hops"); >> exit; >> } >> >> # OPTIONS and NOTIFYs directed to myself >> if(is_method("OPTIONS|NOTIFY") && uri==myself && $rU==$null) { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> # All keep-alive methods regardless of destination >> if ( $hdr(Event) == "keep-alive") { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> if(!sanity_check("17895", "7")) { >> xlog("Malformed SIP request from $si:$sp\n"); >> exit; >> } >> >> BR, >> George >> >> On Wed, 8 Jul 2020 at 10:58, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> check your config operations, because the R-URI seems to be the next >>> string (without quotes): "<BC><EA><8F>" >>> >>> You can try to print $mb in such case to see the entire SIP message >>> buffer. >>> >>> Cheers, >>> Daniel >>> On 08.07.20 09:48, George Diamantopoulos wrote: >>> >>> Hello Daniel, >>> >>> Thanks for the reply. Indeed there is, not sure how I managed to miss >>> that. And it wasn't about the schema after all: >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1254]: parse_uri(): uri too short: <<BC><EA><8F>> >>> (3) >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1328]: parse_sip_msg_uri(): bad uri <<BC><EA><8F>> >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: WARNING: >>> {1 <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:282]: check_ruri_scheme(): failed to parse request uri >>> [<BC><EA><8F>] >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: ERROR: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <script>: Malformed >>> SIP request from 172.30.154.189:5060 >>> >>> Still, not sure what the problem is though... >>> >>> BR, >>> George >>> >>> On Wed, 8 Jul 2020 at 09:30, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> when the ruri scheme check fails, there should be another debug message >>>> saying that. Have you pasted all log messages for the failure case? >>>> >>>> Cheers, >>>> Daniel >>>> On 07.07.20 22:23, George Diamantopoulos wrote: >>>> >>>> Sorry, I realised I copy pasted wrong log messages for Call 1. Here's >>>> the relevant part showing success for call 1 in contrast with Call 2: >>>> >>>> grep 2a859fcc4e1c8f840191a81d7c16e76d kamailio.log | egrep >>>> 'check_ruri_scheme|w_sanity_check' | grep ACK >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 1 >>>> >>>> On Tue, 7 Jul 2020 at 21:34, George Diamantopoulos < >>>> georgediam(a)gmail.com> wrote: >>>> >>>>> Hello all, >>>>> >>>>> I'm not 100% sure this is the only culprit in an issue I'm >>>>> investigating, but superficially it appears that RURI scheme sanity module >>>>> checks from the default config (flags 17895 in REQINIT) fail if the >>>>> RURI in an ACK following a 487 includes parameters. Example from two calls >>>>> from a kamailio instance acting as registrar/usrloc server, INVITE RURIs >>>>> are after usrloc lookup: >>>>> >>>>> Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> These INVITEs produce no complaints. Later, the same registrar >>>>> produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses >>>>> from the next proxy in the path following a CANCEL: >>>>> >>>>> Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> The next proxy (which produced/relayed the 487) processes the ACK for >>>>> Call 1 successfully, but sanity_check at the proxy drops the request for >>>>> Call 2 with: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>>>> >>>>> whereas Call 1 seems OK: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:305]: check_required_headers(): check_required_headers entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:313]: check_required_headers(): check_required_headers passed >>>>> >>>>> Could this be a bug in sanity module? Is there anything one can do in >>>>> config which could result in illegal ACKs being produced for hop-by-hop >>>>> transactions? schema appears to be sip: in both cases... >>>>> >>>>> Thank you. Best regards, >>>>> George >>>>> >>>> >>>> _______________________________________________ >>>> 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 >>>> >>>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Funding: https://www.paypal.me/dcmierla >>> >>>

2 2

Setting up Kamailio as Teams SBC
by Роман С. 08 Jul '20

08 Jul '20

Hello. I'm trying to set up Kamailio as SBC for Teams using https://skalatan.de/en/blog/kamailio-sbc-teams. Setup is completely default (except things mentioned at article), but I use wildcard certificate for TLS. Well, I can't even pass dispatcher: kamcmd dispatcher.list | egrep "RI|FLAG" URI: sip:sip.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 3 URI: sip:sip2.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 2 URI: sip:sip3.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 1 Where do I start to dig?

3 6

sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Hello all, I'm not 100% sure this is the only culprit in an issue I'm investigating, but superficially it appears that RURI scheme sanity module checks from the default config (flags 17895 in REQINIT) fail if the RURI in an ACK following a 487 includes parameters. Example from two calls from a kamailio instance acting as registrar/usrloc server, INVITE RURIs are after usrloc lookup: Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 These INVITEs produce no complaints. Later, the same registrar produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses from the next proxy in the path following a CANCEL: Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 The next proxy (which produced/relayed the 487) processes the ACK for Call 1 successfully, but sanity_check at the proxy drops the request for Call 2 with: DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 whereas Call 1 seems OK: DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:305]: check_required_headers(): check_required_headers entered DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:313]: check_required_headers(): check_required_headers passed Could this be a bug in sanity module? Is there anything one can do in config which could result in illegal ACKs being produced for hop-by-hop transactions? schema appears to be sip: in both cases... Thank you. Best regards, George

2 5

Some Functions not defined in Documentation of Module ims_registrar_pcscf
by Hamid Hashmi 08 Jul '20

08 Jul '20

I have been using a sample code of PCSCF<https://github.com/kamailio/kamailio/blob/master/misc/examples/ims/pcscf/ro…>, where a function "pcscf_assert_identity" is used but it's not defined in module ims_registrar_pcscf documentation. Can you please explain its functionality. The same way other functions are not defined in the documentation. pcscf_assert_called_identity reginfo_handle_notify pcscf_unregister Regards Hamid R. Hashmi

1 0

Kamailio 5.3.4 - Segfault on http_async_client
by José Seabra 07 Jul '20

07 Jul '20

Hello there, Kamailio is terminating with signal 11 and is generating a coredump file. I opened the core file generated by it and it seems to be related to the module http_async_client. In the following link you can find the output of core dump backtrace full: https://pastebin.com/rapPZy9N Below is the http_async_client config that I have: modparam("http_async_client", "workers", 2) modparam("http_async_client", "connection_timeout", 500) modparam("http_async_client", "hash_size", 2048) modparam("http_async_client", "curl_verbose", 0) modparam("http_async_client", "memory_manager", "shm") If the information provided is not enough to troubleshoot this issue let me know. Many thanks Best Regards -- Cumprimentos José Seabra

1 1

Kamailio Buildroot
by Benjamin Marty 07 Jul '20

07 Jul '20

Hello I would like to ask if someone already tried to build Kamailio for the Embedded Distribution Buildroot. I tried it myself but it looks like Kamailio is not that easy to cross compile. Thanks Benjamin

3 3

siptrace not sending data to hepic
by Andrew Chen 07 Jul '20

07 Jul '20

Hi all, I've been assigned a task to investigate why hepic doesn't generate MOS scores ever since we changed the kamailio interface MTU from 9001 -> 1500. During my investigation, this is what I have found out so far: - Putting MTU back to 9001 removes this condition where no MOS was generated. - I did a tcpdump of port 9063 which is the port used to send data to Hepic capture server and saw it reports packets are fragmented: 3:48:55.895505 IP (tos 0x68, ttl 64, id 38049, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP 13:48:55.896451 IP (tos 0x68, ttl 64, id 38050, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP - Hepic support also caused by using 1500 MTU on the interface. - I tried using the following core parameters to see if they could remedy this issue but didn't seem to work: ## MTU related core settings udp4_raw=-1 udp4_raw_mtu=1500 I can't seem to find any parameters in the siptrace module that controls fragmentation of the packets to Hepic. Any suggestions on how else to troubleshoot this? 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.*

2 2

Keep one registration per device(src_ip+port)
by Zhan Bazarov 07 Jul '20

07 Jul '20

Hello. I'd like to keep one registration per device in usrloc table, for other cases I want to store and maintain one contact per AoR. Does kamailio support it out of the box? For second case save("location", "0x01"); working fine. But how about first case? for example, if one of device located at home, and second located at office. - there are two different network and NAT - how I can make both devices ringing when INVITE is sent? If I do classic multiAor scenario - it can lead to issues caused by multiple contacts in 200OK response for a single device and devices are confusing sometimes. So, what I did now: if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); add_contact_alias(); } [REGISTRAR] { if (nat_uac_test(64)) { if (!save("location", "0x04")) { sl_reply_error(); } } else { if (!save("location")) { sl_reply_error(); } } } modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "xavp_contact", "usrlocattrs") modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "close_expired_tcp", 1) modparam("usrloc", "db_insert_update", 1) modparam("usrloc", "expires_type", 1) modparam("usrloc", "matching_mode", 1) modparam("registrar", "method_filtering", 1) modparam("registrar", "max_expires", 3530) modparam("registrar", "min_expires", 1800) modparam("registrar", "received_avp", "$avp(s:rcv)") modparam("registrar", "gruu_enabled", 0) modparam("registrar", "use_path", 1) modparam("registrar", "path_check_local", 1) modparam("registrar", "path_mode", 1) modparam("path", "use_received", 1) How do u think, maybe modparam("registrar", "append_branches", 0) will make some help? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Nathelper ping from dedicated kamailio instance
by Angelo Pellegrinon 07 Jul '20

07 Jul '20

Hi! I am new to kamailio so please forgive me if I write some errors. I am configuring some kamailio instances in different environments to handle big traffic and I have a problem with usrloc + db + nathelper. What have I done: 1. Configured a kamailio instance to handle sip registers with usrloc to database with database mode 2 (memory + db), disabled nathelper pings. This is ok and performs well; 2. Configured a kamailio instance to handle nathelper ping. Same usrloc + db + nathelper config from 1 except that I use db mode 3 (only db) and that I enabled natping_interval . My problem is that it seems that the second kamailio instance does not send pings. The second one connects correctly to db (if I lookup from kamcmd a user registered from the first kamailio I can find it) but it seems that does not read all locations. Can you help me? Many many thanks Angelo

1 0

rtpengine with mutiple IPs on the same network.
by David Villasmil 07 Jul '20

07 Jul '20

Hello guys, I have rtpengine running on a box with ips like this: external: 172.10.0.10 (advertise 10.10.10.1) internal: 172.10.0.11 Both Ips are on the same network. When i do rtpengine_manage i set "direction=external direction=internal", and the SDP offer is correct. The problem I'm having is that when rtp tried to send to the internal network, the OS (i believe) selects the first interface (external) ip 172.10.0.10 to reach the internal network. This I suppose is not a great deal, but I would like to separate them and only use the internal for internal proxying. Help is appreciated! Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 5

Retrieving cert details from tls peer
by Mark Boyce 07 Jul '20

07 Jul '20

Hi all Been trying to grab the TLS cert details from incoming connections, but failing :-( So with lines just before AUTH is called like this; if (proto == TLS) { xlog("L_INFO", "TLSDUMP $ci peer_subject : $tls_peer_subject\n"); Gets met with a log line line this; INFO: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from 1.2.3.4:11797 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 INFO: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: 5.6.7.8:5061 INFO: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate ... INFO: tls [tls_select.c:168]: get_cert(): Unable to retrieve peer TLS certificate from SSL structure This is with verify_certificate and require_certificate set to no in tls.cfg If I try and set the following in tls.cfg [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = yes I see in the logs; INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSs<default>: tls_method=22 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/tls-certs/cert.pem' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/tls-certs/privkey.pem' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 NOTICE: tls [tls_domain.c:1095]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... INFO: tls [tls_domain.c:692]: set_verification(): TLSs<default>: Client MUST present valid certificate INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=1 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 INFO: tls [tls_domain.c:692]: set_verification(): TLSc<default>: Server MUST present valid certificate ... ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417C086:SSL routines:tls_process_client_certificate:certificate verify failed Which looks like verification is being enabled when I add require? Would someone be kind enough to point out what I am missing please? (Assuming it’s not a bug :-) Thanks Mark -- Mark Boyce Dark Origins Ltd

2 8

About routing calls
by ahmed moghazy 07 Jul '20

07 Jul '20

Excuse me i already made integration for kamailio with asterisk but i want to make sip calls through kamailio and media fuctions through asterisk , i want kamailio to make calls away from asterisk , i just wanted asterisk to use its dahdi module .’ Thank you.

