- sr-users - kamailio.org

rtpengine - Key file error
by Serhat Guler 29 Jan '17

29 Jan '17

Hello, I am trying to setup my Kamailio environment on a new Debian system. Everything went well, except I started facing a problem with the rtpengine. It did install it fine and I can view the program parameters via the help menu; however, when I run it with this command "rtpengine --interface 192.168.0.66 --listen-ng=127.0.0.1:22222 -m 30000 -M 35000 -L 2" or any other commands, it gives an error as "CRIT: Fatal error: Bad command line: Key file does not have group 'rtpengine'". Where is this key file it is complaining about ? I couldn't find anything neither in the doc nor anywhere else. Cheers, Serhat

2 2

missing memory after tls cleanup
by Andy 28 Jan '17

28 Jan '17

Hi, the tls total memory before and after a tls connection is cleaned up does not match, during test there were 10 tls connection created for which I see tls_h_close and tls_h_tcpconn_clean were called exactly 10 times for the same connection ids, bit before and after the tls shm memory does not match, using command: kamcmd mod.stats tls shm could there be a leak? please suggest thanks! -- View this message in context: http://sip-router.1086192.n5.nabble.com/missing-memory-after-tls-cleanup-tp… Sent from the Users mailing list archive at Nabble.com.

2 4

Ansible Kamailio role
by Alberto Llamas 27 Jan '17

27 Jan '17

Hello Kamailians, I just wanted to share an Ansible role for the installation and compilation of a Kamailio server. It may required improvements and I will be more than happy to heard them. *https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/ <https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/>* Cheers, -- Alberto Llamas Telecommunications Engineer dCAA|dCAP|KPAC|SSCA

4 4

Re: [SR-Users] FOSDEM 2017
by DanB 26 Jan '17

26 Jan '17

Hey Daniel, Would be there 2 persons for sure, so please count us also in. Thanks, DanB

2 1

parallel media forking during ringing phase
by Tomas Zanet 26 Jan '17

26 Jan '17

Hi all, I'm writing to the community because I need a hint from Kamailio and voip gurus: the goal is adding parallel forking call with early-media to our PABX where there's a Kamailio instance running. I mean media streams should be sent to all call destinations before answering the call order, for instance, to see the incoming video. As far as I know, from signaling side should not be a problem because, using Kamailio, there’s append_branch function which forks many branches to all users. The problem is related to media streams because there’s no way to fork media to all users after 180 + SDP or 183, right? Obviously all users must negotiate the same media because media transcoding is basically impossible, especially for video streams. I think there are two ways to do that: 1) using Kamailio and Asterisk. Asterisk has Queue function with ringall strategy, but I’m not sure if Asterisk supports multi domain scenarios. Have ever used Asterisk with Kamailio in multi domain scenarios? 2) enhancing rttpproxy-ng. I don't really know the effort to achieve that. In your opinion, what’s the easiest way to achieve that? Thanks, Best Regards. Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

2 2

How does Kamailio decide which protocol to use when fwding to another proxy?
by Pranathi Venkatayogi 25 Jan '17

25 Jan '17

Hi, I have two instances of Kamailio acting as edge proxies. One on the customer side and one on the agent side. Like: customer -> proxy1 -> proxy2 -> agent. Both customer and agent are registered to proxy1/proxy2 via TLS. However when proxy1 forwards to proxy2, it is using UDP. How can I force it to use TLS? Attached is the result of nslookup on the domain: translation.sms-test.cyracom.com. Thanks Pranathi [cid:image001.jpg@01D27631.42773770] From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Ryan Wagoner Sent: Tuesday, January 24, 2017 8:26 AM To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Asterisk Proxy Multiple Devices / BLF Issues I'm following the latest Kamailio and Asterisk Realtime guide to offload registrations from my FreePBX / Asterisk setup and possibly load balance down the road. I'm running Kamailio 4.4.5 and Asterisk 11.6-cert15. I realize FreePBX isn't realtime and will work around that with a database view, etc. I was excited to see Kamailio will handle multiple devices registering to the same device/extension and placing / receiving calls works. I did run into an issue when any device unregisters Kamailio always forwards the register with expires 0 to Asterisk. To workaround this I modified the route[REGFWD] and added the if($hdr(Expires)==$null) chunk of code. I wanted to use caller->count, but ran into stale contact records with expires set to deleted. I then tried enumerating the contacts, but don't understand why ulc(caller->expires) is 10 when kamctl ul show shows expires deleted. The code below works, but I was hoping for an explanation of the expires = 10 or if there was a better way to handle this scenario. Additionally I enabled presence (WITH_PRESENCE) but Kamailio responds 489 bad event for subscribe requests from devices registered to it. I was hoping it would proxy these to Asterisk for BLF support. If somebody could point me in the right direction it would be appreciated. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($hdr(Expires)==$null) { reg_fetch_contacts("location", "$sel(contact.uri)", "caller"); $var(i) = 0; $var(j) = 0; while($var(i) < $(ulc(caller=>count))) { if($(ulc(caller=>expires)[$var(i)])!=10) { $var(j) = $var(j) + 1; } $var(i) = $var(i) + 1; } if($var(j)>=1) { return; } } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $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(); } Thanks, Ryan

2 3

kamctl cr reload got 500 failed to re-built tree
by Igor Potjevlesch 25 Jan '17

25 Jan '17

Hello, With more than 173k entriers into carrierroute, now I'm doing "kamctl cr reload", I got the following reply: 500 failed to re-built tree, see log What could going wrong? Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 4

Issue with From address being modified in ACK when UAC module used version 4.1
by Jonathan Hunter 25 Jan '17

25 Jan '17

Hi Guys, running kamailio 4.1.4 and using uac_replace_from, I am seeing a strange issue with the proxying of an ACK message back from a carrier to freeswitch on the ingress path into a network. So its just a normal call inbound, where on outbound leg we modify the From address, on the inbound leg all remains the same. Now after the ingress side receives the 200ok, it sends an ACK as below; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+44792498881474@192.168.24.8:5080;user=phone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Route: <sip:192.168.24.8;lr=on;ftag=carrier.peering.telecom.im+3+863d20a3+1b9801d2;vsf=AAAAAAAAAAYNBgAPAAgLAgZ3UTMACgBXFUsVFwwcDkAKTwMZGh0YAA8KDkEEQ1NYC0NDXgdOFRpSHg1vbmU-> Content-Length: 0 Max-Forwards: 68 However kamailio changes the From address; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 109.73.69.165:5060;branch=z9hG4bKc1ce.47974fc3da2b669a78f2dcc9a057a127.0 Via: SIP/2.0/UDP 4.4.4.4:5060;rport=5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+441444680332@es132y$}-9>.8n?~9,*%(;zyk393;7e&C^NRone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Content-Length: 0 Max-Forwards: 67 Causing FreeSWITCH to not recognise the request, and therefore not send an ACK. There are no rules set against the ACK processing. Has anyone seen this before? We dont know when it started happening which doesnt help, I will look to setup debug on test environment but just wondered if this is an issue thats been seen before? Many thanks in advance. Jon

2 5

Event when acc is written?
by Tobias 25 Jan '17

25 Jan '17

Hi! I'm wondering if there is a way for the acc-module to notify/send an event to Kamailio (so I can pick it up from the config, similar to event_route) once the acc-module has performed a write? As an example I'm using acc and db-mysql to store acc data in MySQL, but instead of having a service polling the MySQL for new acc's written I'd like to make Kamailio send an event once an acc-row has been written. I realize this could most likely be done by using MySQL triggers "ON INSERT", but I'm wondering if there is another (reliable) way to do this directly from within Kamailio? I haven't tested it, but I would guess that if I tried to use for example event_route[dialog:end] there could be times where the event would hit Kamailio before the acc-module have written to missed_calls? Thanks! /Tobias

4 11

Executing perl script with command line parameters.
by Venkatesh Macha 25 Jan '17

25 Jan '17

Hi all, I am trying to execute external Perl script from the Kamailio config file. My script accepts few command line parameters but if I try to pass command line parameters like below, It is not working. exec_msg("/root/test.pl 10000 '$var(env)' '$fU'"); # i tried with exec_avp also How to execute the external Perl scripts with few arguments. Thank you in Advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Executing-perl-script-with-command-… Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] How does Kamailio decide which protocol to use when fwding to another proxy?
by Pranathi Venkatayogi 25 Jan '17

25 Jan '17

In the case below though TLS session is established between proxy1 and proxy2, invite is fwded on UDP. 1. How to force Kamailio to use existing tls session to communicate? (proxy1) 2. What is the command to look at if existing tls sessions are present from proxy1 a. kamcmd tls.list does it, but I need one which works from within config. From: Pranathi Venkatayogi Sent: Tuesday, January 24, 2017 11:02 AM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Subject: How does Kamailio decide which protocol to use when fwding to another proxy? Hi, I have two instances of Kamailio acting as edge proxies. One on the customer side and one on the agent side. Like: customer -> proxy1 -> proxy2 -> agent. Both customer and agent are registered to proxy1/proxy2 via TLS. However when proxy1 forwards to proxy2, it is using UDP. How can I force it to use TLS? Attached is the result of nslookup on the domain: translation.sms-test.cyracom.com. Thanks Pranathi [cid:image002.jpg@01D27654.0A627200]

2 1

kamdbctl error.
by Venkatesh Macha 25 Jan '17

25 Jan '17

Hi all, I am using debian jessie, I just installed kamailio 4.4 from source and trying to create kamailio database using kamdbctl but when i try to run the kamdbctl create i am getting following charset error. /usr/local/sbin/kamdbctl create MySQL password for root: INFO: creating database kamailio ... ERROR 1115 (42000) at line 1: Unknown character set: 'ISO' Mysql Version : 5.5.52 Thank you in advance -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamdbctl-error-tp155288.html Sent from the Users mailing list archive at Nabble.com.