2 1

Kamailio Push Notification for iOS13
by william anderson 07 Jul '20

07 Jul '20

Hello, Yesterday, there was a new release of Linphone IOS app, but i need help from some kamailio expert how to add PUSH notification to the application to receive incoming calls when the phone is not in use (wake up required) from a kamailio server. I have read on internet old tutorials (tsilo module) but most of them are incomplete or do not explain in an easy way how to do it (for an average kamailio user). Anyone could do an easy to understand tutorial ? Regards, William Anderson

2 1

Looking for a VoIP & Kamailio Dev-Ops
by Henning Westerholt 06 Jul '20

06 Jul '20

Dev-Ops VoIP & Kamailio GILAWA is looking for a new colleague to join our team! Tasks: Deploying, developing, and integrating our software based on Kamailio Testing and validation of our customer solutions Solve complex customer problems in the VoIP field Fix Bugs and create extensions in the Kamailio project Requirements: Good background in VoIP telephony and Kamailio Stack Experience with signalling standards (SIP, RTP etc..) and troubleshooting Knowledge and experience in one of the following languages: C/C++, Bash, PHP Experience with Debian or RedHat/CentOS Linux, knowledge of Ansible would be a plus Fluent in English Based in European or neighbouring time zone (e.g. EMEA area) Self-organized and strong communication skills Given the current situation this is a remote position. If you are interested, please send me a (direct) e-mail with an CV or Linkedin link, please include your availability and salary expectations as well. Best regards, Henning Westerholt -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

3 3

Kamailio 5.3.5, usrloc: Error inserting location (integer out of range)
by Stefan 03 Jul '20

03 Jul '20

Hi Maillist Kamailio-Version: 5.3.5 built from source with postgres-db. I get an Error if I try to add a static location with kamctl, see my example: kamctl ul add 1234 sip:1234@10.10.10.10:5060 NULL NULL Log output: Jul 1 09:28:20 kamailio-srv02 /sbin/kamailio[3892]: ERROR: db_postgres [km_dbase.c:266]: db_postgres_submit_query(): 0x7fbe5398c148 PQsendQuery Error: ERROR: integer out of range#012 Query: insert into location (username,contact,expires,q,callid,cseq,flags,cflags,user_agent,path,methods,last_modified,ruid,reg_id,server_id,connection_id,keepalive,partition ) values ('1234','sip:1234@10.10.10.10:5060','1970-01-01 00:00:00',1.000000 ,'dfjrewr12386fd6-343(a)kamailio.rpc',1,0,0,'kamailio SIP Router - RPC Server','NULL',2147483648,'1970-01-01 00:00:00','ulcx-5ef99724-f37-1',0,0,0,0,22) truncated output of kamctl ul show: "AoRs": [{ "Info": { "AoR": "1234", "HashID": 924651761, "Contacts": [{ "Contact": { "Address": "sip:1234@10.10.10.10:5060", "Expires": "permanent", "Q": 1, "Call-ID": "dfjrewr12386fd6-343(a)kamailio.rpc", "CSeq": 1, "User-Agent": "kamailio SIP Router - RPC Server", "Received": "[not set]", "Path": "NULL", "State": "CS_NEW", "Flags": 0, "CFlags": 0, "Socket": "[not set]", "Methods": -2147483648, "Ruid": "ulcx-5ef99724-f37-1", "Instance": "[not set]", "Reg-Id": 0, "Server-Id": 0, "Tcpconn-Id": 0, "Keepalive": 0, "Last-Keepalive": 0, "Last-Modified": 0 } }] } }, { So the log tells me: integer out of range. I think it's this one: "Methods": -2147483648, This let's me thinking that this is a bug. Someone else? Regards Stefan

2 8

403 Forbidden error You must register first with a S-CSCF
by Pavithra Mohanraja 03 Jul '20

03 Jul '20

Hi, I am facing below error when i try to make a call using pjsip .. Registration is successful. I am following the open 5gs VolTE setup. I have not changed any configurations in any of the config files. But I am facing below issue .. Can you please help me what could be the reason. *reason=403 (Forbidden - You must register first with a S-CSCF)* Could anyone please help me .. I am ready to provide anyother info if required . Thanks, Pavithra

1 0

Error in SCSCF
by Pavithra Mohanraja 02 Jul '20

02 Jul '20

Hi , I am facing below error in s-cscf when i send a BYE from alice to bob in kamailio IMS. The problem is in only call termination .Kindly help . Error log in scscf : Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7f4746f89148 alloc'ed from core: core/rvalue.c: rval_new_empty(224) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/socket_info.c:630]: grep_sock_info(): checking if host==us: 11==11 && [10.244.0.57] == [10.244.0.54] Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/socket_info.c:637]: grep_sock_info(): checking if port 4080 (advertise 0) matches port 5060 Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0:sip.example.com:0) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0:sip.example.com:5060) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0: scscf.sip.example.com:4080) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/forward.c:423]: check_self(): *host (0:10.244.0.57:5060 <http://10.244.0.57:5060>) != me* Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: sip:bob at sip.example.com Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: *Domain not served 403* Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: *BYE *( sip:alice@sip.example.com (10.244.0.52:4060) to sip:bob@sip.example.com, -ARSjM.TTZgxhzwDjKRN0DtENWD4bKl0) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7f4746dd7010, 30) called from core: core/ut.c: as_asciiz(161) I am not getting what it checks .. 10.244.0.57 -> UE bob 10.244.0.52 -> pcscf 10.244.0.53 -> icscf 10.244.0.54 -> scscf 10.244.0.56 -> UE alice Kindly help Thanks, Pavithra

2 4

Sql_result_free() not cleaning properly
by Duarte Rocha 01 Jul '20

01 Jul '20

Greetings, I'm trying to clean a SQL result using "sql_result_free()" but i'm not having success. Can you please verify if i'm doing anything wrong here? This is my code : xerr("Pre clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); sql_result_free("Cli_PICKER_Default_Search_Result"); xerr("Pos clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); The result is the following : ERROR: <script>:Pre clean [0] = Iterator 0 = 111111111111 ERROR: <script>:Pos clean [0] = Iterator 0 = 111111111111 Am i using sql_result_free correctly or doing something wrong? Best Regards,

4 4

Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start -UPDATE
by Martin W Woscek 30 Jun '20

30 Jun '20

Problem: The mysql db issues with the scscf db include: table version, disappearing impu and/or UE data retained in tables when none are registered Workarround: In order to get over those various mysql issues listed causing the S-CSCF not to come up properly as boot up, 1. create a fresh new scscf database (kamadbctl [drop|create] scscf and import the msql script from the scscf configs) 2. back up the new scscf db 3. modify the init.d startup script for the scscf to restore the fresh db upon each start/restart of the S-CSCF /etc/init.d/kamailio_scscf .. case "$1" in start|debug) mysql -u root --database=scscf < /etc/kamailio/ims_databases/scscf.sql The S-CSCF has been stable and dependable on each boot. _Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:50 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this virtual machine? 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 Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin

1 0

default config and options for Kamailio 5.4.x
by Daniel-Constantin Mierla 30 Jun '20

30 Jun '20

Hello, with the release of Kamailio 5.4.0 approaching, I opened an item in the tracker to collect suggestions about what should be tuned in the default config file. If you feel like something should be there, add your comment to: * https://github.com/kamailio/kamailio/issues/2381 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

Re: [SR-Users] RTP voice failover
by Yufei Tao 30 Jun '20

30 Jun '20

Hi, RTP FO is possible, e.g. with keepalived+rtpengine+Redis as people suggested. Here's the guide for configuring it which I find very useful: https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications A possible way is to have a pair of rtpengine instances in master-slave setup. When master goes down, RTP continues on slave (new master) after a few seconds gap, when the virtual IP(s) migrate over. Another useful link: https://www.kamailio.org/events/2016-KamailioWorld/Day2/20-Pawel.Kuzak-High… Good luck! Yufei <https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications>

1 0

kamailio and rtpengine in docker
by David Villasmil 29 Jun '20

29 Jun '20

Hello guys, I've got a kamailio on a docker and kamailio on another docker on different boxes. The issue i'm having is that when kamailio tries to enable rtpengine it tries pinging it. The ping arrives at the actual container where rtpengine is running, but for some reason i haven't figured out, rtpengine (or the container) is not responding to the ping. I'm writing here in the hopes that someone has already implemented this an come across this and a solved (for sure) I'd appreciate some help. Many thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Valid URL characters in the RURI
by Sam Ware 29 Jun '20

29 Jun '20

I'm using Sanity Checks. Is an underscore (_) not allowed in the host portion of the RURI? I'm on version 5.3. I reviewed the pase_uri function in the core parse_uri.c but I am not super proficient in C. case URI_HOST_P: switch(*p) { check_host_end; default: if(!isalnum(*p) && (*p != '.') && (*p != '-') && !uri_host_char_allowed(*p)) { goto error_bad_host; } } break; I see it calls out periods (.) and dashes (-) but not underscores. I'm wondering why, being underscores are allowed in URLs. -- Sam D Ware

2 1

Changing Debug Level on the fly
by Sam Ware 29 Jun '20

29 Jun '20

I am trying to change the debug level without restarting the service. My research says this is right but I get an error message. $ sudo -u kamailio kamcmd dbg.set_mod_level core 2 error: 500 - cannot store parameter -- Sam D Ware

2 1

SIP OPTIONS between UE and P-CSCF
by BALL SUN 29 Jun '20

29 Jun '20

Hi We have setup UE with Kamailio IMS, and after the registration is successful. The 2nd SIP REGISTER is sent from UE to P-CSCF through IPSEC tunnel. But we found that the SIP OPTIONS (I believed it is a health check) from P-CSCF to UE is not through the IPSEC tunnel. our question is 1. Should SIP OPTIONS run inside the IPSEC tunnel or it should be outside? 2. if it can be configured to run inside the IPSEC tunnel, what configuration can we change to achieve it? - RBK

2 3

How to add parameter ;user=phone in RURI
by Minh Phan 28 Jun '20

28 Jun '20

Hi List, I'm having trouble adding the parameter user=phone in the RURI. I have tried to use the function add_uri_param from siputils module but it doesn't work (see below). Here is the code I use to add user=phone parameter to FROM, TO and RURI uac_replace_from("", "sip:$var(normalized_number)@sip.domain.com ;user=phone"); uac_replace_to("", "sip:$var(normalized_number)@sip.domain.com;user=phone"); $rU = $var(normalized_number); *add_uri_param*("user=phone"); Thanks in advance for any hints, Best regards, Minh Phan

2 1

RTP voice failover
by Evgeniy 27 Jun '20

27 Jun '20

Is that possible to implement RTP voice fail-over? In case RTP node goes down - the call should not be interrupted. This is kinda strange - but client want to have super redundant system. For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2 nodes (yes - network traffic will increase) Or maybe is a way to recover UDP connections after a crash and replace node state. Maybe use a Kubernetes tools for that purpose ? Here what i found in mailing-list - https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html Many thanks for any hints to this topic.

6 9

Kamailio IMS with ENUM
by BALL SUN 26 Jun '20

26 Jun '20

Hi Is there any information that can help to setup ENUM query in Kamailio IMS? what we wanted to achieve is to when A in domain A (the local domain) makes an INVITE to B, it will check the domain of B from ENUM backend. if it is not on the local domain, it will route to remote IMS core. Thanks - RBK

2 7

Does kamailio sip domain makes sense if i defined some alias
by PICCORO McKAY Lenz 26 Jun '20

26 Jun '20

in kamctlrc i defined an DOMAIN but later that ip was updated.. so my question: it make sense inclusivelly if already defined more allias in kamailiorc.cfg file? in case of more complicated response,, anybody can explain me about that why? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) -RESOLVED
by Martin W Woscek 26 Jun '20

26 Jun '20

Hello, Just wanted to update what fixed our issue. The "contact_user" which is "asterisk" that we were using for the Kamailio context for outgoing SIP trunk/channel in the pjsip.conf file of the Asterisk 16 server was not a valid user in the IMS HSS database (FHoSS implementation). However not being an expert in the IMS, IMHO, the IMS should have kicked out this INVITE as soon as credentials were checked at the front door of the IMS, but rather Kamailio sent it from I-CSCF to S-CSCF to P-CSCF, which is the correct flow, until the P-CSCF sent it back to I-CSCF which then resulted in a 604 HSS user unknown being sent thru the reverse direction back to Asterisk. Anyway, its always good to end Friday on a high note. Thanks, _Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 1:32 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; miconda(a)gmail.com Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Just to clarify what we are seeing, here is a packet from the INVITE from the S-CSCF (6060) to the P-CSCF(4060). I have eliminated the S-CSCF as this packet looks correct. The P-CSCF is not handling the request URI correctly, and simply sends it back to the I-CSCF where I-CSCF cannot find the user in the HSS based on the ruri updated by the S-CSCF with the local IP address of the UE (192.168.20.1). The P-CSCF should send this INVITE to the UE (703222888(a)192.168.20.1<mailto:703222888@192.168.20.1>) No. Arrival Time Source Destination Source Port Destination Port SIP to tag SIP from tag Method Record-Route Userinfo Request-URI Media Port Protocol Length Request-URI Host Port Owner Username Media Type Media Format Origin-Host Type slice_type SIP from address User Part SIP from address Host Part SIP to address User Part SIP to address Host Part Contact URI User Part Contact URI Host Part Request-URI User Part Request-URI Host Part Request-URI Host Port Info 1553 Jun 17, 2020 09:31:21.722451703 Eastern Daylight Time 192.168.20.128 192.168.20.128 6060 4060 d490a30c-42a8-4fc0-9fe9-b7a21537386d INVITE mt sip:7032228888@192.168.20.1:63871;transport=udp 10626,13370 SIP/SDP 2263 63871 - audio,video ITU-T G.711 PCMU,DynamicRTP-Type-101,0,101,101,DynamicRTP-Type-99,99,99 7033331111 asterisk.irisims.org 7032228888 ims522.irisims.org asterisk 192.168.20.131 7032228888 192.168.20.1 63871 Request: INVITE sip:7032228888@192.168.20.1:63871;transport=udp | Frame 1553: 2263 bytes on wire (18104 bits), 2263 bytes captured (18104 bits) on interface any, id 0 Linux cooked capture Internet Protocol Version 4, Src: 192.168.20.128, Dst: 192.168.20.128 User Datagram Protocol, Src Port: 6060, Dst Port: 4060 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:7032228888@192.168.20.1:63871;transport=udp SIP/2.0 Method: INVITE Request-URI: sip:7032228888@192.168.20.1:63871;transport=udp Request-URI User Part: 7032228888 Request-URI Host Part: 192.168.20.1 Request-URI Host Port: 63871 [Resent Packet: False] Message Header Record-Route: <sip:mt@192.168.20.128:6060;lr=on;ftag=d490a30c-42a8-4fc0-9fe9-b7a21537386d;did=7d2.e0f1> Route: <sip:term@pcscf522.irisims.org;lr> Via: SIP/2.0/UDP 192.168.20.128:6060;branch=z9hG4bK285b.2a57abaad57cfd55bc18a23d4e9cefc8.0 Via: SIP/2.0/UDP 192.168.20.128;branch=z9hG4bK285b.793ba7dcf9fcd18ddef225b83896340e.1 Via: SIP/2.0/UDP 192.168.20.131:5060;received=192.168.20.131;rport=5060;branch=z9hG4bKPjef451437-0a4f-4732-ac54-318740956335 From: <sip:7033331111@asterisk.irisims.org>;tag=d490a30c-42a8-4fc0-9fe9-b7a21537386d To: <sip:7032228888@ims522.irisims.org> Contact: <sip:asterisk@192.168.20.131:5060> Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660 [Generated Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660] CSeq: 19314 INVITE Route: <sip:ims522.irisims.org:5060;lr> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 68 User-Agent: Asterisk PBX 16.6.0 Content-Type: application/sdp Content-Length: 1121 Message Body From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 11:21 AM To: 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>> Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Martin W Woscek <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- 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> Funding: https://www.paypal.me/dcmierla

1 0

Question about textops
by Mojtaba 26 Jun '20

26 Jun '20

Hello, Some confusing things while developing in Kamailio: In is_method_f function in textops module, the code checks the HDR_CSEQ_F in msg also, and if the method name appears in CSEQ_F, The result is returned TRUE. These question will arise why in this function, the CSEQ field in message will check? if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL) { LM_ERR("cannot parse cseq header\n"); return -1; /* should it be 0 ?!?! */ } if(m->s==0) return (get_cseq(msg)->method_id&m->len)?1:-1; else return (get_cseq(msg)->method_id==METHOD_OTHER && get_cseq(msg)->method.len==m->len && (strncasecmp(get_cseq(msg)->method.s, m->s, m->len)==0))?1:-1; I think it would be nice to add new function like: is_cseq_method_f in module to avoid some problem understanding. -- --Mojtaba Esfandiari.S

2 2

rtpengine inserted random line into sdp
by Andrew Chen 26 Jun '20

26 Jun '20

Hi forum, I'm starting my rtpengine project and I'm facing a strange problem with rtpengine. I am seeing this in the SDP part of the INVITE: a=rtcp:52021 a=rtcp-mux 2001:470:7:3A7:0:0:0:2 a=direction:active a=oldmediaip:54.153.25.234 As you can see there is a random insert of the local kamailio IP in there and I'm have tried the following to remove it but failed to succeed: - Remove in the offer the combination of ICE and replace-session-connection flags - Switched between rtpengine_offer and rtpengine_manage - Restarted ngcp-rtpengine-daemon process - Used sdpops function sdp_remove_line_by_preifx() Currently running Kamailio 5.1.2. I am running out of ideas. Any suggestions would be greatly appreciated. -- 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.*

2 4

Advertise concept
by Social Boh 25 Jun '20

25 Jun '20

Hello, maybe i don't understood completely the use of Advertise parameter. We use it only if Kamailio is behind a NAT and Kamailio have to comunicate over is PublicIP with provider, other SIP proxy and PBX but never to accept Users REGISTER. Is it right? Regards -- --- I'm SoCIaL, MayBe

2 3

Server error on LIR select next S-CSCF
by BALL SUN 25 Jun '20

25 Jun '20

Hi I have a setup using Kamailio IMS, and when trying to make a SIP call between clients, S-CSCF returns " SIP/2.0 500 Server error on LIR select next S-CSCF", do you have any idea on how to fix? - RBK

2 5

kamailio - 5.3.2 - Possible memleak in Core Module
by José Seabra 25 Jun '20

25 Jun '20

Hello there. During my stress tests against our kamailio servers I detected that it was running out of shm memory. edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:298]: qm_find_free(): qm_find_free(0x7ff370920000, 4224); Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:432]: qm_malloc(): qm_malloc(0x7ff370920000, 4224) called from core: core/sip_msg_clone.c: sip_msg_shm_clone(496), module: core; Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/sip_msg_clone.c:499]: sip_msg_shm_clone(): could not allocate shared memory from shm pool edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1293]: new_t(): out of mem: edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1439]: t_newtran(): new_t failed edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) The output of kamctl stats shmem confirms that kamailio is running out of memory: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 36427", "shmem:free_size = 14123552", "shmem:max_used_size = 268435456", "shmem:real_used_size = 254311904", "shmem:total_size = 268435456", "shmem:used_size = 215787904" ], "id": 10901 } Then I was check what was the module that were consuming more memory with kamcmd mod.stats all shm and the output of this command showed me that Core module was consuming most of shm memory, as you can see bellow is the output of this command shows that Core - build_req_buf_from_sip_req is consuming most of shm. Module: core { sip_msg_shm_clone(496): 4872856 create_avp(175): 24768 msg_lump_cloner(986): 154376 xavp_new_value(106): 780896 build_req_buf_from_sip_req(2187): 183984624 counters_prefork_init(211): 36864 cfg_clone_str(130): 96 cfg_shmize(217): 848 main_loop(1303): 8 init_pt(106): 8 init_pt(105): 8 init_pt(104): 5256 register_timer(995): 232 cfg_register_ctx(47): 64 init_tcp(5021): 8192 init_tcp(5015): 32768 init_tcp(5007): 8 init_tcp(5000): 8 init_tcp(4993): 8 init_tcp(4987): 8 init_tcp(4975): 8 init_avps(90): 8 init_avps(89): 8 init_dst_blacklist(438): 16384 init_dst_blacklist(430): 8 timer_alloc(498): 96 init_dns_cache(361): 8 init_dns_cache(352): 16384 init_dns_cache(344): 16 init_dns_cache(336): 8 init_timer(267): 8 init_timer(266): 16384 init_timer(265): 8 init_timer(264): 8 init_timer(253): 8 init_timer(221): 8 init_timer(210): 278544 init_timer(209): 8 init_timer(197): 8 cfg_child_cb_new(829): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 qm_shm_lock_init(1202): 8 Total: 190229920 } Then I stopped the sipp that was running against kamailio in order to see if the memory consumed by *build_req_buf_from_sip_req* decreased but it didn't, so, it seems that there is a memleak in this version. If the information that I'm sending in this email is not enough to identify the root cause of this, please let me know if there is anything else that I can do here to help identify better what is the main reason. Than you Best Regards -- Cumprimentos José Seabra

2 2

siptrace: methods blacklist modparam
by Nuno Ferreira 25 Jun '20

25 Jun '20

Hello, We are a heavy Homer/Hepic (tks QXIP team) users and on our setup, there are some SIP methods we don’t want to get into the database or send to the capture server because there’s no value-added for call troubleshooting purposes (OPTIONS, KDMQ, HTTP (from JSONRPC)…) Typically we achieve the message filtering using siptrace in “manual” mode and control what to send from the script. This works fine for some situations, but there are a few we need to have the automatic mode on: - topos enabled: using manual mode we don’t get the “wire” version of the signaling - TCP connections: there are a few situations where the destination port is traced with 0 - I believe this is due to the fact the TCP socket not established when the message is collected There are other technics to filter like using a HEP relay proxy dropping unwanted traffic, but having a native kamailio functionality would make things much easier. With that in mind, I’ve created a PR (#2374 <https://github.com/kamailio/kamailio/pull/2374>) that allows to blacklist methods we don’t want to send to the capture server when using automatic trace mode. I believe this is a very helpful change for the community as well. To block SIP OPTIONS and KDMQ we just need to declare a new modparam: modparam("siptrace", "trace_on", 1) modparam("siptrace", "methods_blacklist_auto", 16896) Let me know what you think about it. Nuno -- *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.*

2 1

Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working
by supreeth herle 25 Jun '20

25 Jun '20

Please try to keep the mailing list in loop so that it helps others as well facing this issue. I believe this was resolved in 5.3.3 branch not sure why you are facing this issue. Anyways, make sure that modules loaded in kamailio.cfg of PCSCF is in following order loadmodule "ims_dialog" loadmodule "ims_usrloc_pcscf" #!ifdef WITH_IPSEC loadmodule "ims_ipsec_pcscf" #!endif loadmodule "ims_registrar_pcscf" Best Regards, Supreeth On Wed, 24 Jun 2020 at 13:36, Pavithra Mohanraja <pavimohan3004(a)gmail.com> wrote: > Hi Supreeth, > > I am getting the below error > > ms_usrloc_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_usrloc_pcscf.so] > 0(455) DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:264]: > mod_init(): Successfully bound to PCSCF Usrloc module > 0(455) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): > found export of <bind_ims_ipsec_pcscf> in module ims_ipsec_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_ipsec_pcscf.so] > 0(455) ERROR: ims_ipsec_pcscf [cmd.c:85]: bind_ipsec_pcscf(): > configuration error - trying to bind to ipsec pscscf module before being > initialized > 0(455) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while > initializing module ims_registrar_pcscf > (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_registrar_pcscf.so) > ERROR: error while initializing modules > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:305]: mod_destroy(): > Error destroying spi generator > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: mod_destroy(): > Error destroying port generator > 0(455) DEBUG: tm [t_funcs.c:84]: tm_shutdown(): start > 0(455) DEBUG: tm [t_funcs.c:87]: tm_shutdown(): emptying hash table > 0(455) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): removing semaphores > 0(455) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): destroying tmcb lists > > > Kindly help > > On Wed, Jun 24, 2020 at 5:03 PM supreeth herle <herlesupreeth(a)gmail.com> > wrote: > >> There is no other IPSec configuration needed apart from running Kamailio >> P-CSCF and UE sending a SIP Register with Security-Client information for >> IPSec creation. Kamailio IMS IPSec module creates IPSec tunnels using >> Netlink library. >> >> Best Regards, >> Supreeth >> >> On Wed, 24 Jun 2020 at 13:29, Pavithra Mohanraja <pavimohan3004(a)gmail.com> >> wrote: >> >>> Hi , >>> >>> Could you please tell how to configure ipsec (general configuration not >>> related to kamailio). >>> >>> Thanks, >>> Pavithra >>> >>> On Wed, Jun 24, 2020 at 2:12 PM supreeth herle <herlesupreeth(a)gmail.com> >>> wrote: >>> >>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>> >>>> Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or >>>> any other HSS you are using to enter IMS user details). SCSCF is not able >>>> to find a registered user with that IMPU (i am assuming that you are >>>> getting this response while calling) >>>> >>>> Increase the debug level on S-CSCF and check out which IMPU is being >>>> referred by SCSCF >>>> >>>> Best Regards, >>>> Supreeth >>>> >>>> On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja < >>>> pavimohan3004(a)gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> I am facing below error in kamailio-ims 5.3.4 version. >>>>> >>>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>>> could you please help me how should i proceed with and where the >>>>> registration details will be stored in kamailio ims. >>>>> >>>>> It fails at this line in kamailio.cfg file in scscf . >>>>> >>>>> route[FINAL_TERM] { >>>>> if (lookup("location")) { >>>>> if (uri==myself) { >>>>> if (!t_newtran()) { >>>>> sl_reply_error(); >>>>> exit; >>>>> } >>>>> xlog("Destination Not Found ----\n"); >>>>> t_reply("404","Not Found - destination user >>>>> not found on this S-CSCF"); >>>>> exit; >>>>> } >>>>> >>>>> here lookup(location) means what it means. Could you please help me >>>>> >>>>> >>>>> >>>>> Thanks, >>>>> Pavithra >>>>> >>>>> On Thu, May 14, 2020 at 9:59 PM supreeth herle < >>>>> herlesupreeth(a)gmail.com> wrote: >>>>> >>>>>> Hello Pavithra, >>>>>> >>>>>> I have written a guide for how to integrate an IMS using Kamailio >>>>>> with open5gs EPC, here is the link >>>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can >>>>>> find steps to setup rtpengine in that guide. >>>>>> >>>>>> Regarding requirement of rtpengine, its mandatory if you want to use >>>>>> the scripts in that example folder without modification. Also, read the >>>>>> READE of sipwise/rtpengine github repo, its very informative (in a gist >>>>>> rtpengine offer much more feature than rtpproxy) >>>>>> >>>>>> why can't we have sip client to sip client rtps flowing like how it >>>>>>> is happening without NAT. >>>>>>> >>>>>> >>>>>> For this, i would definitely suggest to read about SIP working. This >>>>>> is a perfect link https://github.com/onmyway133/blog/issues/284 >>>>>> which offers extensive material (short and very interesting to read with >>>>>> examples) related to SIP. >>>>>> >>>>>> Best Regards, >>>>>> Supreeth >>>>>> >>>>>> >>>>>> On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja < >>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>> >>>>>>> Hi , >>>>>>> >>>>>>> I didn't configure rtpengine ..instead of that only I used >>>>>>> rrpproxy..will it not work? >>>>>>> >>>>>>> If not please tell me how to configure rtpengine and also tell me if >>>>>>> it is really madatory .. why can't we have sip client to sip client rtps >>>>>>> flowing like how it is happening without NAT. >>>>>>> >>>>>>> That's my doubt ..please clarify .. >>>>>>> >>>>>>> Thanks. >>>>>>> >>>>>>> On Thu, May 14, 2020, 8:57 PM supreeth herle < >>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>> >>>>>>>> HI, >>>>>>>> >>>>>>>> Hold on a moment, >>>>>>>> >>>>>>>> >>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>> 10.40.10.30 - hss >>>>>>>>> >>>>>>>> >>>>>>>> So i assume you are using the examples provided in the ims folder >>>>>>>> of kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have >>>>>>>> you modified the P-CSCF configuration to work with RTPProxy? >>>>>>>> >>>>>>>> Best regards, >>>>>>>> Supreeth >>>>>>>> >>>>>>>> On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja < >>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi , >>>>>>>>> I have attached working rtp pcap file. But it is getting reverted >>>>>>>>> back to me . >>>>>>>>> >>>>>>>>> As you said , i have tried restarting rtpproxy several times. but >>>>>>>>> it didnt work . >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Thu, May 14, 2020 at 7:11 PM supreeth herle < >>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Thanks for the pcap. Sorry, to ask this but can you also send >>>>>>>>>> pcap of scenario where you see RTP packets? >>>>>>>>>> >>>>>>>>>> Also, did you restart the rtpproxy after altering the >>>>>>>>>> /etc/default/rtpproxy ? >>>>>>>>>> >>>>>>>>>> i.e. >>>>>>>>>> >>>>>>>>>> $ systemctl restart rtpproxy >>>>>>>>>> >>>>>>>>>> Best Regards, >>>>>>>>>> Supreeth >>>>>>>>>> >>>>>>>>>> On Thu, 14 May 2020 at 14:51, Pavithra Mohanraja < >>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Hi Supreeth, >>>>>>>>>>> I have attached pcap files . kindly check and let me know. >>>>>>>>>>> filter of rtp packets and check. >>>>>>>>>>> >>>>>>>>>>> *filter : sip || rtp* >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Thu, May 14, 2020 at 5:15 PM supreeth herle < >>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Hi Pavithra, >>>>>>>>>>>> >>>>>>>>>>>> Thanks for the clarification. Can you please attach a pcap of >>>>>>>>>>>> the call where you are not seeing the RTP packets? And, btw is the call >>>>>>>>>>>> getting dropped or something in that scenario? >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> Supreeth >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Thu, 14 May 2020 at 13:34, Pavithra Mohanraja < >>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Hi Supreeth, >>>>>>>>>>>>> >>>>>>>>>>>>> I have kamailio ims installed in 4 vms inside openstack. >>>>>>>>>>>>> >>>>>>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>>>>>> 10.40.10.30 - hss >>>>>>>>>>>>> >>>>>>>>>>>>> so all these vms are running in same subnet. >>>>>>>>>>>>> I have a floating ip associated to pcscf. >>>>>>>>>>>>> Now I have two zoiper clients outside openstack >>>>>>>>>>>>> zoiper1 - 10.0.2.15 (NAT Enabled Network adaptor) >>>>>>>>>>>>> zoiper2 - 10.0.2.15 >>>>>>>>>>>>> >>>>>>>>>>>>> when i am making a call from zoiper1 to zoiper2, call is going >>>>>>>>>>>>> properly . SIP messages are coming properly in wireshark packets. >>>>>>>>>>>>> when i filter for RTP packets in wireshark, it is not coming. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> when i have the zoiper clients inside openstack itself ,i can >>>>>>>>>>>>> able to see RTP Packets in wireshark between two sip clients without any >>>>>>>>>>>>> NAT related configurations done. >>>>>>>>>>>>> >>>>>>>>>>>>> Kindly help me. >>>>>>>>>>>>> Thanks. >>>>>>>>>>>>> >>>>>>>>>>>>> On Thu, May 14, 2020 at 4:49 PM supreeth herle < >>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Can you explain some more when you mean by "RTP alone is not >>>>>>>>>>>>>> coming."? And, how are you checking/verifying that? >>>>>>>>>>>>>> >>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:12, Pavithra Mohanraja < >>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Yes I have configured as you mentioned only . In my >>>>>>>>>>>>>>> pcscf only i have rtpproxy also configured. >>>>>>>>>>>>>>> Floating IP i have given in listen address also as >>>>>>>>>>>>>>> listen = 10.40.10.3:4060 advertise 10.45.4.22:4060 >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 10.40.10.3 - private ip >>>>>>>>>>>>>>> 10.45.4.22 - floating ip >>>>>>>>>>>>>>> Still RTP alone is not coming. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Is it the only way of configuring NAT ? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Thu, May 14, 2020 at 4:37 PM supreeth herle < >>>>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hello Pavithra, >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Have you tried advertising the floating IP of the VM >>>>>>>>>>>>>>>> running RTPProxy as follows? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Edit /etc/default/rtpproxy file as follows: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> # Additional options that are passed to the daemon. >>>>>>>>>>>>>>>> EXTRA_OPTS="-l 172.24.15.30 -d DBUG:LOG_LOCAL0" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> here, "-l <PUBLIC_IP>" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hope it helps. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:01, Pavithra Mohanraja < >>>>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Could anybody please give me the approach how to configure >>>>>>>>>>>>>>>>> NAT in kamailio IMS. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I have installed kamailio IMS in openstack environment >>>>>>>>>>>>>>>>> which has private ips for all four vms (each component in one vm). >>>>>>>>>>>>>>>>> pcscf has one floating IP attached. I have enabled "WITH_NAT" and >>>>>>>>>>>>>>>>> configured rtpproxy . >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> when i make a call from outside of openstack using 2 >>>>>>>>>>>>>>>>> zoiper clients with outbound proxy keeping floating ip of pcscf, I am >>>>>>>>>>>>>>>>> getting 200 ok with all SIP packets correctly flowing but there is no RTP >>>>>>>>>>>>>>>>> Packet. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> When i try the same within openstack environment without >>>>>>>>>>>>>>>>> using public ip, I am getting RTP Packets. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly tell me the procedure of NAT Enablement or do i >>>>>>>>>>>>>>>>> need to do anything other than this. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly help because i am stuck with this for long time. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>>> >>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>> >>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>> >>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> Kamailio (SER) - Users Mailing List >>>>>>> sr-users(a)lists.kamailio.org >>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (SER) - Users Mailing List >>>>>> sr-users(a)lists.kamailio.org >>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>