2 5

Asterisk Proxy Multiple Devices / BLF Issues
by Ryan Wagoner 24 Jan '17

24 Jan '17

I'm following the latest Kamailio and Asterisk Realtime guide to offload registrations from my FreePBX / Asterisk setup and possibly load balance down the road. I'm running Kamailio 4.4.5 and Asterisk 11.6-cert15. I realize FreePBX isn't realtime and will work around that with a database view, etc. I was excited to see Kamailio will handle multiple devices registering to the same device/extension and placing / receiving calls works. I did run into an issue when any device unregisters Kamailio always forwards the register with expires 0 to Asterisk. To workaround this I modified the route[REGFWD] and added the if($hdr(Expires)==$null) chunk of code. I wanted to use caller->count, but ran into stale contact records with expires set to deleted. I then tried enumerating the contacts, but don't understand why ulc(caller->expires) is 10 when kamctl ul show shows expires deleted. The code below works, but I was hoping for an explanation of the expires = 10 or if there was a better way to handle this scenario. Additionally I enabled presence (WITH_PRESENCE) but Kamailio responds 489 bad event for subscribe requests from devices registered to it. I was hoping it would proxy these to Asterisk for BLF support. If somebody could point me in the right direction it would be appreciated. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($hdr(Expires)==$null) { reg_fetch_contacts("location", "$sel(contact.uri)", "caller"); $var(i) = 0; $var(j) = 0; while($var(i) < $(ulc(caller=>count))) { if($(ulc(caller=>expires)[$var(i)])!=10) { $var(j) = $var(j) + 1; } $var(i) = $var(i) + 1; } if($var(j)>=1) { return; } } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $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(); } Thanks, Ryan

1 0

(no subject)
by jay binks 24 Jan '17

24 Jan '17

Hi Guys, Im using a keepalived solution between 2 kamailio boxes, with net.ipv4.ip_nonlocal_bind =1 . On the "non master" I pretty much constantly get these logs in syslog. What Im looking for is a way to suppress these because they are expected in my configuration. is there any way to tell kamailio to not log this... ( or at least, not at CRIT and ERROR ) *Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: ERROR: tm [../../forward.h:208]: msg_send(): udp_send failed* *Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: ERROR: <core> [udp_server.c:576]: udp_send(): sendto(sock,0x7fc0732d5ab0,380,0,xxx.xxx.xxx.xxx:5060,16): Invalid argument(22)* *Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: CRITICAL: <core> [udp_server.c:581]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net* -- Sincerely Jay

4 9

How to determine correct port to set in record-route header
by Pranathi Venkatayogi 23 Jan '17

23 Jan '17

Hi, I am using Kamailio behind NAT, unable to figure how to make it put “public ip” in Record-route header, I am manually inserting the hard-coded header myself as below. However now I am having trouble choosing the right port number in all scenarios. $var(dstPort) = 5061; if (dst_port==5060) { $var(dstPort) = 5060; } insert_hf("Record-Route: <sip:MY_PUBLICIP_ADDR:$var(dstPort);transport=tls;lr;nat=yes>\r\n", "Record-Route"); In one scenario I see the conflicting port numbers in “via” header vs “Record route”. The ACK is being set to port 5060 based on Record route header and is not being received by the callee. 2017-01-23 14:52:49.970233 [blink.exe 2780]: SENDING: Packet 11, +0:00:36.411867 10.0.27.108:58217 -(SIP over TLS)-> 172.31.211.31:5061 SIP/2.0 180 Ringing Via: SIP/2.0/TLS 63.149.103.72:5061;received=172.31.211.31;branch=z9hG4bK4799.7067bd2a48063748a4c353fa408eaefa.0 Via: SIP/2.0/UDP 10.0.16.52;rport=5060;branch=z9hG4bK4799.935a27c6ef5ad112225964cbe7c1be44.0;i=2 Via: SIP/2.0/TLS 10.11.200.12:51793;rport=51793;received=10.11.200.12;branch=z9hG4bKPj49f6aeca3f9b4155ab4c5304b544aa4d;alias Record-Route: <sip:63.149.103.72:5060;transport=tls;lr;nat=yes> Record-Route: <sip:10.0.16.52:5061;transport=tls;lr;nat=yes> Call-ID: 09ce10efa6a946bf9445ccc21857393e From: "cust1" <sip:cust1@devtranslation.sms-test.cyracom.com>;tag=95082caabecc42548c2fec5ccd29e5de To: <sip:spanish@translation.sms-test.cyracom.com>;tag=ec365cc8489b48a7bc5725d21b7d97a1 CSeq: 17422 INVITE Server: Blink 3.0.0 (Windows) Contact: <sip:20745891@10.0.27.108:58216;transport=tls> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Content-Length: 0 Questions: 1. How can I let in-built record-route function automatically choose “public ip” so I can get rid of my manual insertion altogether. 2. If no option, what is the right way to choose the port on which packet is being received, so it is same as what is on “VIA”. 3. Any other pointers to improve the logic here?

1 0

NAPTR/SRV lookups fail when use_dns_failover is enabled.
by Jeremy Gadd 23 Jan '17

23 Jan '17

Hello, everyone! I'm trying to enable dns_srv_lb but I'm stuck just before I can get there. If I don't enable use_dns_failover, lookups work and calls route but always to the first host. We need to load balance, so that doesn't work very well. If I enable use_dns_failover, the lookups stop working completely and we get a dns lookup failure like so: ERROR: tm [ut.h:296]: uri2dst2(): ERROR: uri2dst: failed to resolve " redacted.hostname.here.com" :unresolvable A or AAAA request (-7) This works: use_dns_cache=on dns_use_search_list=no dns_try_naptr=on # use_dns_failover=on This breaks it: use_dns_cache=on dns_use_search_list=no dns_try_naptr=on use_dns_failover=on I have checked to see that use_dns_failover is compiled in: version: kamailio 4.4.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2

2 4

"Named Telephone Event" (I think) problem
by Pete Kelly 23 Jan '17

23 Jan '17

Hi I am running Kamailio 4.4 and have come across a strange SBC which is sending DTMF within the SDP body of a NOTIFY with a Content-Type audio/telephone-event. Has anybody come across this before? The content itself comes through on a tcpdump/ngrep trace as four dots "...." and Kamailio puts an empty string into the logs if I try to log the contents of $rb, however Wireshark manages to decode it as something readable "\005\000\a\320" and usable. (essentially the digit 5) Is there any way I can get access to this decoded version within Kamailio at all? Even if I have to pass it out to an external program? Many thanks for any help anyone can offer! Pete

2 1

Best way to log incoming/outgoing sip messages?
by Pranathi Venkatayogi 23 Jan '17

23 Jan '17

Hi, What is the best way to get a complete dump of incoming/outgoing SIP messages of Kamailio server? I encountered “SIPTrace” module, but it writes to database. Is there a way we can write to syslog instead? Thanks Pranathi

5 4

Kamailio IMS Secure Websocket closes connection
by Serhat Guler 22 Jan '17

22 Jan '17

Dear all, I am trying to make a call between 2 sipml5 client over WSS. Everything works fine and as expected when using WS. While using WSS, the clients are registered successfully, but when I make a call, the called party is disconnected from the network and the caller gets "User offline" message. There a wireshark capture here: https://drive.google.com/file/d/0B6LMw8kmoAMYMGhrYXNUVXJySEk/view?usp=shari… The call is made from Alice to Bob, the port Nrs: 4060: PCSCF, 5060:ICSCF, and 6060 SCSCF. WWS is running on 4443. The error message that is shown in Firefox console: "The connection to wss://192.168.0.11:4443/ was interrupted while the page was loading." Thanks in advance, Serhat

1 0

Get ip from contact on redirect
by Diego Nadares 20 Jan '17

20 Jan '17

Hi Guys, Anybody knows how to get the ip of the new contact of the redirect? I need it to update one xavp variable that contains the INVITE destination. 6 2017-01-20 12:44:55.138473 172.16.213.38 *172.16.208.121* SIP/SDP 85 Request: INVITE sip:01152780776@172.16.208.121:5060 | 9 2017-01-20 12:44:55.142296 172.16.208.121 172.16.213.38 SIP 661 Status: 302 Moved temporarily | 10 2017-01-20 12:44:55.156268 172.16.213.38 172.16.208.121 SIP 423 Request: ACK sip:01152780776@172.16.208.121:5060 | 16 2017-01-20 12:44:55.293349 172.16.213.38 *172.16.208.111* SIP/SDP 95 Request: INVITE sip:1152780776@172.16.208.111:5060;user=phone | My cfg: # Manage failure routing cases failure_route[MANAGE_FAILURE] { .... if (t_check_status("302")) { xlog("Got a 302 - redirecting"); get_redirects("*:1"); * $xavp(ip_b[$avp(current_route)]=>name) = "172.16.208.111";#Hardcoded for testing* xlog("Got a 302 - redirecting $hdr(Contact)"); route(RELAY); #t_relay(); } Thanks in advance. Diego.

2 3

Siremis Multi domain
by Annus Fictus 20 Jan '17

20 Jan '17

Hi, I think Siremis not work correctly in a multi domain scenario if the extension number are the same for two domains. I explain a little more. I have two extension: 1 200(a)sip1.domain.org 2 200(a)sip2.domain.org On the Siremis I create the user 200 with the SIPuser role. When I login with this user, and go to subscriber data i can see bots extensions. I think is relate to this select on kamailio database: .....SELECT T0.id, T0.username, T0.domain, T0.password, T0.email_address, T0.ha1, T0.ha1b, T0.rpid FROM `subscriber` T0 WHERE T0.username='200' LIMIT 10 Any hint? Regards

2 7

UAC authentication
by anfecora 20 Jan '17

20 Jan '17

can someone please guide me where to find the usage of auc module, thanks in advance. I am registering the uac in one end then i want to forward that registration to an external registrar, if i use $du some domain, some of the ipv6 info is going to be forwarded, i need to make it look more like an a b2bua than a outbound proxy therefore i need to use the auc module(also i need to cover all ipv6 info since i am proxying from ipv4 to ipv6). i am able to register the client to kamailio, then kamailio send the registration to the external registrar, but when the registrar sends back the challenge uac does not reply, any ideas how can achieve this. uac(linphone etc) --registration--->kamailio(registrar module)------register(uac)--->SBC sbc----invite---->kamailio(lookup)---------invite---->uac(linphone etc) this is the code i am using. # Handle SIP registrations route[REGISTRAR] { xlog("REGISTERING "); if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if(af==INET6) setbflag(FLB_IPV6); xlog("IT IS IPV6"); if (!save("location")) sl_reply_error(); route(REGSBCFWD); exit; } } route[REGSBCFWD]{ xlog("CREATING OTHER LEG TO THE SBC"); if(!is_method("REGISTER")) { return; } xlog("CREATING PACKAGE $si -- $fU-- $au--$du-- $tU --$ru"); $var(rip) = "sip.somedomain.com"; $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + "5060"; $uac_req(furi)="sip:" + $tU + "@" + $var(rip); $uac_req(turi)="sip:" + $tU + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $tU + "@" + "10.0.0.222" + ":" "5060" + ">\r\n"; if($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(); } REGISTER sip:sip.somedomain.com:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.222:5060 ;branch=z9hG4bK285f.13a7f0c0000000000000000000000000.0. To: <sip:14052423@sip.somedomain.com>. From: <sip:14052423@sip.somedomain.com >;tag=a82eaf71666096790d1397845fbd3254-ebaa. CSeq: 10 REGISTER. Call-ID: 357e21ae51d8416d-26605(a)10.0.0.222. Max-Forwards: 70. Content-Length: 0. User-Agent: kamailio (4.4.4 (x86_64/linux)). Contact: <sip:140524@10.0.0.222:5060>. Expires: 3600. . SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.0.0.222:5060 ;branch=z9hG4bK4b53.ad842757000000000000000000000000.0. To: <sip:14052423@sip.somedomain.com>. From: <sip:14052423@sip.somedomain.com >;tag=a82eaf71666096790d1397845fbd3254-37da. Call-ID: 357e21ae51d8416d-26603(a)10.0.0.222. CSeq: 10 REGISTER. WWW-Authenticate: Digest realm="sip.somedomain.com", nonce="f5831435-3e69-4b78-ab6d-dcb0897e477b", algorithm=MD5, qop="auth". User-Agent: somedomain.com Host:some proxy. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Content-Length: 0. .