2 5

Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- 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> Funding: https://www.paypal.me/dcmierla

1 1

Kamailio IMS using returned SRV records for transport slelection
by BALL SUN 24 Jun '20

24 Jun '20

Hi All I am new to Kamailio IMS, and I just setup a testing environment where having P-CSCF, I-CSCF and S-CSCF. I would like to use make below works 1. SIP client <-> P-CSCF (either UDP or TCP, based on the transport tag) 2. P-CSCF <-> I-CSCF (TCP by default) 3. I-CSCF <-> S-CSCF (TCP by default) For point 2, I observed that P-CSCF actually will send DNS SRV query to DNS server for resolving _sip._udp. and _sip._tcp. I believed we can just disable the "_sip._udp." so that P-CSCF will always use TCP to communication wih I-CSCF. However, for point 3, I didnt notice I-CSCF send DNS SRV query, My question is how we can force I-CSCF to do that or is there any configuration that we should enable such that I-CSCF always use TCP to S-CSCF? - RBK

2 2

SECFILTER reload from Siremis
by Social Boh 24 Jun '20

24 Jun '20

Hello list, is posible to reload SECFILTER data directly from Siremis? or the only way is: kamcmd secfilter.reload Regards -- --- I'm SoCIaL, MayBe

3 3

dialog's track_cseq_updates seems to break uac_auth() in this configuration
by George Diamantopoulos 24 Jun '20

24 Jun '20

Hello again, This might turn out to be a bug, but I'm posting here first because my config is fairly complicated and it might not be the sanest it could be. The problem manifests as follows: with the sole change being the enabling of track_cseq_updates for the dialog module, kamailio fails to process the 100 Trying it receives following authentication initiated by the uac_auth() function. Without the track_cseq_updates option enabled, the call proceeds successfully, albeit with the CSeq for the auth-carrying INVITE having the same value as the original. Here's a simplified version of the configuration routes involved in this: request_route { ... route(DISPATCH_PROVIDER); ... } route[DISPATCH_PROVIDER] { if ( !ds_select_dst("10", "8") ) { t_send_reply("503", "Downstream carrier unavailable"); exit; } t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_branch("PROVIDER_BRANCH"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } failure_route[DISPATCH_PROVIDER_FAILURE] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(arealm) = "authrealm.com"; $avp(auser) = "authusername"; $avp(apass) = "verys3kr1t"; uac_auth(); t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); $du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp); // I'll explain why this is here later in this e-mail t_relay(); exit; } } branch_route[PROVIDER_BRANCH] { uac_replace_from("$dlg_var(my_new_from)"); uac_replace_to("$dlg_var(my_new_to)"); $rU = "my new RURI user"; } event_route[tm:branch-failure:DISPATCH_PROVIDER_FAILOVER] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { return; # next DST - only for 5xx or local timeout } else if ( t_check_status("5[0-9][0-9]") || (t_branch_timeout() && !t_branch_replied()) ) { if ( ds_next_dst() ) { t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } else { xlog("L_NOTICE", "--- SCRIPT_DISPATCH_PROVIDER_FAILOVER: Failed to route request to PROVIDER! Giving Up. Negative response will be sent upstream.\n"); } } } One will rightfully wonder why use both branch-failure routes and failure_route to handle things. Well, it's not obvious from this because the relevant parts are removed for brevity, but I'm generally using branch failure event routes to try other destinations in the same destination set, and failure routes to additionally try other downstream "providers" in case all entries is the current destination set have failed. I don't think these actions/routes are relevant because the problem manifests without any of these failover mechanisms (of switching over to the next provider) engaging. Regarding the line "$du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp);" in the failure_route, which might seem a little odd, this was added because without it, uac_auth() will send the auth-carrying INVITE always to the first destination in the destination set even if ds_next_dst has been called from the branch-failure event route. Unfortunately I haven't been able to determine if this actually helped, because I have only received a 5xx error once from the downstream peer before the "fix" and haven't been able to reproduce this since then. But that's another issue unrelated to the one I'm asking about here. So what happens with track_cseq_updates enabled is, after uac_auth() successfully sends the authenticated INVITE and the peer starts sending provisional responses, kamailio doesn't seem to acknowledge them. Instead, it will retransmit the auth INVITE as if there was a firewall preventing the 1xx responses from being admitted. Then, the timeout will engage and the configuration script will behave as if t_branch_timeout returns true (it will do ds_next_dst). So I was wondering if this is to be expected with this configuration, or if this should be reported as a bug. Thanks! Best regards, George

2 2

Kamailio-5.3.3. NAT and RTP not working
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi , Could anybody please give me the approach how to configure NAT in kamailio IMS. I have installed kamailio IMS in openstack environment which has private ips for all four vms (each component in one vm). pcscf has one floating IP attached. I have enabled "WITH_NAT" and configured rtpproxy . when i make a call from outside of openstack using 2 zoiper clients with outbound proxy keeping floating ip of pcscf, I am getting 200 ok with all SIP packets correctly flowing but there is no RTP Packet. When i try the same within openstack environment without using public ip, I am getting RTP Packets. Kindly tell me the procedure of NAT Enablement or do i need to do anything other than this. Kindly help because i am stuck with this for long time. Thanks.

4 26

multiple domains with microsoft team
by Leanderson Correia 24 Jun '20

24 Jun '20

I'm testing here to use multiple domains with microsoft teams and I'm having trouble making and receiving calls. It seems that when connecting a call, kamailio keeps the tls connection active and reuses it, making the domain parameters used the wrong ones for new call. Can someone help me, or give a tip? -- Att, Lean

3 2

Presence and call queueing - Kamailio and Asterisk
by Pover A.V. 24 Jun '20

24 Jun '20

Summary: Using Kamailio to load balance between Asterisk servers. Trouble getting call queuing working because can't get device state hints to work reliably. - Hello I'd be grateful for any advice that can be offered for this problem to do with presence and call queueing. I have 3 Kamailio (v5.3.4) servers that load-balance incoming traffic from various sources: a legacy PBX, soft phone clients, and a SIP trunking service. Sitting behind the 3 Kamailio servers are 2 Asterisk (v16 LTS) servers. Kamailio acts as a registrar and user location service. Asterisk is routing calls and (should be) providing device state information for each extension. An external script builds a list of PJSIP endpoints based on the content of the Kamailio subscriber table, and deploys it to the Asterisk servers whenever a change is detected. The AOR for each record is configured to point to Kamailio. No authentication details are attached to these endpoints, as Kamailio is doing authentication before dispatching a request to Asterisk. Whenever Kamailio receives an INVITE, it passes this to an Asterisk server using the dispatcher module (round-robin algorithm '4'); asterisk then processes the request. At the moment, everything is working except for presence/device states and call queueing. I've tried to set up 'presence', 'pua', 'dialog' and '(presence/pua)_dialoginfo' modules on Kamailio with limited success. The presence state when a user toggles 'do not disturb' on their soft phone is passed to other users who are subscribed (watching), however when a user makes a call, no presence updates are seen by subscribed users (the extensions show as 'Idle' or 'Available' instead of 'In Use' or 'Busy'). Asterisk is able to report that an extension is 'Ringing' or 'InUse' when a call is made to that extension, because an INVITE to an internal extension matches a pattern in the dialplan and Asterisk routes it to a PJSIP endpoint with the same name. However, when somebody makes a call, Asterisk is unable to track the state of the extension making the call because the call is going via Kamailio so Asterisk is not seeing the caller as an internal extension. We need a way to make Asterisk think the call is coming from an internal device if the caller ID matches a PJSIP endpoint. Both Asterisk servers are configured to share device state updates with each other, and this is working. Without being able to see device state for an internal calling party, we will be unable to reliably set up call queues in Asterisk. Using the 'mohqueue' module on Kamailio is not something we want to do because of its reliance on a database, which is creating a single point of failure - if the database server is down for any reason, Kamailio refuses to start. Ideally, we would achieve some form of distributed call queueing, however the simplest option seems to be setting up a queue on one Asterisk server and having an active-passive high availability configuration, should a server go down. I'm looking for advice on how best to keep track of what is happening in a call, and for Asterisk to know the states of each extension and make these available in a dialplan hint. Or any other suggestions that might help to achieve call queueing in this setup. Thank you

2 1

Dialog Profiles
by Ilie Soltanici 24 Jun '20

24 Jun '20

Hello, Is there any way in Kamailio to get the profile values the current dialog is part of? For example, I have one dialog with the following profiles: profiles: { { test: abcd } test: 1234 } I need in the reply route to get the values for the test dialog profile. (abcd, 1234). I know I can use get_profile_size, but that will give the number of profiles of the current dialog, but I need the values to loop through. Thanks.

2 2

Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin

2 1

Kamailio ims video calls
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi, Could anyone please give me some idea on how video call works in IMS. Does it need rtp proxy to work? Or is it the same way how voice call works ? It would be good , if anybody share some links on video calls as well voice conferencing. Thanks, pavithra

2 1

kamailio 5.2.2 enabling hashing_type
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello, I'm revisiting some initial issues I had with P-CSCF. I could not set the hashing type in my PCSCF (kamalio 5.2.2) so I simply commented it out and moved on. But now, I may need it as the P-CSCF is not handling INVITES coming from I-CSCF that originated from external SIP server, where the endpoint (UE) is registered to the IMS but the P-CSCF sends the INVITE back to the S rather that the registered UE and fails. kamailio.cfg: modparam("ims_usrloc_pcscf", "hashing_type", 1) So with that, when I set the hashing_type to 1,1 or 2 I get the following starting the pcscf: Starting the P-CSCF: Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Not starting Kamailio IMS P-CSCF: invalid configuration file! Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * 0(5766) ERROR: <core> [core/modparam.c:140]: set_mod_param_regex(): parameter <hashing_type> of type <2> not found in module <ims_usrloc_pcscf> Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: 0(5766) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 364, column 47: Can't set module parameter Am I missing some module that I did not include at compile time? According to the docs, it has no additional dependencies other than the ims_usrloc_pcscf.so https://kamailio.org/docs/modules/5.0.x/modules/ims_usrloc_pcscf.html#idp35… Any help is appreciated. Thanks, _Martin

2 1

kamailio5.2.2 calling t_relay(host, port)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, Im trying to call t_relay() with host and port as documented in: https://www.kamailio.org/docs/modules/5.2.x/modules/tm.html However, I get a subst_str(): no match on P-CSCF start and therefore fails. Is t_relay(host, port) supported as documented? Am I missing something? Thanks, _Martin

2 1

Expired transaction
by Marat Gareev 24 Jun '20

24 Jun '20

Hello! I'm trying to manually reproduce expired transactions which couldn't be cleared. I had one idea, but it was unsuccessful. I tried to send SIGSTOP to a group of Kamailio processes while the transaction is live. The idea is to freeze processes for a while, then continue to work, so that by this time the transaction deletion time was missed. But after a few seconds, the transaction is still cleared. Do you have any ideas for reproduction? P.S. I'm sorry If you received my incoherent mails. These were attempts to reply to the thread, seems to be my mail client or sr-users mailing list doesn't support In-Reply-To header.

2 1

ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found
by Denys Pozniak 23 Jun '20

23 Jun '20

Hello! Could someone point me to the possible source of the problem? Jun 16 09:51:06 proxy-edge /usr/sbin/kamailio[9645]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9640]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9600]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9660]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Modules settings: *loadmodule "pua.so"* *loadmodule "pua_dialoginfo"* *loadmodule "dialog.so" * *modparam("pua", "default_expires", 1200)* *modparam("pua", "outbound_proxy", LOCAL_IP_ADDR)* *modparam("pua_dialoginfo", "include_callid", 1)* *modparam("pua_dialoginfo", "include_localremote", 1)* *modparam("pua_dialoginfo", "include_tags", 1)* *modparam("pua_dialoginfo", "send_publish_flag", 8)* *modparam("dialog", "dlg_flag", 4)modparam("dialog", "db_mode", 0)modparam("dialog", "enable_stats", 1)modparam("dialog", "dlg_match_mode", 1)modparam("dialog", "default_timeout", 7200) modparam("dialog", "hash_size", 8192)* *request_route{* *...* *setflag(8);* *...* *}* Generated SIP PUBLISH: *PUBLISH sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com> SIP/2.0* *Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bK3e55.afdce977000000000000000000000000.0* *To: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>* *From: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>;tag=e98d9fc94bada9746d67946ab287a76f-ba77* *CSeq: 10 PUBLISH* *Call-ID: 72ef85f374c8e703-9659(a)10.10.10.10 <72ef85f374c8e703-9659(a)10.10.10.10>* *Content-Length: 578* *User-Agent: proxy* *Max-Forwards: 70* *Event: dialog* *Expires: 7201* *Content-Type: application/dialog-info+xml* *<?xml version="1.0"?>* *<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>">* * <dialog id="162092997-5060-464(a)BJC.BGI.CA.BBI" call-id="162092997-5060-464(a)BJC.BGI.CA.BBI" direction="initiator">* * <state>Trying</state>* * <remote>* * <identity>sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com></identity>* * <target uri="sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com>"/>* * </remote>* * <local>* * <identity>sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com></identity>* * <target uri="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>"/>* * </local>* * </dialog>* *</dialog-info>* -- BR, Denys Pozniak

2 5

icscf start error
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I have installed kamailio - 5.3.4 version where i am facing below issues in icscf while starting kamailio. db_mysql is loaded but .. what could be the issue? icscf : Aliases: *: ims.sip.example.com:* 0(789) INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(791) INFO: pike [pike.c:97]: pike_init(): PIKE - initializing 0(791) INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size *0(791) ERROR: <core> [db.c:204]: db_bind_mod(): Module db_con1=>mysql not found. Missing loadmodule? 0(791) ERROR: ims_icscf [db.c:83]: ims_icscf_db_bind(): ims_icscf_db_bind: cannot bind to database module! Did you forget to load a database module ? 0(791) ERROR: ims_icscf [ims_icscf_mod.c:270]: mod_init(): Failed to initialise ims_icscf module 0(791) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_icscf (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_icscf.so)* ERROR: error while initializing modules 0(791) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... 0(791) INFO: cdp [diameter_peer.c:367]: diameter_peer_destroy(): destroy_diameter_peer(): Terminating all children... 0(791) INFO: cdp [diameter_peer.c:386]: diameter_peer_destroy(): destroy_diameter_peer(): All processes terminated. Cleaning up. 0(791) INFO: cdp [worker.c:139]: worker_destroy(): Unlocking workers waiting on empty queue... 0(791) INFO: cdp [worker.c:142]: worker_destroy(): Unlocking workers waiting on full queue... 0(791) CRITICAL: cdp [diameter_peer.c:426]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test 0(791) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped 0(791) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized

2 3

Kamailio v5.3.5 Released
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, Kamailio SIP Server v5.3.5 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.5. For more details about version 5.3.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/06/kamailio-v5-3-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

reason=404 (Not Found - destination user not found on this S-CSCF)
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I am using kamailio ims-5.3.4 where i am facing this kind of problem during route(term). call registration is happening successfully. In call Establishment , during route(term) , I am facing destination user not found . what could be the problem?. can anyone please help me. Thanks, Pavithra

2 2

Releasing Kamailio v5.3.5
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, I am considering to release next week (likely on Monday, June 22) the next minor version out of branch 5.3, respectively 5.3.5. Should anyone be aware of issues not reported to the bug tracker or commits for fixes in master not backported to 5.3 branch, let us know. For new issues, the best is to report them on the tracker. For commits, you can provide links/hashids or make pull requests. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Re: [SR-Users] 403 domain not served error in scscf
by David Villasmil 20 Jun '20

20 Jun '20

you shouldn't write to me directly. You can write to the list and address me. Anyway, kamailio thinks the message received is NOT directed to it. You should add an "alias" with the IP you're sending to (or the FQDN) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 On Sat, Jun 20, 2020 at 9:01 PM Pavithra Mohanraja <pavimohan3004(a)gmail.com> wrote: > Hi David, > Could you please help me in this issue > > ---------- Forwarded message --------- > From: Pavithra Mohanraja <pavimohan3004(a)gmail.com> > Date: Fri, Jun 19, 2020, 1:36 PM > Subject: 403 domain not served error in scscf > To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > > > Hi, > I am using kamailio IMS - 5.3.4 > when I am making a call between two clients using zoiper , Call > establishment is done successfully . > > But during BYE packet sent from caller to callee , it is reaching till > S-CSCF and giving 403 Domain not served .. Kindly help me in this regard. > > > I have sent lot of mails regarding this error and no replies .. Kindly > help .. > > Error in scscf : > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7fa1e6945ca8 alloc'ed > from core: core/rvalue.c: rval_new_empty(224) > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/socket_info.c:628]: grep_sock_info(): checking if host==us: 10==11 && > [*10.45.5.22] == [10.45.4.181]* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/socket_info.c:635]: grep_sock_info(): *checking if port 4080 > (advertise 0) matches port 40629* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/forward.c:412]: *check_self(): host != me* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: > sip:alice at sip.example.com > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>:* > Domain not served 403* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: BYE ( > sip:bob@sip.example.com (10.45.4.177:4060) to sip:alice@sip.example.com;transport=TCP, > y-B071ZDjFS-z4NLxREyVw..) > > > Kindly help .. > 10.45.5.22 - sip client alice(a)sip.example.com > 10.45.4.181 - scscf > > > please help.. >

2 1

How do I test the Kamailio service ？
by zhaosl＠shinetechchina.com 20 Jun '20

20 Jun '20

Hi : Now I set up a Kamailio Service on the server, it works now. But I didn`t know how to test the server by trying the voice tool.Can you give me some advices , tools or methods ? Thanks. zhaosl(a)shinetechchina.com

3 2

Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

Hi can you share your kamaili config. I want to add some extra columns to cdr table like what you did. Thanks. Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

2 1

Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

1 0

403 domain not served error in scscf
by Pavithra Mohanraja 19 Jun '20

19 Jun '20

Hi, I am using kamailio IMS - 5.3.4 when I am making a call between two clients using zoiper , Call establishment is done successfully . But during BYE packet sent from caller to callee , it is reaching till S-CSCF and giving 403 Domain not served .. Kindly help me in this regard. I have sent lot of mails regarding this error and no replies .. Kindly help .. Error in scscf : Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7fa1e6945ca8 alloc'ed from core: core/rvalue.c: rval_new_empty(224) Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/socket_info.c:628]: grep_sock_info(): checking if host==us: 10==11 && [*10.45.5.22] == [10.45.4.181]* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/socket_info.c:635]: grep_sock_info(): *checking if port 4080 (advertise 0) matches port 40629* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/forward.c:412]: *check_self(): host != me* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: sip:alice at sip.example.com Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>:* Domain not served 403* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: BYE ( sip:bob@sip.example.com (10.45.4.177:4060) to sip:alice@sip.example.com;transport=TCP, y-B071ZDjFS-z4NLxREyVw..) Kindly help .. 10.45.5.22 - sip client alice(a)sip.example.com 10.45.4.181 - scscf please help..

1 0

Development for v5.4.x is frozen
by Daniel-Constantin Mierla 19 Jun '20

19 Jun '20

Hello, the formal notification that the development for the next major version 5.4.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.4 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, pull requests can still be done, but they will be merged after branching 5.4. Can still be done commits with documentation improvements, enhancements to related tool (e.g., kamctl, kamcmd), exporting missing KEMI functions and completing the functionality of the new modules added in 5.4. Once the branch 5.4 is created, new features can be pushed again to master branch as usual. From that moment, the v5.3.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.4 is going to be built at: * https://www.kamailio.org/wiki/features/new-in-5.4.x Upgrade guidelines will be collected at: * https://www.kamailio.org/wiki/install/upgrade/5.3.x-to-5.4.0 Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.3.x to 5.4.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

MD5 and SHA-256 instead of MD5 or SHA-256...
by Aymeric Moizard 18 Jun '20

18 Jun '20

Hi All, I'd like to improve my setup by switching to SHA-256. However, as a first step, I would like to offer both MD5 and SHA-256 in 2 different WWW-Authenticate header. If I'm correct, this is not doable with the latest auth module? Is this a planned feature? As an alternative, I would like to decide the algorithm in the script instead of a module parameter. It looks to me this is also not doable? Again, is this a planned feature? Thanks to all, Regards Aymeric -- Antisip - http://www.antisip.com

5 10

Freezing development for Kamailio v5.4.0
by Daniel-Constantin Mierla 18 Jun '20

18 Jun '20

Hello, based on the discussions during the last online devel meeting, the plan is to release the next major version (Kamailio v5.4.0) around the mid of this summer. Therefore I propose to freeze the development on Thursday, June 18, 2020. Then we will have about 4-6 weeks of testing before the release is out. Master will be unfrozen when the branch 5.4 is created (usually that was done after 3-4 weeks in the testing phase). If you are planning to introduce new features in the 5.4 release series, be sure you push them to master branch or create a pull request till the freezing date. If anyone else wants a different time line for 5.4.0 release, let us know and we can discuss to match the best for everyone. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 4

removing xavp from the stack.
by David Villasmil 18 Jun '20

18 Jun '20

Hello guys, if i do something like: $(xavp(my_var)) = "1"; $(xavp(my_var)) = "2"; $(xavp(my_var)) = "3"; it is store like: $(xavp(my_var)) = "1"; -> index 2 $(xavp(my_var)) = "2"; -> index 1 $(xavp(my_var)) = "3"; -> index 0 and later i need to use and remove from the stack, like this: use and remove: $(xavp(my_var)) --> would remove index 0 Then the stack would remain like: $(xavp(my_var)) = "1"; -> index 1 $(xavp(my_var)) = "2"; -> index 0 Simply popping the index 0 and reindex the remaining... is this possible? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

dispatcher seems to use the default client certificate
by Mack Hendricks 18 Jun '20