2 3

Return body/string from Perl library
by David Cunningham 20 Jan '17

20 Jan '17

Hello, We are using Kamailio's perl module and want our custom library to set the body on a SIP message. Is it possible to set a SIP message body from within Perl, or can we return a string to the Kamailio configuration to then use with set_body()? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019

2 3

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 19 Jan '17

19 Jan '17

root@homer02:~# netstat -altp Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 0 0 homer02.xxxxx.xxx:sip *:* LISTEN 16018/kamailio tcp 0 0 homer02.xxxxx.xxx:5066 *:* LISTEN 16018/kamailio Thanks. JR > If you use netstat, what is in the recv queue for tcp packets on sip ports? > > netstat -altp > > Cheers, > Daniel > > > On 18/01/2017 16:43, JR Richardson wrote: >> Yes, this is a sipcapture node. I'm listening on a switch port that is >> set to mirror my VoIP traffic. I see all SIP UDP/TCP packets on the >> mirror port and the Ethernet port of the host node. Just don't see any >> TCP packets process in kamailio, debug 3. UDP packets are processed as >> expected. >> >> My config is using port mirror for the capture parameters: >> >> modparam("sipcapture", "capture_on", 1) >> modparam("sipcapture", "hep_capture_on", 0) >> modparam("sipcapture", "raw_ipip_capture_on", 0) >> modparam("sipcapture", "raw_moni_capture_on", 1) >> >> modparam("sipcapture", "raw_sock_children", 4) >> modparam("sipcapture", "raw_interface", "eth1") >> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") >> modparam("sipcapture", "promiscious_on", 1) >> modparam("sipcapture", "raw_moni_bpf_on", 1) >> >> Is there a method I could diagnose if the SIP TCP Packets are getting >> from the kernel network process and the kamailio process? >> >> # ngrep -d eth1 -W byline host x.x.x.x | /var/run/kamailio/kamailio.pid >> >> Or pipe to kamailio local unix socket? >> >> I don't know, I'm just guessing. >> >> Thanks. >> >> JR >> >> >>> Somehow is not clear for me how you have the configuration there ... >>> before commenting further, this needs to be clarified. >>> >>> The node you presented the config is a sipcapture instance, right? What >>> is sending traffic to it? Is another kamailio with siptrace module? Or >>> the sipcature agent? Or you have a port mirroring in the router? >>> >>> Cheers, >>> Daniel >>> >>> >>> On 17/01/2017 16:37, JR Richardson wrote: >>>>> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >>>>>> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >>>>>> using raw socket capture, I think this may be a sipcapture issue, >>>>>> debuging kamailio shows normal startup and processing of UDP SIP >>>>>> packets, but does not show any activity with TCP packets. >>>>> I never used HOMER sofar but when I saw your first message my thoughts >>>>> was that this can't work in a simple way since for TCP you need to >>>>> complete a 4 way handshake before you can start to send data. >>>>> >>>> Interesting. Are you referring to handshaking on the network stack or >>>> SIP TCP TLS handshaking? I guess I can see it two ways. >>>> >>>> 1) if your talking about TCP/IP handshake, even though the SIP packet >>>> comes into the mirror port on the host node, the kernel processing the >>>> TCP packet is not establishing a valid connection due to no TCP >>>> handshake because its only a monitor port, no transmit back, then the >>>> kernel network stack does not pass the SIP TCP packet to the kamailio >>>> process for capture because it drops the packet due to no valid >>>> handshake? >>>> >>>> 2) the kernel network stack is passing the SIP TCP packet to the >>>> kamailio process, but since kamailio cannot handshake back it drops >>>> the packet and does not process through the sipcapture module. This >>>> kinda breaks the whole capture ability for homer with SIP TCP. Using >>>> ngrep, I see all SIP TCP packets, invite -->, trying <--, session >>>> progress <--, request timeout <--, ack -->, etc... >>>> >>>> So how would I diagnose if the network stack is the culprit? Debugging >>>> kamailio is pretty straight forward, setup and listening for SIP TCP, >>>> but never see any processing of any TCP packets. >>>> >>>> Thanks. >>>> >>>> JR >

2 1

Phone number formats per country
by Daniel-Constantin Mierla 19 Jan '17

19 Jan '17

Hello, slightly off-topic, but related to voip -- does anyone have a link that can quickly share to a site with up to date details about phone numbering plans for as many country as possible? Ideally to include the toll and premium number prefixes, the split between mobile and fixed lines if it is the case. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

4 6

ds_is_from_list and NAPTR / SRV
by Björn Bylander 19 Jan '17

19 Jan '17

Hello, I’m trying to resolve an issue with the ds_is_from_list function from the dispatcher module. I’m calling it as follows: ds_is_from_list(”42”, "3", "sip:sip.foo.bar”) This makes Kamailio log an error: ERROR: dispatcher [dispatch.c:2327]: ds_is_addr_from_list(): could not resolve sip.foo.bar sip.foo.bar doesn’t have any A records but there are NAPTR and SRV records: $ host -t naptr sip.foo.bar sip.foo.bar has NAPTR record 20 0 "s" "SIP+D2T" "" _sip._tcp.sip.foo.bar. sip.foo.bar has NAPTR record 10 0 "s" "SIP+D2U" "" _sip._udp.sip.foo.bar. $ host -t srv _sip._tcp.sip.foo.bar _sip._tcp.sip.foo.bar has SRV record 10 0 5060 sip1.foo.bar. _sip._tcp.sip.foo.bar has SRV record 10 0 5060 sip2.foo.bar. $ host -t srv _sip._udp.sip.foo.bar _sip._udp.sip.foo.bar has SRV record 10 0 5060 sip2.foo.bar. _sip._udp.sip.foo.bar has SRV record 10 0 5060 sip1.foo.bar. $ host sip1.foo.bar. sip1.foo.bar has address 10.47.11.1 $ host sip2.foo.bar. sip2.foo.bar has address 10.47.11.2 The following DNS options are set: dns_srv_lb = yes dns_try_naptr = yes dns_naptr_ignore_rfc = no The Kamailio version is 4.2.8 (installed from the Debian binary package) on Debian Wheezy. Shouldn’t ds_is_from_list be able to resolve the NAPTR and SRV records? Can I change anything to be able to use ds_is_from_list with the specified domain? Thanks, Björn Bylander

2 1

Upgrade from 4.3 to 4.4
by Laura 18 Jan '17

18 Jan '17

Dear list, I'm planning the upgrade fromn 4.3.6 tree to 4.4.x version on Debian distribution. Can i run the apt-get update and upgrade without problem.. or is possible that i will found some error during upgrade ? Is the db schema upgraded or not from 4.3 up to 4.4 ? Regards Laura

2 1

Please explain : t_relay() vs t_relay_to_udp()
by Abdoul Osséni 18 Jan '17

18 Jan '17

hello all, I want to use kamailio like as a proxy SIP (outbound proxy) to transcode tls to udp like us: call flow is: Softphone --> tls --> Kamailio server (kamailio listen on 443 port) --> udp --> Asterisk server (asterisk listen on 5060 port) I know Kamailio can convert tls to udp. According tm documentation http://kamailio.org/docs/modules/devel/modules/tm.html, I think I must use t_relay_to_udp(). Do you confirm I must replace t_relay() function by t_relay_to_udp() function? Thank for your help. Regards Abdoul.

2 2

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 18 Jan '17

18 Jan '17

Yes, this is a sipcapture node. I'm listening on a switch port that is set to mirror my VoIP traffic. I see all SIP UDP/TCP packets on the mirror port and the Ethernet port of the host node. Just don't see any TCP packets process in kamailio, debug 3. UDP packets are processed as expected. My config is using port mirror for the capture parameters: modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 0) modparam("sipcapture", "raw_ipip_capture_on", 0) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "raw_interface", "eth1") modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") modparam("sipcapture", "promiscious_on", 1) modparam("sipcapture", "raw_moni_bpf_on", 1) Is there a method I could diagnose if the SIP TCP Packets are getting from the kernel network process and the kamailio process? # ngrep -d eth1 -W byline host x.x.x.x | /var/run/kamailio/kamailio.pid Or pipe to kamailio local unix socket? I don't know, I'm just guessing. Thanks. JR > Somehow is not clear for me how you have the configuration there ... > before commenting further, this needs to be clarified. > > The node you presented the config is a sipcapture instance, right? What > is sending traffic to it? Is another kamailio with siptrace module? Or > the sipcature agent? Or you have a port mirroring in the router? > > Cheers, > Daniel > > > On 17/01/2017 16:37, JR Richardson wrote: >>> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >>>> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >>>> using raw socket capture, I think this may be a sipcapture issue, >>>> debuging kamailio shows normal startup and processing of UDP SIP >>>> packets, but does not show any activity with TCP packets. >>> I never used HOMER sofar but when I saw your first message my thoughts >>> was that this can't work in a simple way since for TCP you need to >>> complete a 4 way handshake before you can start to send data. >>> >> Interesting. Are you referring to handshaking on the network stack or >> SIP TCP TLS handshaking? I guess I can see it two ways. >> >> 1) if your talking about TCP/IP handshake, even though the SIP packet >> comes into the mirror port on the host node, the kernel processing the >> TCP packet is not establishing a valid connection due to no TCP >> handshake because its only a monitor port, no transmit back, then the >> kernel network stack does not pass the SIP TCP packet to the kamailio >> process for capture because it drops the packet due to no valid >> handshake? >> >> 2) the kernel network stack is passing the SIP TCP packet to the >> kamailio process, but since kamailio cannot handshake back it drops >> the packet and does not process through the sipcapture module. This >> kinda breaks the whole capture ability for homer with SIP TCP. Using >> ngrep, I see all SIP TCP packets, invite -->, trying <--, session >> progress <--, request timeout <--, ack -->, etc... >> >> So how would I diagnose if the network stack is the culprit? Debugging >> kamailio is pretty straight forward, setup and listening for SIP TCP, >> but never see any processing of any TCP packets. >> >> Thanks. >> >> JR > -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