18 Jun '20

Hey All, I'm attempting to use dispatcher to send probe messages using TLS for two different domains. I'm providing the socket attribute, which maps to a certificate in /etc/kamailio/tls.cfg. But, it seems to always select the default client cert, which is not the certificate I want to use. My attrs column in dispatcher looks like this: socket=tls:142.93.159.231:5061;ping_from=sip:mack.dopensource.com socket=tls:142.93.159.231:5062;ping_from=sip:levin.dopensource.com Is there some way to force dispatcher to do TLS cert matching based on the host:ip? Thanks -Mack

5 5

Remove a Line from SDP
by egemen ulus 18 Jun '20

18 Jun '20

Hi all, I am trying to remove a line from the SDP body of INVITE on kamailio 4.4 via SDPOPS module but it does not work even though there is no any error message. This is how I configure kamailio.cfg, it matches with if condition and executes "sdp_remove_line_by_prefix" function, but when I check sended INVITE, I still see the line that beginning with "a=rtpmap". I thought, in the RELAY function, the sdp body might not prepared yet? Is there any suggestion to remove a line from the SDP body? route[RELAY] { ... ... if(is_method("INVITE")) { if( $ct =~ "0553847<tel:+10553847>aaaa"){ xlog("L_WARN", "$ct check_this \n"); sdp_remove_line_by_prefix("a=rtpmap"); } } } /path/kamailio[10332]: exec: *** cfgtrace:request_route=[RELAY] c=[/path/kamailio.cfg] l=614 a=16 n=if /path/kamailio[10332]: exec: *** cfgtrace:request_route=[RELAY] c=[/path/kamailio.cfg] l=611 a=26 n=xlog /path/kamailio[10332]: WARNING: <script>: <sip:0553847<tel:+10553847>aaaa@x.x.x.x:5060> check_this /path/kamailio[10332]: exec: *** cfgtrace:request_route=[RELAY] c=[/path/kamailio.cfg] l=612 a=25 n=sdp_remove_line_by_prefix /path/kamailio[10332]: exec: *** cfgtrace:request_route=[RELAY] c=[/path/kamailio.cfg] l=622 a=16 n=if Regards Egemen U. Get Outlook for Android<https://aka.ms/ghei36>

2 2

null branch(count) variable
by Ilie Soltanici 18 Jun '20

18 Jun '20

Hello, As per documentation after lookup() function I have to get the number of branches from the main branch by using $branch(count) variable, but for some reason, I'm getting always only <null> value, can someone help me to understand what I'm doing wrong? lookup(); xlog("Branch Count: $branch(count)\n"); Kamailio Version: 5.2.7 Thanks.

3 4

403 domain not served error in IMS kamailio
by Pavithra Mohanraja 18 Jun '20

18 Jun '20

Hi, Could any one please help me in this issue ASAP. This is logs from SCSCF. Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/socket_info.c:630]: grep_sock_info(): checking if host==us: 10==10 && [172.17.0.8] == [172.17.0.5] Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/socket_info.c:637]: grep_sock_info(): checking if port 4080 (advertise 0) matches port 5060 Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:172.17.0.8:5060) vs. (0: sip.example.com:0) Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:172.17.0.8:5060) vs. (0: sip.example.com:5060) Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:172.17.0.8:5060) vs. (0: scscf.sip.example.com:4080) Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [*core/forward.c:423]: check_self(): host (0:172.17.0.8:5060 <http://172.17.0.8:5060>) != me* Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: ERROR: <script>: sip:bob at sip.example.com *Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: ERROR: <script>: Domain not served 403* Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: ERROR: <script>: BYE ( sip:alice@sip.example.com (172.17.0.3:4060) to sip:bob@sip.example.com, -KT6eSWl7Dm77fe5hbNbeiHR2kwBjni0) Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7fd1f6db5010, 30) called from core: core/ut.c: as_asciiz(161) scscf.cfg # SIP / UDP listen=udp:172.17.0.5:4080 # SIP / TCP listen=tcp:172.17.0.5:4080 # SIP / TCP/TLS listen=tls:172.17.0.5:4081 #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "scscf.sip.example.com" #!define URI "sip:scscf.sip.example.com:4080" alias=scscf.sip.example.com:4080 alias=sip.example.com:5060 alias=sip.example.com # ENUM-Server to query:

1 0

docker images for kamailio IMS
by Pavithra Mohanraja 18 Jun '20

18 Jun '20

Hi , Could anyone kindly tell me if there is docker images for kamailio IMS. I can able to see docker images for only Kamailio as a SIP server. Please help. Thanks, Pavithra

4 10

Re: [SR-Users] kamailio 5.2.2 enabling hashing_type -Please Ignore
by Martin W Woscek 17 Jun '20

17 Jun '20

Found my answer, use correct version of doc: https://kamailio.org/docs/modules/5.2.x/modules/ims_usrloc_pcscf.html#idm10… From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Wednesday, June 17, 2020 8:47 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] kamailio 5.2.2 enabling hashing_type Hello, I'm revisiting some initial issues I had with P-CSCF. I could not set the hashing type in my PCSCF (kamalio 5.2.2) so I simply commented it out and moved on. But now, I may need it as the P-CSCF is not handling INVITES coming from I-CSCF that originated from external SIP server, where the endpoint (UE) is registered to the IMS but the P-CSCF sends the INVITE back to the S rather that the registered UE and fails. kamailio.cfg: modparam("ims_usrloc_pcscf", "hashing_type", 1) So with that, when I set the hashing_type to 1,1 or 2 I get the following starting the pcscf: Starting the P-CSCF: Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Not starting Kamailio IMS P-CSCF: invalid configuration file! Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * 0(5766) ERROR: <core> [core/modparam.c:140]: set_mod_param_regex(): parameter <hashing_type> of type <2> not found in module <ims_usrloc_pcscf> Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: 0(5766) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 364, column 47: Can't set module parameter Am I missing some module that I did not include at compile time? According to the docs, it has no additional dependencies other than the ims_usrloc_pcscf.so https://kamailio.org/docs/modules/5.0.x/modules/ims_usrloc_pcscf.html#idp35… Any help is appreciated. Thanks, _Martin

1 0

Kamailio IMS support
by Pavithra Mohanraja 16 Jun '20

16 Jun '20

Hi, I am trying to connect EPC with IMS. EPC is separate installation of comac-in-a-box . IMS is kamailio based. Do i have SGi interface configurations in kamailio so that i can establish communication between EPC and IMS. One more clarification. I have seen PCRF communication can be done using CDP .Is there any way to build PCRF as part of kamailio. Kindly help .. Thanks, Pavithra

2 1

Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start
by Martin W Woscek 16 Jun '20

16 Jun '20

Hi Henning, The following are the mysql related logging I see on initial spinup of the Kamailio 5.2.2 IMS VM: Jun 16 06:15:28 ubuntu systemd[1]: Starting MySQL Community Server... Jun 16 06:15:33 ubuntu systemd[1]: Started MySQL Community Server. Jun 16 06:16:09 ubuntu /usr/local/sbin/kamailio[1617]: ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): driver error on query: Table 'scscf.impu_subscriber' doesn't exist (1146) Jun 16 06:16:09 ubuntu kernel: traps: kamailio[1617] general protection fault ip:7f3dfa2874f0 sp:7ffe387dc8c0 error:0 in libmysqlclient.so.20.3.16[7f3dfa261000+33c000] Our mysql version: mysql --version mysql Ver 14.14 Distrib 5.7.29, for Linux (x86_64) using EditLine wrapper Thanks, _Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:50 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this virtual machine? 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 Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin

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I use kamailio with fusionpbx And I have a question
by idan katz 16 Jun '20

16 Jun '20

HI I use kamailio with fusionpbx The fusionpbx on one server with a fixed IP address The kamailio on one server with a fixed IP address Two separate servers I use domains I redirect the domains to the kamailio server To register for phones Everything is fine and it works fine But I'm trying to connect to the fusionpbx GUI In a specific domain I'm not going into the fusionpbx The domain is targeted to kamailio Because of the telephone registration I need to provide a domain GUI for my clients Is there any way to perform only port 80 browser routing That I can access the GUI at the s fusionpbx Without prejudice to the routing of telephone registration in kamailio Thanks in advance for their reply Idan

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Re: [SR-Users] How to resolve these mod_param errors in Kamailio 5.3.1?
by Henning Westerholt 15 Jun '20

15 Jun '20

Hello, please keep the list in CC have a look to the docs later in the README: 3.13. xavp_dst_mode (int) Control what fields are added to the XAVP specified by xavp_dst parameter. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/> From: Muhammad Maab Saleem <Maab.Saleem(a)telenor.com.pk> Sent: Monday, June 15, 2020 7:10 AM To: Henning Westerholt <hw(a)skalatan.de> Subject: RE: How to resolve these mod_param errors in Kamailio 5.3.1? Hi, Henning. Thanks for your response. However, as I pointed out to him, this only fixes one of the parameter. I have three others where there is a conflict. I can't find any replacements for them. Do you have any pointers regarding them? Best, Maab Sensitivity: Internal From: Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> Sent: Saturday, June 13, 2020 6:50 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Cc: Muhammad Maab Saleem <Maab.Saleem(a)telenor.com.pk<mailto:Maab.Saleem@telenor.com.pk>> Subject: RE: How to resolve these mod_param errors in Kamailio 5.3.1? Hello, To quote the e-mail from Ovidiu some days ago: "Take a look at the new param: https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fkamailio.…> Up to version 5.2, regular AVPs were used, but starting with version 5.3 the regular AVPs were replaced by an XAVP." So you need to replace the old parameter with the new one. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fskalatan.…> Kamailio services - https://gilawa.com<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgilawa.co…> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Muhammad Maab Saleem Sent: Friday, June 12, 2020 8:37 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] How to resolve these mod_param errors in Kamailio 5.3.1? This is the error I get when I run ./kamailio -f /etc/kamailio/kamailio.cfg -E ________________________________ loading modules under config path: /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 378, column 61: Can't set module parameter 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <grp_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 379, column 61: Can't set module parameter 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <cnt_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 380, column 61: Can't set module parameter 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <sock_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 381, column 63: Can't set module parameter ERROR: bad config file (4 errors) ________________________________ These are the lines 378-381 from the config file modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") modparam("dispatcher", "sock_avp", "$avp(DISPATCHER_SOCK_AVP)") Best, Maab Sensitivity: Internal

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How to resolve these mod_param errors in Kamailio 5.3.1?
by Muhammad Maab Saleem 13 Jun '20

13 Jun '20

This is the error I get when I run ./kamailio -f /etc/kamailio/kamailio.cfg -E ________________________________ loading modules under config path: /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 378, column 61: Can't set module parameter 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <grp_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 379, column 61: Can't set module parameter 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <cnt_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 380, column 61: Can't set module parameter 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <sock_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 381, column 63: Can't set module parameter ERROR: bad config file (4 errors) ________________________________ These are the lines 378-381 from the config file modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") modparam("dispatcher", "sock_avp", "$avp(DISPATCHER_SOCK_AVP)") Best, Maab Sensitivity: Internal

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Dispatcher module and hash algorithm
by Luis Rojas G. 13 Jun '20

13 Jun '20

Hello, I noticed a couple of strange behaviors, using this version : kamailio-5.3.3-4.1.x86_64 I am using algorithm "0". Hash over Call-ID. The reason for that is that I am just re-sending the message, using send_udp(), to a destination in a group from file dispatcher.list. Kamailio is completely stateless, it does not need to remember anything. And using a hash over Call-ID I can be sure that any other message related to the same call will be sent to the same destination : retransmissions of the INVITE, CANCEL, ACK, BYE, etc. I general, it works, but I detected two problems. I have a group of 5 destinations, where every server is running at 80% of the nominal load. It's a 4 + 1 configuration, for redundancy. I know it's not 100% evenly distributed using hash over Call-ID, but in practice it's almost perfectly even. However, if one node is down, instead of having the four alive running at 100%, I get three running still at 80% capacity, and one is sent 160% of the load (which of course can't process). it seems, all traffic that was supposed to be sent to the failed node is transferred to the same unique destination. That implementation makes my solution for redundancy worthless. Why didn't the implementation use, for instance, a re-hash over the hash? That would allow to redistribute "evenly" over the remaining servers. I tried to solve this by repeating the entries in dispatcher.list, as I noticed that kamailio doesn't check if different lines contain duplicated uris. For instance, if I have two lines with "destination A", and one line with "destination B", like this: 0 sip:server_A 0 sip:server_A 0 sip:server_B I see that "destination A" receives twice the amount of SIP OPTIONS that "destination B" receives. And so, I used this dispatcher.list : 0 sip:server_1 0 sip:server_2 0 sip:server_3 0 sip:server_4 0 sip:server_5 0 sip:server_1 0 sip:server_3 0 sip:server_5 0 sip:server_2 0 sip:server_4 0 sip:server_1 0 sip:server_4 0 sip:server_2 0 sip:server_5 0 sip:server_3 0 sip:server_1 0 sip:server_5 0 sip:server_4 0 sip:server_3 0 sip:server_2 (it's not a random order. It follows a sequence) I thought : "what a genius I am. This way, if any node fails, and kamailio select the "next" to send the traffic to, it will distribute evenly over the rest". It doesn't work. I detected something that looks like a bug. If one entry is repeated at least once, for instance : 0 sip:server_1 0 sip:server_1 0 sip:server_2 0 sip:server_3 0 sip:server_4 0 sip:server_5 if server_1 is down, kamailio still sends some of the INVITEs to it. Not all of them, but many. I can see the SIP OPTIONS been sent to it, and the response "ICMP Destination Unreachable", so Kamailio knows that server_down. However, still sends INVITE requests to that failed node. Look at this trace : You can see all the failed SIP OPTIONS. Still, kamailio sends traffic to that server. It does not send traffic to a failed destination if it's listed only once in dispatcher.list Any ideas? Thanks in advance, Luis -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

1 1

RPC Command tm.clean
by Marat Gareev 12 Jun '20

12 Jun '20

Hello. I have a philosophical question about tm.clean <https://kamailio.org/docs/modules/5.1.x/modules/tm.html#tm.rpc.clean> command. Unfortunately, I don’t understand what this command is for, because in my view, expired transactions are automatically deleted. This is all I could find: pc command tm.clean to clean long time expired transactions (for troubleshooting purposes) [1] <https://www.kamailio.org/wiki/features/new-in-5.1.x> So, can you give an example when a transaction can remain in memory? Thanks!