Kamailio v4.4.5 Released
by Daniel-Constantin Mierla 18 Jan '17

18 Jan '17

Hello, Kamailio SIP Server v4.4.5 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.5. For more details about version 4.4.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/01/kamailio-v4-4-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Planning the release of v4.4.5
by Daniel-Constantin Mierla 18 Jan '17

18 Jan '17

Hello, I am considering to release v4.4.5 with latest fixes from branch 4.4 later this week, on Wednesday or Thursday (Jan 19 or 20). As usual, if anyone has fixes that were not backported yet, do it if you are a developer or write to sr-dev to be sure it is not missed. Sumbit also the new issues you are ware of and they are not on the bug tracker. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Record-Route Header Question
by Andres 17 Jan '17

17 Jan '17

Hi, I am troubleshooting some strange call setup problems using an Edgewater Networks SBC. A couple of Polycom phones sit behind the SBC and they connect to a SIP Server in the cloud. The SBC is in passthrough mode. What I see that I am not sure about is: INVITE from SIP Server to SBC IP (no Record-Route header...so far no surprise here) INVITE from SBC to Phone (Record-Route header inserted by SBC but with the IP of the SIP Server) I do not think the SBC is supposed to add a Record-Route header with the IP of the remote SIP Server. I think It should only be allowed to add Record-Route headers with its own IP. Does that sound right? The RFC lists the usage as: "20.30 Record-Route The Record-Route header field is inserted by proxies in a request to force future requests in the dialog to be routed through the proxy." It says nothing about being able to arbitrarily force future requests to go through a remote IP. Please advise. Thanks, -- Technical Support

2 2

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 17 Jan '17

17 Jan '17

> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >> using raw socket capture, I think this may be a sipcapture issue, >> debuging kamailio shows normal startup and processing of UDP SIP >> packets, but does not show any activity with TCP packets. > > I never used HOMER sofar but when I saw your first message my thoughts > was that this can't work in a simple way since for TCP you need to > complete a 4 way handshake before you can start to send data. > Interesting. Are you referring to handshaking on the network stack or SIP TCP TLS handshaking? I guess I can see it two ways. 1) if your talking about TCP/IP handshake, even though the SIP packet comes into the mirror port on the host node, the kernel processing the TCP packet is not establishing a valid connection due to no TCP handshake because its only a monitor port, no transmit back, then the kernel network stack does not pass the SIP TCP packet to the kamailio process for capture because it drops the packet due to no valid handshake? 2) the kernel network stack is passing the SIP TCP packet to the kamailio process, but since kamailio cannot handshake back it drops the packet and does not process through the sipcapture module. This kinda breaks the whole capture ability for homer with SIP TCP. Using ngrep, I see all SIP TCP packets, invite -->, trying <--, session progress <--, request timeout <--, ack -->, etc... So how would I diagnose if the network stack is the culprit? Debugging kamailio is pretty straight forward, setup and listening for SIP TCP, but never see any processing of any TCP packets. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

Using the auth_ephemeral module
by Hemanshu Vadehra 17 Jan '17

17 Jan '17

I'm involved in setting up a Kamailio instance and was hoping to make use of the auth_ephemeral module for authentication. But the module documentation doesn't quite make clear how exactly the module is to be employed or the web service set up. Does anyone have a working example? Regards, Hemanshu Vadehra hemanshu.v(a)directi.com

5 8

usrloc module with dbmode=2 does not insert records to database
by Vik Killa 17 Jan '17

17 Jan '17

Hello, we've noticed that the usrloc module does not "sync" all the records from memory into the database. I use a bash script to generate in-memory AoRs ( http://paste.debian.net/plain/908521) then i perform kamcmd ul.flush and no records are inserted. We have tried various usrloc parameters but none seem to work Here is our basic setup # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "timer_interval", 120) modparam("usrloc", "timer_procs", 4) We are using postgresql. are we missing something? Thanks /V

2 11

Info: github template for reporting issues
by Daniel-Constantin Mierla 17 Jan '17

17 Jan '17

Hello, yesterday I added a template for opening new issues on github.com bug tracker. You can see it in action at: - https://github.com/kamailio/kamailio/issues/new Its bare text can be seen at: - https://raw.githubusercontent.com/kamailio/kamailio/master/.github/ISSUE_TE… More about the issue templates for github can be found at: - https://help.github.com/articles/creating-an-issue-template-for-your-reposi… Beside informing about this addition, I would like to get also feedback about it. Do you find it useful or you think it's going to be a hassle? If it should be kept, is anything that you think it should be added or removed? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

LCR module dont_strip_or_tag_flag
by Annus Fictus 17 Jan '17

17 Jan '17

Hello, I'm trying LCR module and dont_strip_or_tag_flag parameter. If I use this line in the module configuration: modparam("lcr", "dont_strip_or_tag_flag", 10) and then restart kamailio, i receive this error: ERROR: <core> [modparam.c:141]: set_mod_param_regex(): parameter <dont_strip_or_tag_flag> of type <2> not found in module <lcr> I'm using 4.4.4 version Any hint? Thank you Regards

3 3

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 16 Jan '17

16 Jan '17

Hey Daniel, Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, using raw socket capture, I think this may be a sipcapture issue, debuging kamailio shows normal startup and processing of UDP SIP packets, but does not show any activity with TCP packets. I have enabled/disabled the different capture methods and no TCP SIP packets are processed. Also, I didn't mention before I'm testing on 2 different instances, one is a Homer 3.5 (Kamailio 4.1.2) and another is Homer 5.0 (Kamailio 4.4.0). I'll post to the Homer list and see what they say. Thanks. JR > is it this about a sipcature node? If yes, then it receives the packets > over HEP, meaning that is not going to the normal processing path. > Kamailio detects it is not a SIP packet, executes some callbacks for > non-sip packets registered by modules (in this case sipcapture). The > module detecting it's a packet for it, can execute the request_route > once it builds a local sip packet. > > To troubleshoot further, run with debug=3 in kamailio.cfg and see what > you get in the logs. > > Cheers, > Daniel > > On 13/01/2017 16:33, JR Richardson wrote: >> Iptables is not blocking, but it was worth a check. >> >> Thanks. >> >> JR >> >> >> I assume you have ruled out firewall? It's something that can nab even >> experienced people: >> >> # iptables -Ln >> >> -- Alex >> >> On Thu, Jan 12, 2017 at 03:25:27PM -0600, JR Richardson wrote: >> >>> Hi All, >>> >>> Just enabled SIP TCP on a homer capture server, I can see the SIP TCP >>> Sessions on the server with ngrep, just like all the UDP traffic. I >>> have Kamailio listening on TCP ports but its not capturing any TCP >>> traffic. >>> >>> kamailio.cfg: >>> >>> #disable_tcp=yes >>> listen=tcp:10.99.99.99:5060 #monitor port >>> listen=udp:10.99.99.99:5060 #monitor port >>> >>> loadmodule "pv.so" >>> loadmodule "db_mysql.so" >>> loadmodule "sipcapture.so" >>> loadmodule "textops.so" >>> loadmodule "rtimer.so" >>> loadmodule "xlog.so" >>> loadmodule "sqlops.so" >>> loadmodule "htable.so" >>> loadmodule "sl.so" >>> loadmodule "siputils.so" >>> >>> >>> modparam("sipcapture", "capture_on", 1) >>> modparam("sipcapture", "hep_capture_on", 0) >>> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") >>> modparam("sipcapture", "raw_interface", "eth1") >>> modparam("sipcapture", "raw_ipip_capture_on", 0) >>> modparam("sipcapture", "table_name", "sip_capture") >>> modparam("sipcapture", "raw_sock_children", 4) >>> modparam("sipcapture", "db_insert_mode", 0) >>> modparam("sipcapture", "raw_moni_capture_on", 1) >>> modparam("sipcapture", "promiscious_on", 1) >>> modparam("sipcapture", "raw_moni_bpf_on", 1) >>> modparam("sipcapture", "capture_node", "homer02") >>> modparam("sipcapture", "authorization_column", "authorization") >>> >>> >>> ## logging all INVITES top of the [route] block >>> if (is_method("INVITE|REGISTER")) { >>> xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\" >>> from \"$si\"\n"); >>> >>> Logging reports all SIP UDP traffic to logs fine, but no TCP traffic. >>> >>> root@homer02:~# netstat -al >>> Active Internet connections (servers and established) >>> Proto Recv-Q Send-Q Local Address Foreign Address State >>> tcp 0 0 homer02.me.com:sip *:* LISTEN >>> >>> >>> I don't think this is a homer issue because logging invites is prior >>> to any homer processing. I'm thinking this is something simple I'm >>> overlooking, any help is much appreciated. >>>

2 1

Remove/Append headers in 200 canceling and 200 ok to BYE
by Diego Nadares 16 Jan '17

16 Jan '17

Hi List, I'm trying to add and remove some headers in 200 canceling and 200 ok to BYEs with no success. I don't know in which route to add my remove_hf and append_hf. It seems that I can't catch it on reply either on request route. I tried adding it in event_route[tm:local-response] { xlog("L_ERR", "TM LOCAL REPONSE!"); route(MANAGE_HEADERS); } This is my route to manage hdrs: route[MANAGE_HEADERS]{ xlog("L_ERR", "MANAGE HDRS !"); keep_hf("TH"); # for topoh append("AHDR1:asdf\r\n"); return; } I suppose SERVER HDR should be removed but: U 2017/01/12 12:45:41.566733 172.16.213.38:5060 -> 172.16.50.41:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 172.16.50.41:5060;branch=z9hG4bKgg8ekd3080braugkm700.1. CSeq: 102 CANCEL. From: <sip:xx@172.16.50.41>;tag=SD7b10b01-as5ba5e269. To: <sip:xxx@172.16.213.38>;tag=4a86e96addfb27a943bc86e1872032b6-0a8b. Call-ID: SD7b10b01-6a7d1089793bda79481f25cf88c023ed-c540dl1. Server: kamailio (4.4.4 (x86_64/linux)). Content-Length: 0. Any help will be very appreciated. Thanks in advance! Diego

3 9

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 16 Jan '17

16 Jan '17

Iptables is not blocking, but it was worth a check. Thanks. JR I assume you have ruled out firewall? It's something that can nab even experienced people: # iptables -Ln -- Alex On Thu, Jan 12, 2017 at 03:25:27PM -0600, JR Richardson wrote: > Hi All, > > Just enabled SIP TCP on a homer capture server, I can see the SIP TCP > Sessions on the server with ngrep, just like all the UDP traffic. I > have Kamailio listening on TCP ports but its not capturing any TCP > traffic. > > kamailio.cfg: > > #disable_tcp=yes > listen=tcp:10.99.99.99:5060 #monitor port > listen=udp:10.99.99.99:5060 #monitor port > > loadmodule "pv.so" > loadmodule "db_mysql.so" > loadmodule "sipcapture.so" > loadmodule "textops.so" > loadmodule "rtimer.so" > loadmodule "xlog.so" > loadmodule "sqlops.so" > loadmodule "htable.so" > loadmodule "sl.so" > loadmodule "siputils.so" > > > modparam("sipcapture", "capture_on", 1) > modparam("sipcapture", "hep_capture_on", 0) > modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") > modparam("sipcapture", "raw_interface", "eth1") > modparam("sipcapture", "raw_ipip_capture_on", 0) > modparam("sipcapture", "table_name", "sip_capture") > modparam("sipcapture", "raw_sock_children", 4) > modparam("sipcapture", "db_insert_mode", 0) > modparam("sipcapture", "raw_moni_capture_on", 1) > modparam("sipcapture", "promiscious_on", 1) > modparam("sipcapture", "raw_moni_bpf_on", 1) > modparam("sipcapture", "capture_node", "homer02") > modparam("sipcapture", "authorization_column", "authorization") > > > ## logging all INVITES top of the [route] block > if (is_method("INVITE|REGISTER")) { > xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\" > from \"$si\"\n"); > > Logging reports all SIP UDP traffic to logs fine, but no TCP traffic. > > root@homer02:~# netstat -al > Active Internet connections (servers and established) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp 0 0 homer02.me.com:sip *:* LISTEN > > > I don't think this is a homer issue because logging invites is prior > to any homer processing. I'm thinking this is something simple I'm > overlooking, any help is much appreciated. > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses > Chasing the Azeotrope > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 2

Kamailio segfault on TM module
by José Seabra 16 Jan '17

16 Jan '17

Hi there, I have noticed a segfault on my kamailio server but as i have coredumps disabled and i don't know how i can reproduce it, i can't provide that kind of information. Anyway i don't know if anyone of you can help with the following information: segfault at 600 ip 00007f5cb97af1ea sp 00007ffed96b8d60 error 6 in tm.so[7f5cb9718000+11d000] ERROR: tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 404 reply when a final 200 was sent out ERROR: sl [sl.c:269]: send_reply(): failed to reply stateful (tm) WARNING: db_postgres [km_dbase.c:281]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] ALERT: <core> [main.c:739]: handle_sigs(): child process 2048 exited by a signal 11 CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 14 ALERT: <core> [main.c:742]: handle_sigs(): core was not generated INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received Database server was down during this segfault, but the segfault was on TM module not on database module. Thank you Regards José Seabra

2 1

dbtext module issue
by Satish Patel 16 Jan '17

16 Jan '17

I am trying use dbtext module instead of mysql and following is my configuration. We are using old SER version because of custom written module. Do you think following config looks good. # -- permissions params -- modparam("permissions", "db_url", "dbtext:///etc/ser/dbtext") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") In function if ( !allow_trusted() && !search ("^User-Agent:.*Some Perl client.*") ) { xlog("L_WARN", "Drop if User-Agent isn't Perl client"); drop; break; }; My dbtest contents $ cat /etc/ser/dbtext/trusted src_ip(str) proto(str) from_pattern(str) 10.0.2.20:any:.* When i run ser in fork=no mode i am seeing following ser -f ser.cfg Content of result table_version(int) 2 Content of [trusted] src_ip(str) proto(str) from_pattern(str) 10.0.2.20:any:.* Content of result proto(str) from_pattern(str) "any" ".* " Content of result proto(str) from_pattern(str) "any" ".* " it look like it read dbtext file and load data but when i use sipp to generate some call then i am seeing following result only, that means its not loading IP address from table, why its extracting two field but not src_ip ? Content of result proto(str) from_pattern(str) "any" ".* "

2 2

comma as param.count delimiter produces parse error
by Juha Heinanen 15 Jan '17

15 Jan '17

Parameter list transformation param.count produces parse error when delimiter is comma (,) character. >From document: {param.value,name[, delimiter]} 'delimiter' allows you to specify a single character to use as the parameter delimiter. For example, when parsing HTTP URL query strings use '&'. When I have in config statement: $var(count) = $(var(servers){param.count,,}); I get to syslog: Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: pv [pv_trans.c:2645]: tr_parse_paramlist(): invalid separator in transformation: count,,} Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: <core> [core/pvapi.c:1629]: tr_lookup(): error parsing [{param.count,,}] Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: <core> [core/pvapi.c:1010]: pv_parse_spec2(): bad tr in pvar name "var" Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: <core> [core/pvapi.c:1036]: pv_parse_spec2(): invalid parsing in [$(var(servers){param.count,,})] at (4) Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) : <core> [core/cfg.y:3407]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 6426, column 19-48: Can't get from cache: $(var(servers){param.count,,}) Jan 13 07:48:15 lohi sip-proxy[18000]: ERROR: bad config file (1 errors) -- Juha

2 2

Kamailio not processing SIP TCP
by JR Richardson 12 Jan '17

12 Jan '17

Hi All, Just enabled SIP TCP on a homer capture server, I can see the SIP TCP Sessions on the server with ngrep, just like all the UDP traffic. I have Kamailio listening on TCP ports but its not capturing any TCP traffic. kamailio.cfg: #disable_tcp=yes listen=tcp:10.99.99.99:5060 #monitor port listen=udp:10.99.99.99:5060 #monitor port loadmodule "pv.so" loadmodule "db_mysql.so" loadmodule "sipcapture.so" loadmodule "textops.so" loadmodule "rtimer.so" loadmodule "xlog.so" loadmodule "sqlops.so" loadmodule "htable.so" loadmodule "sl.so" loadmodule "siputils.so" modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 0) modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") modparam("sipcapture", "raw_interface", "eth1") modparam("sipcapture", "raw_ipip_capture_on", 0) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "db_insert_mode", 0) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "promiscious_on", 1) modparam("sipcapture", "raw_moni_bpf_on", 1) modparam("sipcapture", "capture_node", "homer02") modparam("sipcapture", "authorization_column", "authorization") ## logging all INVITES top of the [route] block if (is_method("INVITE|REGISTER")) { xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\" from \"$si\"\n"); Logging reports all SIP UDP traffic to logs fine, but no TCP traffic. root@homer02:~# netstat -al Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State tcp 0 0 homer02.me.com:sip *:* LISTEN I don't think this is a homer issue because logging invites is prior to any homer processing. I'm thinking this is something simple I'm overlooking, any help is much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

C++ module
by Patrick 12 Jan '17

12 Jan '17

I added a C++ module to kamalio at some point, and now I am trying to integrate that module with another C++ internal library, but I'm getting all sorts of headaches when trying to compile. The latest problem is during the "make install" part of the build - it seems like the objects from the module are "infecting" the link line of the internal library it's dependent on. Here is the Makefile for the module. At the time, the only way I figured to link in the C++ object was adding them to the LIBS. Now, i think this interferes with the 'make install' dependencies. Any thoughs on how to do this properly would be great! Thanks *Module Makefile* # # db_cassandra module makefile # # # WARNING: do not run this directly, it should be run by the master Makefile include ../../Makefile.defs CXX=g++ LD=g++ auto_gen= NAME=db_arl.so ifeq ($(CROSS_COMPILE),) CURL_BUILDER=$(shell \ if pkg-config --exists libcurl; then \ echo 'pkg-config libcurl'; \ else \ which curl-config; \ fi) endif DEFS+=-DKAMAILIO_MOD_INTERFACE -DRTC_COMMON_TRACE *LIBS += db_arl_base.o jsoncpp.o* -lcurl -pthread CXXFLAGS=$(CFLAGS:-Wno-deprecated option=) CXXFLAGS+= -Wno-write-strings -Wno-deprecated -Wno-unused-function -Wno-sign-compare -Wno-strict-aliasing CXXFLAGS+= -I ${COL_LOCAL_BUILDS_ROOT}/ExternalLibs/boost/V1.60.0_27/ CXXFLAGS+= -I ./jsoncpp-dist/ SERLIBPATH=../../lib SER_LIBS+=$(SERLIBPATH)/srdb2/srdb2 SER_LIBS+=$(SERLIBPATH)/srdb1/srdb1 *SER_LIBS+=$(SERLIBPATH)/rtctrace/rtctrace* include ../../Makefile.modules jsoncpp.o: jsoncpp-dist/jsoncpp.cpp @echo COL_LOCAL_BUILDS_ROOT=${COL_LOCAL_BUILDS_ROOT} @echo "Compiling $<" $(CXX) $(CXXFLAGS) $(CFLAGS) $(C_DEFS) $(DEFS) -c $< -o $@ db_arl_base.o: db_arl_base.cpp db_arl_base.h @echo COL_LOCAL_BUILDS_ROOT=${COL_LOCAL_BUILDS_ROOT} @echo "Compiling $<" $(CXX) $(CXXFLAGS) $(CFLAGS) $(C_DEFS) $(DEFS) -c $< -o $@ db_arl.so: db_arl_base.o jsoncpp.o During 'make install' the objects in *red* have no business in the link line for the internal library /opt/rh/devtoolset-4/root/usr/bin/g++ -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -mtune=opteron -Wall -I/home/pshea/_localbuilds/RTC/Common/2.0.0_247//Applications_rhel6_x64_dts4_pic/include -DNAME='"kamailio"' -DVERSION='"4.2.4"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.4.7"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=4002004 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DDBG_F_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -std=c++0x -I/home/pshea/_localbuilds/RTC/Common/2.0.0_247//Applications_rhel6_x64_dts4_pic/include -c rtctrace.c -o rtctrace.o -MMD -MP /opt/rh/devtoolset-4/root/usr/bin/g++ -shared -m64 -Wl,-O2 -Wl,-E -Wl,-soname,librtctrace.so.1 rtctrace.o * db_arl_base.o jsoncpp.o* -lcurl -pthread -L/home/pshea/_localbuilds/RTC/Common/2.0.0_247//Applications_rhel6_x64_dts4_pic/lib -lrtccommon -L/home/pshea/_localbuilds/ExternalLibs/boost/V1.60.0_22/Server_Linux26_x64_dts4_pic/lib -lboost_system -lboost_thread -lpthread -o librtctrace.so.1.0 *g++: error: db_arl_base.o: No such file or directory* *g++: error: jsoncpp.o: No such file or directory*