3 3

Queries regarding Load balancing setup
by Abdul Rasheed 12 Jun '20

12 Jun '20

Hi , I am a newbie who started working on Kamailio only few days ago. I am trying to setup kamailio in such a way that kamailio acts as a load balancer for 2 or more asterisk systems. i have following queries. 1. As i understand , all the SIP registration will be done with Kamailio and the Media of the calls will be routed to Asterisks in a round robin fashion. In such a case , is it required that all the extension numbers to be used are to be configured in all the asterisk systems included in the setup ? 2. If extension 1000 is in asterisk1 and extension 1001 is in asterisk2 , then is it possible to make internal calls between these extensions ? 3. Is it possible to have TLS based registrations to Kamailio 4. My plan is to build a system which can handle 10000s of extensions . So the system is supposed to be scaled up by adding more asterisk servers in order to accommodate more and more extensions as required . THanks and Regards Abdul Rasheed

2 2

Prometheus module stat
by David Villasmil 12 Jun '20

12 Jun '20

Hello all, Anyone here uses the prometheus module? There’s a core stat that includes a hyphen (“-“) and Prometheus doesn’t support that, only underscores are allowed kamailio_usrloc_location-contacts I’m thinking maybe I can create a custom counter and set it to that stat but with the correct name, is that possible? -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

facing Call Connection and Disconnection issue in Kamailio SIP Server
by Rajkumar Samikkannu 12 Jun '20

12 Jun '20

Hi There, This is Raj. We have successfully installed Kamailio SIP Server by following this documentation link https://kamailio.org/docs/tutorials/5.3.x/kamailio-install-guide-git/#insta… We created some users and those users are in the Subscription Table. We registered(logged in) some users on our client side. We could get Acknowledgement while we registered(logged in). We couldn't get any acknowledgement while we tried to make a call or send a message. We got a "Request Timeout" Error in Client Side. If anyone knows about this issue, please guide us what we should do. Thanks in advance.

2 1

Which mode to capture "written to network" traffic with homer and topos
by Karsten Horsmann 12 Jun '20

12 Jun '20

Hi Mailinglist, some of my kamailio-version is 5.3.4 with topos. During the last discussion about sip_trace_mode("t") i found new behaivor. With this tracingmode i see the "internal" before topos writing. Here my questions: Can i change it to "whats on the wire" (topos rewritten mode)? I didnt checked all modes. And if so, can i switch that on-the-fly with $var for example? For topos debugging its fine with sip_trace_mode("t") - not for "hey carrier, you made some mistakes". #!ifdef WITH_HOMER #Siptrace modparam("siptrace", "force_send_sock", "MY_LOCAL_SENDSOCK") # local interface from where to send the duplicated traffic modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE") modparam("siptrace", "hep_mode_on", 1) # enable/disable Homer encapsulate mode modparam("siptrace", "hep_version", 3) # HEPv3 the timestamp and capture agent ID will be included in the HEP header modparam("siptrace", "trace_to_database", 0) # trace to database modparam("siptrace", "trace_on", 1) # enable/disable trace modparam("siptrace", "trace_mode", 0) # default 0, if 1 then you dont need call siptrace flag or sip_trace() modparam("siptrace", "hep_capture_id", 8) # capture agent id Limitation: 32-bit for HEPv3. #!endif #!ifdef WITH_HOMER # Set the tracing mode: message, transaction or dialog. Need 5.3.x sip_trace_mode("t"); #!endif *** captured on the topos kamailio: 2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060 INVITE sip:+49XXXXXXXXXXXX@172.24.52.3:5060 SIP/2.0 Record-Route: <sip:172.24.32.28;r2=on;lr=on;ftag=1ea69f36;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAO3VzZXI9cGhvbmU-;vst=AAAAAAAAAAAAAAAAAAAAAAAAAABLXVQATFVCVUheUUgcXAIBHnBvcnQ9dGNwO3VzZXI9cGhvbmU-;proxy_media=yes;nat=yes> Record-Route: <sip:100.80.0.20;transport=tcp;r2=on;lr=on;ftag=1ea69f36;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAO3VzZXI9cGhvbmU-;vst=AAAAAAAAAAAAAAAAAAAAAAAAAABLXVQATFVCVUheUUgcXAIBHnBvcnQ9dGNwO3VzZXI9cGhvbmU-;proxy_media=yes;nat=yes> Via: SIP/2.0/UDP 172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74 Via: SIP/2.0/TCP AA.AAA.AAA.AAA:5060;rport=5060;branch=z9hG4bKeca7a7f789c9f647560353e3cf1cd592.ba7930dc Max-Forwards: 52 To: <sip:+49XXXXXXXXXXXX@some.provider.example> From: <sip:+49YYYYYYYYYYY@some.provider.example>;tag=1ea69f36 Call-ID: 3af87db089119841(a)AA.AAA.AAA.AAA Contact: <sip:TFFRA002@AA.AAA.AAA.AAA:5060;transport=tcp> Supported: histinfo,replaces CSeq: 647169 INVITE Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REFER, REGISTER, UPDATE P-Asserted-Identity: <sip:+49BBBBBBBBBB@some.provider.example;user=phone> Content-Type: application/sdp Content-Disposition: session Content-Length: 298 X-Group-SBC: 2006 X-Routing-SBC: un2tr-cgn-trunk --------------------------------------- *** captured on the kamailio behind 172.24.52.3 2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060 INVITE sip:+49XXXXXXXXXXXX@172.24.52.3:5060 SIP/2.0 Via: SIP/2.0/UDP 172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74 Max-Forwards: 52 To: <sip:+49XXXXXXXXXXXX@some.provider.example> From: <sip:+49YYYYYYYYYYY@some.provider.example>;tag=1ea69f36 Call-ID: 3af87db089119841(a)AA.AAA.AAA.AAA Supported: histinfo,replaces CSeq: 647169 INVITE Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REFER, REGISTER, UPDATE P-Asserted-Identity: <sip:+49BBBBBBBBBB@some.provider.example;user=phone> Content-Type: application/sdp Content-Disposition: session Content-Length: 298 X-Group-SBC: 2006 X-Routing-SBC: un2tr-cgn-trunk Contact: <sip:btpsh-5ee1105a-1903c-1@172.24.32.28> -- Cheers *Karsten Horsmann*

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Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start
by Martin W Woscek 12 Jun '20

12 Jun '20

What version of MySQL should we be at for kamailio 5.2.2? Thanks, Martin ________________________________ From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:49:54 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this virtual machine? 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> On Behalf Of Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin

1 0

How to change contact in SIP INVITE
by Trần Đức Thành 09 Jun '20

09 Jun '20

Hi, Anyone can help Sincerely, ---- TRAN DUC THANH Technician ZLINK TECHNOLOGY CO., Ltd Add: 4th Floor, 14 Phao Dai Lang Str, Dong Da Dist., Hanoi, Vietnam 186 Bis Tran Quang Khai, Dist 1, HCM, Vietnam Tel: (84-24) 7 309 99 68 / (84-28) 7309 99 68 Mobile: (84)988612049 - (84)904706808 - Skype ID: thanhtd07 Email: [ mailto:tranthanh@voip.com.vn | tranthanh(a)voip.com.vn ] ZLINK - VoIP Solution - Software Licensing - Server&Network Device [ http://voip.com.vn/ | VoIP.com.vn ] - [ http://azsoft.com/ | AzSoft.com ] - [ http://serverhot.net/ | ServerHot.Net ]

3 2

Reconnect call
by Gaurav Bmotra 09 Jun '20

09 Jun '20

hi how we can implement , that if call got disconnected due to low network and will reconnect .. pl help me -- *Regards:* Gaurav Kumar

1 0

Does DMQ auto-discover cluster members + how?
by George Diamantopoulos 09 Jun '20

09 Jun '20

Hello all (again), When running dmq.list_nodes on one of my production kamailio servers, I get several entries corresponding to staging instances of kamailio. These staging instances are located on separate broadcast domains (different VLANs), and none of them are configured in the notification_address parameter for the dmq module. How do they end up there, and how do I get rid of them? Thanks! George

3 3

RTPProxy Traversal for all Calls
by Mike B 09 Jun '20

09 Jun '20

All, Hopefully a quick answer on this one. I am looking for away to have all traffic traverse the RTP relay, not only NAT. Thinking there should be an easy way to do this with RTPPROXY but haven't been able to figure it out. Using Kamailio 5.3 with RTPProxy 2.2 Thanks, Mike

2 1

Homer 7 and Kamailio
by Social Boh 09 Jun '20

09 Jun '20

Hello, Does Kamailio works with homer sip capture server 7 and HEPv3? I'm trying to test without success. Regards -- --- I'm SoCIaL, MayBe

2 6

Re: [SR-Users] Issue of calls drop 32s
by Rex Lin (林昱頡) 09 Jun '20

09 Jun '20

Hi Alex, Thanks for your reply. ICE/STUN haven’t been activated by our clients. All transmission traffic pass through a proxy server. Below is the screenshot of a normal call process. [cid:image001.png@01D62AC5.FBF723D0] Best Regards, Rex -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Alex Balashov Sent: Thursday, May 14, 2020 12:18 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Issue of calls drop 32s It's a tell-tale sign of an end-to-end ACK from the caller not being forwarded properly to the callee. The server should not be forwarding such an ACK "to itself"; this suggests that its Request URI may be improperly constructed (it should be equivalent to the remote Contact URI of the callee in the 200 OK message), or that its Route headers are somehow incorrect (they should mirror the Record-Route set in the 200 OK). -- Alex On Wed, May 13, 2020 at 04:07:16PM +0000, Rex Lin (林昱頡) wrote: > Hi, > > > It’s weird that the calls always drop after 32s while the callee is using public ip. Also, Server forwards ACK to itself with UDP, instead of forwarding to callee with TLS. > > > Moreover, I cannot see “Received” in AOR as the callee finish the registration. > > > I recognized it’s marked by nat_uac_test(“19”) and set with fix_nated_register(), the problem can be resolved. > > > My scenario is as below, for your reference. > > 1.Kamailio version is 4.4.7, listening public ip. > > 2.Caller it’s behind NAT. > > 3.Clients of Linphone. > > > Do you know the reason why it happened? > > > > Best Regards, > > Rex > _______________________________________________ > 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 -- 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/ _______________________________________________ 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

2 5

kamailio 522 S-CSCF failing to start
by Martin W Woscek 08 Jun '20

08 Jun '20

Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin

2 1

RTP port range in kuberenetes for kamailio
by Gaurav Bmotra 08 Jun '20

08 Jun '20

hi i have issue audio issue with kamailio in kubernetse cluster anybody , idea how to fix it in kubernetses thank you -- *Regards:* Gaurav Kumar

3 4

How to change contact in SIP INVITE
by Trần Đức Thành 08 Jun '20

08 Jun '20

Hi everyone, I use kamailio and have some problem, need support. My scenario: - Kamailio ( x.x.x.x ) --> SIP provider (y.y.y.y) When i callout, SIP INVITE have 2 Record Route (1 x.x.x.x , 1 127.0.0.1) and 2 Via (1 x.x.x.x , 1 127.0.0.1), SIP provider reject it. I try to enable topos, SIP INVITE now have 1 Record Route x.x.x.x , 1 Via x.x.x.x. Everything look ok but Contact be masked like <sip:abcdefghijk@x.x.x.x> and SIP Provider reject it. Any one know how to have Contact as <sip:0123456789@x.x.x.x> where 0123456789 is our SIP number. Sincerely, ---- TRAN DUC THANH IT Manager ZLINK TECHNOLOGY CO., Ltd Add: 4th Floor, 14 Phao Dai Lang Str, Dong Da Dist., Hanoi, Vietnam 186 Bis Tran Quang Khai, Dist 1, HCM, Vietnam Tel: (84-24) 7 309 99 68 / (84-28) 7309 99 68 Mobile: (84)988612049 - (84)904706808 - Skype ID: thanhtd07 Email: [ mailto:tranthanh@voip.com.vn | tranthanh(a)voip.com.vn ] ZLINK - VoIP Solution - Software Licensing - Server&Network Device [ http://voip.com.vn/ | VoIP.com.vn ] - [ http://azsoft.com/ | AzSoft.com ] - [ http://serverhot.net/ | ServerHot.Net ]

1 0

Re%3A RPC Command tm.clean&In-Reply-To=<9c051ed0-20b5-cf64-3e9a-cc60111a7e7c%40gmail.com>
by Marat Gareev 06 Jun '20

06 Jun '20

Hi, Thanks for your reply! What do you mean by "freezing operation system"? Can I manually reproduce stalled transactions? I tried to send SIGSTOP to a group of Kamailio processes while the transaction is live. The idea is to freeze processes for a while, then continue to work, so that by this time the transaction deletion time was missed. But after a few seconds, the transaction is still cleared.