2 7

Difference between HTTP clients (utils.so vs http_client.so)
by Joel Serrano | VOZELIA 12 Jan '17

12 Jan '17

Hi all, Currently I see there are 2 http clients: utils.so --> http_query() http_client.so --> http_client_query() Any advice on which to use? Any pros/cons of using one over the other? Thanks, Joel. Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado.Les informamos que los datos contenidos en el presente mail, han sido facilitados por usted o recogidos de Fuentes Accesibles al Público. Dicha información personal, será incluida en un fichero bajo la responsabilidad de VOZELIA TELECOM S.L.Todos los ficheros que trata nuestra empresa se encuentran debidamente inscritos ante la Agencia Española de Protección de Datos.Vd. Como propio interesado podrá ejercitar sus derechos de Acceso, Cancelación, Rectificación u Oposición, en el mail: administracion(a)vozelia.com o en la dirección Avenida de Manoteras 8 escalera 3 2ºB, 28050 Madrid.Tal y como establece la Ley Orgánica de Protección de Datos 15/1999 de 13 de Diciembre.Para ejercer sus derechos desde Panama podrá hacerlo en la dirección Vía Argentina, Edificio Mediteranean Loft. Panama.En caso de no ser el destinatario y haya recibido este mensaje por error, agradeceremos que nos lo comunique inmediatamente al remitente sin difundir, almacenar o copiar su contenido.La divulgación o el suministro, en todo o en parte, a cualquier tercero, no podrá ser realizada sin el previo, expreso y escrito consentimiento de VOZELIA TELECOM S.L. .

3 4

RPC exact command to change voicemail IP or Port
by Annus Fictus 12 Jan '17

12 Jan '17

Hello list, I can't find the exact RPC command to change IP or port of voicemail server (module CFG_RPC): the default Kamailio configuration is: */#!ifdef WITH_VOICEMAIL/**/ /**/# VoiceMail Routing on offline, busy or no answer/**/ /**/#/**/ /**/# - by default Voicemail server IP is empty to avoid misrouting/**/ /**/voicemail.srv_ip = "" desc "VoiceMail IP Address"/**/ /**/voicemail.srv_port = "5060" desc "VoiceMail Port"/**/ /**/#!endif/* Thank you Regards

2 8

how to immediately close a tls connection
by Andy 12 Jan '17

12 Jan '17

Hi, trying to close a tls connection immediately, have ref to the connection (tcpconn) pointer please suggest, thanks! -- View this message in context: http://sip-router.1086192.n5.nabble.com/how-to-immediately-close-a-tls-conn… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio and default paramenters to a specific gatewy
by Laura 11 Jan '17

11 Jan '17

Dear all, I'm using kamailio 4.3.6 and LCR modules.. i was wondering if it was possibile to add some special paramenter to all call sent to a specific gateway. I was try to use the " params" field on lcr_gw table but it's not working like i was think.. The issue is simple.. i want to add to field "Privacy: id" to any call i send over a specific gateway.. What is the best and easy way to do that ? Any idea is appreciate. Regards Laura

3 9

Diameter AVP handling from kamaiio.cfg
by Zoltán Miricz 10 Jan '17

10 Jan '17

Dear All, I'm wondering is it possible to catch Diameter Messages and their AVP values from kamailio.cfg just like for SIP messages? (Using pseudo variables) I looked into this function from the attached S-CSCF example. route[REG_MAR_REPLY] { #this is async so to know status we have to check the reply avp xlog("L_DBG","maa_return code is *$avp(s:maa_return_code)*\n"); Sadly it gives back only 1, -1, -2. (which is based on cxdx_mar.c the rc) What I mainly looking for is the EXPERIMENTAL_RESULT (experimental_rc) and some different AVP values besides. Thank you very much in advance! *Cheers,* *Zoltan*

2 5

Kamailio on Cluecon Weekly – Jan 11, 2017
by Daniel-Constantin Mierla 10 Jan '17

10 Jan '17

Hello, for those interested in participating to a (video) chat tomorrow, Wed, Jan 11, I will be joining the ClueCon Weekly at 12:00CT (USA) to discuss about latest Kamailio, what's new and the plans for the near future. Integration with FreeSwitch is going to be among the topics. You can join the video conference call with a webrtc capable browser. You can also join only for audio using a SIP phone or dialing from PSTN or mobile networks. More specific details are available at: - https://www.kamailio.org/w/2017/01/kamailio-on-cluecon-weekly-jan-11-2017/ Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Daniel-Constantin Mierla 10 Jan '17

10 Jan '17

Hello, are you using topoh module? A pcap with all sip messages part of the dialog, from the initial INVITE to the last BYE transaction would be important to have in order to investigate how the headers are changed. Cheers, Daniel On 28/12/2016 13:11, Nihar Ranjan Deb wrote: > > Hi All, > > > > I am a new user of kamailio. We have setup a kamilio server with RTP > proxy. Most of the features are working fine except BYE. > > > > > > Say User A and B are registered in kamailio server. A (caller) > connected to B(called), call established. > > > > 1. If Caller (A) disconnects the call, everything goes fine. B gets > disconnected and call releases. > > > > 2. But if Called party (B) disconnects the call, A never gets > disconnected. I checked trace as well, in this case caller (A) gets a > BYE with B's callid. So A rejects call with "481 call leg transaction > doesn't exists". > > Basically A party and B party has separate dialog established with > Kamilio. So they have two separate callids . All messages are > exchanged correctly. Only when B party disconnects call, kamailio > sends BYE to A party with B party callid. > > > > Please help me to resolve this issue. Do let me know if any more > information required. > > > > > > ------ INVITE-------------------------- > > > > INVITE sip:966268@*** SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport > > Max-Forwards: 70 > > Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> > > To: <sip:966268@***> > > From: "334031"<sip:334031@****>;tag=4406552c > > Call-ID: *82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc* > > CSeq: 2 INVITE > > Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, > OPTIONS, MESSAGE > > Content-Type: application/sdp > > Proxy-Authorization: Digest > username="334031",realm="*****",nonce="****",uri="sip:966268@***",response="***",algorithm=MD5 > > Supported: replaces > > User-Agent: X-Lite release 4.9.6 stamp 82158 > > Content-Length: 334 > > > > v=0 > > o=- 13127393523589215 1 IN IP4 192.168.1.36 > > s=X-Lite release 4.9.6 stamp 82158 > > c=IN IP4 192.168.1.36 > > t=0 0 > > m=audio 57116 RTP/AVP 9 8 120 0 84 101 > > a=rtpmap:120 opus/48000/2 > > a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 > > a=rtpmap:84 speex/16000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=sendrecv > > > > > > > > -------------------------BYE-------------------------------- > > > > > > BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 > > Via: SIP/2.0/UDP > ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 > > Via: SIP/2.0/UDP > 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* > > From: <sip:966268@***>;tag=1665200125 > > To: "334031" <sip:334031@***>;tag=4406552c > > Call-ID: > *!!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRweMH4N6RemHYMI9Y0tV*** > > CSeq: 3 BYE > > Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> > > Max-Forwards: 69 > > User-Agent: Yealink SIP-T20P 9.72.0.80 > > Content-Length: 0 > > > > > > -------------------------- 481--------------------- > > > > SIP/2.0 481 Call/Transaction Does Not Exist > > Via: SIP/2.0/UDP > ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.164.228.12 > > Via: SIP/2.0/UDP > 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* > > To: "334031" <sip:334031@***>;tag=4406552c > > From: <sip:966268@***>;tag=1665200125 > > Call-ID: > !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRweMH4N6RemHYMI9Y0tV** > > CSeq: 3 BYE > > Accept-Language: en > > Content-Length: 0 > > > > > > > > > > Regards, > > Nihar > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

[Error] Install Kamailio 4.4 in Mac 10.12
by Luis De Marchi 10 Jan '17

10 Jan '17

Hi, Sorry, my english, I'm using the translator. I'm having trouble installing Kamailio for Mac 10.12 Terminal: Luis:kamailio luisdemarchi$ sudo make all CC (gcc) [M db_mysql.so] km_db_mysql.o km_db_mysql.c:41:10: fatal error: 'mysql.h' file not found #include <mysql.h> ^ 1 error generated. make[1]: *** [km_db_mysql.o] Error 1 make: *** [modules] Error 1 Luís De Marchi Diretor Administrativo +55 (41) 99815-6800 +55 (41) 3027-7669 www.snowmanlabs.com <http://www.snowmanlabs.com/>

4 4

t_precheck_trans
by Annus Fictus 10 Jan '17

10 Jan '17

Hi, where can I find documentation for this function present on the default script? In TM module not exist (Kamailio 4.4.4) Thank you Regards

3 3

use case
by Thufir Hawat 09 Jan '17

09 Jan '17

I'm ok with Linux and am dipping my toe into Asterisk by running an AWS EC2 instance. What is the most frequent usage for kamailio? Would it mainly be for SIP to SIP? So that the URI would be for a specific domain? thanks, Thufir

3 2

using kamailio to forward ipv6 to ipv4
by anfecora 09 Jan '17

09 Jan '17

Hi guys hope you have a great new year. I would appreciate if anyone can point me in to the right direction . I need to build a proxy to translate from ipv6 to ipv4, but kamailio should not process the registers or invites, only translate from ipv6 to ipv4 and forward all registrations over. is that possible? thanks.

4 5

why a register error?
by Olvera, Victor 09 Jan '17

09 Jan '17

As I attempt to use X-lite I get a 504 error. On the CSCFs I get the following: pcscf 0(2066) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2066) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.1:55880) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) icscf 11(2150) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 0(2138) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2138) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.76:4060) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) 0(2138) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 0 scscf 11(2047) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 192.168.56.78 port 3868 11(2047) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 9(2045) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket Great regards, Victor Olvera victor.olvera(a)verizonwireless.com<mailto:victor.olvera@verizonwireless.com> mobile # 001-214-923-9405

2 1

Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Nihar Ranjan Deb 09 Jan '17

09 Jan '17

Anyone any help on this. Regards, Nihar From: Nihar Ranjan Deb [mailto:nihar.ranjan@microtalkgroup.com] Sent: Wednesday, December 28, 2016 5:42 PM To: 'sr-users(a)lists.sip-router.org' Subject: BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists". Hi All, I am a new user of kamailio. We have setup a kamilio server with RTP proxy. Most of the features are working fine except BYE. Say User A and B are registered in kamailio server. A (caller) connected to B(called), call established. 1. If Caller (A) disconnects the call, everything goes fine. B gets disconnected and call releases. 2. But if Called party (B) disconnects the call, A never gets disconnected. I checked trace as well, in this case caller (A) gets a BYE with B's callid. So A rejects call with "481 call leg transaction doesn't exists". Basically A party and B party has separate dialog established with Kamilio. So they have two separate callids . All messages are exchanged correctly. Only when B party disconnects call, kamailio sends BYE to A party with B party callid. Please help me to resolve this issue. Do let me know if any more information required. ------ INVITE-------------------------- INVITE sip:966268@*** SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport Max-Forwards: 70 Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> To: <sip:966268@***> From: "334031"<sip:334031@****>;tag=4406552c Call-ID: 82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc CSeq: 2 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="334031",realm="*****",nonce="****",uri="sip:966268@***",response=" ***",algorithm=MD5 Supported: replaces User-Agent: X-Lite release 4.9.6 stamp 82158 Content-Length: 334 v=0 o=- 13127393523589215 1 IN IP4 192.168.1.36 s=X-Lite release 4.9.6 stamp 82158 c=IN IP4 192.168.1.36 t=0 0 m=audio 57116 RTP/AVP 9 8 120 0 84 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -------------------------BYE-------------------------------- BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* From: <sip:966268@***>;tag=1665200125 To: "334031" <sip:334031@***>;tag=4406552c Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> Max-Forwards: 69 User-Agent: Yealink SIP-T20P 9.72.0.80 Content-Length: 0 -------------------------- 481--------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.1 64.228.12 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* To: "334031" <sip:334031@***>;tag=4406552c From: <sip:966268@***>;tag=1665200125 Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Accept-Language: en Content-Length: 0 Regards, Nihar

1 0

VERY IMPORTANT: deciding when to remove the MI code
by Daniel-Constantin Mierla 06 Jan '17

06 Jan '17

Hello, we started discussing about removing MI (so called management interface) for very long time, more or less since 2008. The RPC should remain the control interface, given its better structure for commands, parameters, etc ... MI is custom protocol using a line-oriented communication via fifo or socket file with kamailio (e.g., implemented mi_fifo and mi_datagram modules). RPC is the alternative, a more standardized concept, with better structured format. I think it's time to set a clear roadmap for doing the removal. Overall, it will be easier to maintain the code, right now being duplicated code for doing the same operation over MI or RPC, and MI shows its limitations (or complexity to deal with) for advanced needs (see the discussions about how to provide multi-line value parameters over MI). So, I want to know if there are many relying on MI directly and they still want to keep it, what would be the expected duration they need for upgrading their tools to work with RPC interface, other relevant aspects people have in favour of mi vs rpc. I am even willing to do the removal in time befire freezing the 5.0 branch. We will ensure a clean start of 5.x series. The main concern from my point of view is kamctl -- but I think we can preserve the compatibility for kamctl commands and parameters (so command line execution of kamctl will be the same), but the output might be different. That's because it should be easy to updated it to communicate with jsonrpc-s module, but then it will get json-formatted results. To summarize, two big questions to answer: a) Are you ok to remove the MI code/commands? b) If yes to a), are you ok to be done for v5.0? Not providing feedback will be considered as 'yes' for both questions, so **speak up if you want MI to be kept or delay it removal**. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

6 30

presentity tabla, faster update when drop TCP/TLS connection
by João Resende 05 Jan '17

05 Jan '17

I'm testing kamailio with a SIP client with presence support, when I get a connection drop from a client (i.e when the client lost the access to the Internet connection) I do not receive the change in the status to offline immediately, I just received when the timeout from the presentity database finishes. Exists a way to define this? or it is a bug? or should be a new feature? I know that I'm able to change the expiry parameter from the presentity table, but that increases my battery consumption and the data exchange, the same happens if I reduce the publish interval on the client side.

3 9

Re: [SR-Users] using kamailio to forward ipv6 to ipv4
by anfecora 05 Jan '17

05 Jan '17

Yes i am planning on proxing media. The case is that client will only talk ipv6 but the registrar only have ipv4 connectivity. Therefore need to create a proxy to translate this dialogs. Thi thnking on using kamailio to receive the registration modify the pckets and forward registrations in ipv4. I was thinking in using the asterisk kamailio example but instead of asterisk behind will be an sbc. Therefore kamailio will listen in ipv6 and translate over to ipv4 including media of course via rtpproxy. On Jan 5, 2017 9:43 AM, "Frank Carmickle" <frank(a)carmickle.com> wrote: > On Jan 5, 2017, at 12:24 PM, anfecora <anfecora(a)gmail.com> wrote: > > Hi guys hope you have a great new year. > I would appreciate if anyone can point me in to the right direction . > > I need to build a proxy to translate from ipv6 to ipv4, but kamailio should not process the registers or invites, only translate from ipv6 to ipv4 and forward all registrations over. A bit more information about the situation would be helpful. What exactly are you trying to do? > is that possible? 6 to 4 translation is possible, but until we understand the situation I’m not sure we can be that helpful. Are these phones? If so are they dual stack? Is the underlying network dual stack? There’s a lot of work to do to get this right in a Kamailio config. You’ll need to run rtpengine to proxy the media if the endpoints are not dual stack. Who keeps the mapping of the v6 to v4 and v4 to v6 translations? How is that mapping determined? I’d recommend ds-lite if you are in a position to alter the network and the phones/systems allow for it. HTH —FC > thanks. > > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

CDR - acc_cdrs
by Annus Fictus 05 Jan '17

05 Jan '17

Hello, I'm trying to save Kamailio CDR on acc_cdrs table without success. My relevant configuration look like: ----- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 ----- #!define WITH_ACCDB ---- loadmodule "dialog.so" loadmodule "acc.so" modparam("dialog", "dlg_flag", 0) ---- # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) 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) #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_log_enable", 1) modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$td") #!endif Any hint is really appreciate. Regards

2 4

unexpected char - status 6
by AL S 05 Jan '17

05 Jan '17

Hi, Kamailio is throwing error message from "parse_to.c" with status=6. The source code shows status 6 as URI_ENCLOSED. However, tcpdump of the call does not show any enclosed angle bracket is missing. (tcpdump pcap file is attached.) Kamailio log Jan 4 10:18:21 orl-ar-openser01 /usr/sbin/kamailio[2374]: ERROR: <core> [parser/parse_to.c:626]: ERROR: parse_to : unexpected char [#015] in status 6: <<<sip:2057030997@72.31.12.161;tag=gK04001cd3>> . FROM and TO in INVITE received by Kamailio From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161>;tag=gK04001cd3 FROM and TO in INVITE forward by Kamailio for the same call From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161\000\000>;tag=gK04001cd3 PS: this error message is happening on all 4 kamailio servers (3.3.0) PS: call failure only happens in re INVITE (call stays active longer than 15 min) Your help would be appreciated. Thanks, AS

2 2

RTPProxy with VPN
by Rodrigo Moreira 05 Jan '17

05 Jan '17

Hello, I am having problems with RTPPROXY. I need it to work as a bridge. On one side will have a VPN tunnel called a tun0, on the other an eth0 interface. Is it possible for RTPPROXY to run correctly in this scenario? How can I test if RTPPROXY is running correctly? Is there any command? What would the configuration look like inside /etc/default/rtpproxy Please help me. Best regards -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Re: [SR-Users] Regarding usage of Redis Module in Kamailio core
by Vicente Hernando 05 Jan '17

05 Jan '17

Hello Hemanth, this minimal kamailio cfg file works for me to test ndb_redis module. #!KAMAILIO debug=4 log_stderror=yes children=4 fork=yes #disable_tcp=yes mpath="modules/" loadmodule "ndb_redis.so" # An already configured redis server is needed here modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379;db=0") request_route { redis_cmd("srv1", "SET foo bar", "r"); exit; } Changing the parameters to suit your configuration, you can use it to debug your system. Regards, Vicente. On 01/05/2017 07:55 AM, hemanth(a)advaitamtech.com wrote: > > Hello Vicente, > > Thanks for your response, > > In response to your query, > > Yes i have Built ndb_redis module and i can see ndb_redis.so in /usr/local/lib64/kamailio/modules folder, I have installed hiredis depenency also, > > I have also modified the kamailio.cfg file loadmodule and modparam list, > > I feel like the way i am using the redis_cmd is wrong, I think i need to add it in route section of kamailio.cfg file, Can you please guide me regarding how to use redis db in order to update user information to DB on registration of each user. > > Thanks & Regards > Hemanth > > > Message: 8 > Date: Wed, 4 Jan 2017 14:41:42 +0530 (IST) > From: "hemanth(a)advaitamtech.com" <hemanth(a)advaitamtech.com> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] Regarding usage of Redis Module in Kamailio core > Message-ID: <1483521102.18025940(a)apps.rackspace.com> > Content-Type: text/plain;charset=UTF-8 > > Hello, > > I am using Kamailio as a SIP proxy, > > I want to store Registered user information in Redis DB, > > I am able to configure REDIS DB, But i am not able to run the redis command from > > Kamailio core using the function > > "redis_cmd("srvN","SET key 1", "r")" > > I am getting undefined reference error as below > > "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" > > can anyone please help me to fix this issue > > Thanks & Regards > Hemanth > > > > > ------------------------------ > > Message: 9 > Date: Wed, 4 Jan 2017 10:50:05 +0100 > From: Vicente Hernando <vhernando(a)systemonenoc.com> > To: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Regarding usage of Redis Module in Kamailio > core > Message-ID: <b003d2aa-ddd0-574d-b9e3-9b91e44976f1(a)systemonenoc.com> > Content-Type: text/plain; charset=utf-8; format=flowed > > Hello Hemanth, > > just a quick response, I would check if you compiled ndb_redis module > correctly (also if you have hiredis library dependency). > > Also I would check if you loaded ndb_redis module in your configuration > file. > > > Regards, > Vicente. > > On 01/04/2017 10:11 AM, hemanth(a)advaitamtech.com wrote: >> Hello, >> >> I am using Kamailio as a SIP proxy, >> >> I want to store Registered user information in Redis DB, >> >> I am able to configure REDIS DB, But i am not able to run the redis command from >> >> Kamailio core using the function >> >> "redis_cmd("srvN","SET key 1", "r")" >> >> I am getting undefined reference error as below >> >> "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" >> >> can anyone please help me to fix this issue >> >> Thanks & Regards >> Hemanth >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > ------------------------------ > > End of sr-users Digest, Vol 140, Issue 3 > **************************************** > > > >

1 0

Kamailio cancel branch only after receiving replies
by Aqs Younas 05 Jan '17

05 Jan '17

Greetings list, I am forking a call to multiple destinations and want to keep the only branch which sends quicker first 180/183 reply and cancel the remaining branches. Below is my related cfg snippet. modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("tm", "cancel_b_method", 2) ... route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); append_branch(); append_branch(); route(RELAY); exit; } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { xlog("L_INFO","Received $rs (IP:$si:$sp)\n"); if(status=~"18[03]"){ t_cancel_branches("others"); xlog("L_INFO","cancelled all other branches\n"); } route(NATMANAGE); } } ... But I see kamailio does not instantly send CANCEL to other branches after it has received 180/180 from any branch. Kamailio is sending CANCEL to other branches after they start sending 180/183 one by one. How can I cancel all other branches instantly? I am sure there is something wrongly configured in my configuration. Any suggestion is much appreciated. Best Regards.