1 0

insert new timestamp value to cdr table
by mohsen khashei 05 Jun '20

05 Jun '20

Hi I want to add a new column to acc table.I created a new column in the acc table of type timestamp and named it ring_time. in every call i put the ring time to a $dlg_var like this: $dlg_var(ringtime) = $Ts; then I add a extra column in config like this: modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;ring_time=$dlg_var(ringtime)") but when I try to test it always i get: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Incorrect datetime value: '1591361996' for column `kamailio`.`acc`.`ring_time` at row 1 (1292) Jun 5 17:29:59 kamailio /usr/sbin/kamailio[22901]: ERROR: {2 102 INVITE 105a0f4a3d99a0a5558355e54b43f4e1@192.168.1.121:5060} <core> [db_query.c:244]: db_do_insert_cmd(): error while submitting query Jun 5 17:29:59 kamailio /usr/sbin/kamailio[22901]: ERROR: {2 102 INVITE 105a0f4a3d99a0a5558355e54b43f4e1@192.168.1.121:5060} acc [acc.c:477]: acc_db_request(): failed to insert into database

2 1

How to overwrite DisplayName
by Stefano Larosa 05 Jun '20

05 Jun '20

Hi and first of all thanks for your amazing work. I have a kamailio 5.1.1 and i need to change the DisplayName before sending INVITE to another Asterisk server. I need that "Sergio"<+33558700000(a)80.169.227.18> Becomes "Simona"<+33558700000(a)80.169.227.18> I've tried just changing $fn value: $fn="Simona" Or using uac_replace_from("Simona ",""); But INVITE msg is always From: "Sergio"<sip:+33558700000@80.169.227.18;transport=UDP>;tag=1ac9e066 I've tried this options in different position in kamailio.cfg but i can figure out the problem. Do you have any suggest? Thank you, Stefano.

3 5

Kamailio under NAT
by Aidar Kamalov 05 Jun '20

05 Jun '20

Hello, I am trying to start kamailio+rtpengine+asterisk in docker with docker-compose (not in network host mode). And I faced behavior that I can't understand. When my phone(fanvil) sends a register to kamailio with his ip in contact - 192.168.88.55 (to host's ip address 192.168.88.55), kamailio 200OK answers with "received=172.18.0.1" param in Via. (172.18.0.1 it is host's docker interface, kamailio's ip is 172.18.0.4) After that my phone unregisters with Expires: 0 and sends a new register with 172.18.0.1 in contact. Is that correct? Is there any way to fix this? 2020/06/05 12:05:51.764617 172.18.0.1:46142 -> 172.18.0.4:5060 REGISTER sip:192.168.88.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.100:5756;branch=z9hG4bK14305253915361123;rport From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060> Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 1 REGISTER Contact: <sip:1000@192.168.88.100:5756> Max-Forwards: 70 Expires: 3600 Supported: path User-Agent: Fanvil X3S 2.10.2.6887 0c383e217eb9 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 2020/06/05 12:05:51.765716 172.18.0.4:5060 -> 172.18.0.1:46142 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.88.100:5756 ;branch=z9hG4bK14305253915361123;rport=46142;received=172.18.0.1 From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060 >;tag=1cbccc00ba5f7d13255bc12726ccacc5.59da Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 1 REGISTER Contact: <sip:1000@192.168.88.100:5756>;expires=3600 Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 2020/06/05 12:05:51.779447 172.18.0.1:46142 -> 172.18.0.4:5060 REGISTER sip:192.168.88.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.100:5756;branch=z9hG4bK251317807114669601;rport From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060> Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 2 REGISTER Contact: <sip:1000@192.168.88.100:5756> Max-Forwards: 70 Expires: 0 Supported: path User-Agent: Fanvil X3S 2.10.2.6887 0c383e217eb9 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 2020/06/05 12:05:51.780479 172.18.0.4:5060 -> 172.18.0.1:46142 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.88.100:5756 ;branch=z9hG4bK251317807114669601;rport=46142;received=172.18.0.1 From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060 >;tag=1cbccc00ba5f7d13255bc12726ccacc5.f669 Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 2 REGISTER Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 And new register: 2020/06/05 12:05:51.780287 172.18.0.1:46142 -> 172.18.0.4:5060 REGISTER sip:192.168.88.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.100:5756;branch=z9hG4bK2696168232620424454;rport From: akamvm <sip:1000@192.168.88.55:5060>;tag=2649930762 To: akamvm <sip:1000@192.168.88.55:5060> Call-ID: 257831362525663-2610472014851(a)172.18.0.1 CSeq: 1 REGISTER Contact: <sip:1000@172.18.0.1:46142> Max-Forwards: 70 Expires: 3600 Supported: path User-Agent: Fanvil X3S 2.10.2.6887 0c383e217eb9 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 2020/06/05 12:05:51.781208 172.18.0.4:5060 -> 172.18.0.1:46142 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.88.100:5756 ;branch=z9hG4bK2696168232620424454;rport=46142;received=172.18.0.1 From: akamvm <sip:1000@192.168.88.55:5060>;tag=2649930762 To: akamvm <sip:1000@192.168.88.55:5060 >;tag=1cbccc00ba5f7d13255bc12726ccacc5.4b3b Call-ID: 257831362525663-2610472014851(a)172.18.0.1 CSeq: 1 REGISTER Contact: <sip:1000@172.18.0.1:46142>;expires=3600 Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 -- Aydar A. Kamalov

1 0

Kamailio 5.3.0 and HEPv3 oversized traffic
by Karsten Horsmann 04 Jun '20

04 Jun '20

Hi list, I saw some strange behavior in Kamailio 5.3.0 with HEPv3. My Kamailio serves webrtc and sip tls and captures that to an Homer 7.7 instance. I see only the UAC Parts of the message and not the answers from Kamailio in that scenario. Maybe I reached the max Paket size for udp. Hard to debug. In production I saw everything (but there I used and older Kamailio 5.2.x and Homer 5.5 which can handle fragment udp). Any ideas how to hunt the root cause of the problem? Cheers Karsten

5 11

kamcmd mtree.match
by Volker Denneberg 03 Jun '20

03 Jun '20

Hi, given this mtree kamcmd mtree.list siteId { tname: siteId tprefix: 00492419697994 tvalue: { 00492419697994 2388 } } { tname: siteId tprefix: 492415310752 tvalue: { 492415310752 2388 } } i wonder why this lookup fails: kamcmd mtree.match siteId 49241531075212 0 error: 500 - Invalid Parameters My expectation was that 2nd tprefix should match. I am using modparam("mtree", "db_table", "mtrees") modparam("mtree", "char_list", "0123456789+") modparam("mtree", "mt_tree_type", 0) # payload type for tree data : string Thanks in advance, Volker

3 5

Re: [SR-Users] Request documentation of Path Flow of Kamailio
by Daniel Cassar 03 Jun '20

03 Jun '20

Cheers Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Henning Westerholt <hw(a)skalatan.de> Sent: Wednesday, June 3, 2020 9:39:36 PM To: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: RE: Request documentation of Path Flow of Kamailio Hello, you can find many presentations (which includes diagrams etc..) in the archive of past Kamailioworld conferences on kamailio.org and also on youtube. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt> Sent: Wednesday, June 3, 2020 9:38 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Henning Westerholt <hw(a)skalatan.de> Subject: Re: Request documentation of Path Flow of Kamailio One more thing if you have some diagrams that also be considered as how communication should pass from one client to another from a different network please can you send it to me and thanks. Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt<mailto:Daniel.Cassar.a100127@mcast.edu.mt>> Sent: Wednesday, June 3, 2020 9:33:19 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> Subject: Re: Request documentation of Path Flow of Kamailio Thanks :) Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> Sent: Wednesday, June 3, 2020 9:20:42 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Cc: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt<mailto:Daniel.Cassar.a100127@mcast.edu.mt>> Subject: RE: Request documentation of Path Flow of Kamailio Hello Daniel, thanks for your interest in Kamailio. Looking forward to see the results of your work, please free to give some feedback if its finished. ?? Being an open source project, there is many documentations available e.g. in our wiki. But most of its is more reference character and probably not something that you can directly use in an academic setting. But there is already some work outside in academia which might be useful for you. Check e.g. this diploma thesis out, which describe the implementation of Kamailio as a telephone system for CERN: https://cds.cern.ch/record/1560053/files/CERN-THESIS-2013-073.pdf?subformat… You can find more works that is using Kamailio e.g. with google scholar: https://scholar.google.com/scholar?hl=en&as_sdt=0%2C5&q=kamailio&btnG= 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 Daniel Cassar Sent: Wednesday, June 3, 2020 8:08 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Request documentation of Path Flow of Kamailio Dear Sir/Madam, I am a student, doing a dissertation about the performance of VPN throughout VoIP Services whereas I decided to configure Kamailio with Siremis Web User Interface on Linux 18.04LTS. I also am using GNS3 as I am trying to connect the server with 3 different routers as a form of sites with different network. What am trying to do is that from the SIP I would try to call from one site to another and see the performance however I need some informations of the Kamailio Server. I need the path communication of how Kamailio works meaning the traffic, protocols being used and these are multicast or unicast when transfer from one site to another. Is this information available somewhere as I seem not to find it anywhere which is why I'm sending you to find out if you have these documentations that I request. I hope to hear from you soon and thanks. Sincerely, Daniel Cassar

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Re: [SR-Users] Request documentation of Path Flow of Kamailio
by Daniel Cassar 03 Jun '20

03 Jun '20

Thanks :) Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Henning Westerholt <hw(a)skalatan.de> Sent: Wednesday, June 3, 2020 9:20:42 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt> Subject: RE: Request documentation of Path Flow of Kamailio Hello Daniel, thanks for your interest in Kamailio. Looking forward to see the results of your work, please free to give some feedback if its finished. ?? Being an open source project, there is many documentations available e.g. in our wiki. But most of its is more reference character and probably not something that you can directly use in an academic setting. But there is already some work outside in academia which might be useful for you. Check e.g. this diploma thesis out, which describe the implementation of Kamailio as a telephone system for CERN: https://cds.cern.ch/record/1560053/files/CERN-THESIS-2013-073.pdf?subformat… You can find more works that is using Kamailio e.g. with google scholar: https://scholar.google.com/scholar?hl=en&as_sdt=0%2C5&q=kamailio&btnG= 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> On Behalf Of Daniel Cassar Sent: Wednesday, June 3, 2020 8:08 PM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] Request documentation of Path Flow of Kamailio Dear Sir/Madam, I am a student, doing a dissertation about the performance of VPN throughout VoIP Services whereas I decided to configure Kamailio with Siremis Web User Interface on Linux 18.04LTS. I also am using GNS3 as I am trying to connect the server with 3 different routers as a form of sites with different network. What am trying to do is that from the SIP I would try to call from one site to another and see the performance however I need some informations of the Kamailio Server. I need the path communication of how Kamailio works meaning the traffic, protocols being used and these are multicast or unicast when transfer from one site to another. Is this information available somewhere as I seem not to find it anywhere which is why I'm sending you to find out if you have these documentations that I request. I hope to hear from you soon and thanks. Sincerely, Daniel Cassar

1 1

Request documentation of Path Flow of Kamailio
by Daniel Cassar 03 Jun '20

03 Jun '20

Dear Sir/Madam, I am a student, doing a dissertation about the performance of VPN throughout VoIP Services whereas I decided to configure Kamailio with Siremis Web User Interface on Linux 18.04LTS. I also am using GNS3 as I am trying to connect the server with 3 different routers as a form of sites with different network. What am trying to do is that from the SIP I would try to call from one site to another and see the performance however I need some informations of the Kamailio Server. I need the path communication of how Kamailio works meaning the traffic, protocols being used and these are multicast or unicast when transfer from one site to another. Is this information available somewhere as I seem not to find it anywhere which is why I'm sending you to find out if you have these documentations that I request. I hope to hear from you soon and thanks. Sincerely, Daniel Cassar

1 0

Trouble loading multiple javascripts with KEMI
by Benjamín Visón 03 Jun '20

03 Jun '20

I am using the module *app_jsdt *with *KEMI *and I am having trouble loading multiple javascript files. The interpreter only notices the last file that's loaded. What I am trying to do is separate my logic so that everything is not inside the kamailio.conf.js (or the main file the contains the necessary functions if you will). modparam("app_jsdt", "load", "/etc/kamailio/file1.js") modparam("app_jsdt", "load", "/etc/kamailio/file2.js") cfgengine "jsdt" In the example above, only functions in file2 are recognized. I am aware that I can use jsdt_run or jsdt_load but from KEMI I don't see a clear way to link this with things like events and timers or replies that don't necessarily go through the *request_route*. (As a plus, I am wondering on performance on Javascript vs Python/Lua and if it's viable on high load, if anyone can throw 2 cents on that it would be helpful!) Saludos, [image: image.png] [image: image.png] <https://www.facebook.com/bvisonl> <https://twitter.com/benjaminvison> [image: image.png] <https://instagram.com/bvisonl/> Benjamín Visón / IT Engineer / Software Developer bvisonl(a)gmail.com / (829)-664-5163

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