3 3

location
by Slava Bendersky 05 Jan '17

05 Jan '17

Hello Everyone, Why save location will ignore on pv var when trying save on reply 200 OK. if(is_method("REGISTER") && fnmatch("$rs", "200")) { xlog("L_INFO", "OnReply: Registration is $rs OK saving --> $avp(rcvcontact) in db\n"); $var(oar) = "sip:" + $tU + "@" + $(avp(rcvcontact){uri.host}) + ":" + $(avp(rcvcontact){uri.port}); if(!save("location", "0x04", "$var(oar)")) { xlog("L_INFO", "Error saving the location\n"); } Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=619 a=26 n=xlog Dec 26 11:43:20 cavprx00 kamailio: 4(5546) INFO: <script>: OnReply: Registration is 200 OK saving --> sip:4300@client_pub_ip:31376;transport=tls in db Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=620 a=63 n=assign Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=624 a=16 n=if Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=621 a=27 n=save Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-1] (1 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-2] (2 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-2 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1004]: db_update_ucontact_ruid(): ruid:uloc-5860b463-10c5-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1133]: db_update_ucontact_ruid(): contact:sip:4300@10.18.130.27:5060;transport=udp ----> LAN side contact Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 10 chars, out: 10 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 21 chars, out: 21 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 33 chars, out: 33 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 40 chars, out: 40 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 20 chars, out: 20 chars Slava.

2 7

Kamailio v5.0 - development frozen
by Daniel-Constantin Mierla 05 Jan '17

05 Jan '17

Hello, development (master) branch of Kamailio enters now in pre-release phase for version 5.0.0. Therefore, no new feature should be pushed to master until we create a dedicated branch for 5.0 (expected to be in about 4 weeks or so). If in doubt to push or not a commit to master, push it first on a personal branch (or attach to an email) and discuss it on sr-dev. The new modules can be a bit more dynamic if there is need to get them to the right shape (e.g., like decision to rename functions, parameters or adjust database structure). We hope to get many people involved in testing, to reach a stable state before releasing 5.0.0. If you want to get involved and need assistance, don't hesitate to write to mailing lists. Besides the new features, there were two major changes for 5.0: 1) source code tree restructuring - this should not affect the stability of the code, only installation scripts or packaging may still need tuning 2) mi (management/control interface) code has been removed. SIP routing code should not be affected by this change that much, but testing of RPC commands needs a special care. There are few RPC commands not ported yet from the MI code, they can be done during the testing period Moreover, help with updating the wiki page for migration from 4.4 to 5.0 as well as what is new in 5.0 is very appreciated. I will try post updates about them very soon. Many thanks to everyone involved in development of 5.0 and the early testers that played with master branch so far. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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unexpected char [#015]
by AL S 04 Jan '17

04 Jan '17

Hi, Kamailio is complaining about unexpected char [#015] in TO header of INVITE: Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: <core> [parser/parse_to.c:626]: ERROR: parse_to : unexpected char [#015] in status 6: <<<sip:2057030997@72.31.12.161;tag=gK006cb08b>> . Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: <core> [parser/msg_parser.c:180]: ERROR: get_hdr_field: bad to header Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:2057030997@] Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: tm [t_lookup.c:1118]: ERROR: reply cannot be parsed Here is the TO header for the same call in both text and hex format. I am unable to see any ^M char in this TO header. To: <sip:2057030997@72.31.12.161;tag=gK006cb08b 546f3a203c7369703a323035373033303939374037322e33312e31322e3136313e3b7461673d674b30303663623038620d0a Any help would be appreciated. AS,

2 2

IMPORTANT: MI code removed (or disabled)
by Daniel-Constantin Mierla 04 Jan '17

04 Jan '17

Hello, as of now, no active MI reladed code should be in master branch -- the internal library kmi and the mi_datagram, mi_fifo, mi_xmlrpc, mi_rpc and pua_mi modules were removed. Couple of modules still have some mi code disabled with ifdefs, they are pending the port to RPC commands. These are: - carrierroute - ims_dialog - mohqueue - p_usrloc - pdb - rtpengine - userblacklist - utils They should be updated in the next few days. With MI code removed, the code is slimmer and the development effort is reduced -- in the recent past there were various issue rising due to usage of the old line-based MI commands. >From now on, the RPC interface has to be used for interacting with kamailio at runtime. The tools: - kamctl - was already migrated to use RPC via jsonrpcs module. - kamcmd - was developed to use only RPC interface via ctl module. - kamcli - is also working with RPC via jsonrpcs module Helping with testing the RPC commands is very appreciated -- open an issue on bug tracker whenever you discover a problem. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Re: [SR-Users] sr-users Digest, Vol 140, Issue 3
by hemanth＠advaitamtech.com 04 Jan '17

04 Jan '17

Hello Vicente, Thanks for your response, In response to your query, Yes i have Built ndb_redis module and i can see ndb_redis.so in /usr/local/lib64/kamailio/modules folder, I have installed hiredis depenency also, I have also modified the kamailio.cfg file loadmodule and modparam list, I feel like the way i am using the redis_cmd is wrong, I think i need to add it in route section of kamailio.cfg file, Can you please guide me regarding how to use redis db in order to update user information to DB on registration of each user. Thanks & Regards Hemanth Message: 8 Date: Wed, 4 Jan 2017 14:41:42 +0530 (IST) From: "hemanth(a)advaitamtech.com" <hemanth(a)advaitamtech.com> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Regarding usage of Redis Module in Kamailio core Message-ID: <1483521102.18025940(a)apps.rackspace.com> Content-Type: text/plain;charset=UTF-8 Hello, I am using Kamailio as a SIP proxy, I want to store Registered user information in Redis DB, I am able to configure REDIS DB, But i am not able to run the redis command from Kamailio core using the function "redis_cmd("srvN","SET key 1", "r")" I am getting undefined reference error as below "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" can anyone please help me to fix this issue Thanks & Regards Hemanth ------------------------------ Message: 9 Date: Wed, 4 Jan 2017 10:50:05 +0100 From: Vicente Hernando <vhernando(a)systemonenoc.com> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Regarding usage of Redis Module in Kamailio core Message-ID: <b003d2aa-ddd0-574d-b9e3-9b91e44976f1(a)systemonenoc.com> Content-Type: text/plain; charset=utf-8; format=flowed Hello Hemanth, just a quick response, I would check if you compiled ndb_redis module correctly (also if you have hiredis library dependency). Also I would check if you loaded ndb_redis module in your configuration file. Regards, Vicente. On 01/04/2017 10:11 AM, hemanth(a)advaitamtech.com wrote: > Hello, > > I am using Kamailio as a SIP proxy, > > I want to store Registered user information in Redis DB, > > I am able to configure REDIS DB, But i am not able to run the redis command from > > Kamailio core using the function > > "redis_cmd("srvN","SET key 1", "r")" > > I am getting undefined reference error as below > > "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" > > can anyone please help me to fix this issue > > Thanks & Regards > Hemanth > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ End of sr-users Digest, Vol 140, Issue 3 ****************************************

1 0

Regarding usage of Redis Module in Kamailio core
by hemanth＠advaitamtech.com 04 Jan '17

04 Jan '17

Hello, I am using Kamailio as a SIP proxy, I want to store Registered user information in Redis DB, I am able to configure REDIS DB, But i am not able to run the redis command from Kamailio core using the function "redis_cmd("srvN","SET key 1", "r")" I am getting undefined reference error as below "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" can anyone please help me to fix this issue Thanks & Regards Hemanth

2 1

Update: removing MI code
by Daniel-Constantin Mierla 04 Jan '17

04 Jan '17

Hello, based on lib linking in Makefiles, the following modules still use MI, categorized by their state from my point of view. Hopefully some other people will help with those listed in 2) to 6). 1) to be removed - mi_datagram - mi_fifo - mi_rpc - mi_xmlrpc - pua_mi 2) rpc commands implemented - dialog - others modules depend on some mi callback from it (qos and sst) - qos - this looks ready to wipe out the mi code, but didn't get the time not analyze properly as I don't use it usually 3) some rpc commands implemented and the rest should not be very complex - carrierroute - I implemented the dump rpc command, but it was reported that it has issues, so it needs follow up. 4) rpc commands have to be implemented, expecting not to be very complex - cplc - imc 5) expecting some degree of complexity, but they are important modules - rtpengine - rtpproxy 6) not familiar with the mi commands in these modules, so not able to assert what to expect - ims_dialog - some rpc commands are implemented, not sure if the rest of MI are used/usefull - mohqueue - it doesn't seem complex to implement rpc commands at first sight, but the indentation style didn't allowed a fast analyze on a quick look - p_usrloc - several mi commands - sst - uses some callback for MI from dialog module. qos has something similar with rpc alternative already implemented - userblacklist - several mi commands - utils - there are few mi commands related to some forwarding rules: https://www.kamailio.org/docs/modules/devel/modules/utils.html#idp21741924 The 1) to 5) should be done in a way or another, before of after freeze of 5.0.0. But 6) would require the devs of the modules (or the MI parts) to help if they want those commands via RPC. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Kamailio as loadbalancer + NAT pinning
by Vladyslav Zakhozhai 03 Jan '17

03 Jan '17

Hi, I want to use Kamailio as SIP loadbalancer and NAT helper in front of freeswitch boxes. Main tasks here are: 1. load balancing SIP requests to freeswitch boxes 2. fixing contact information during registrations and invites for nat'ed clients (kamailio is not registrar) 3. fixing sdp (or proxying rtp via rtpproxy) for nat'ed clients 4. security (check SIP messages, pike, etc) 5. outbound proxy for freeswitch boxes The main question here is about nat pinning. Acctording to module doc ( http://kamailio.org/docs/modules/4.4.x/modules/nathelper.html) I need nathelper module and usrloc module. So I can proxy REGISTER requests to freeswitch boxes and in onreply_route check for REGISTER statuses: 1. if no success we use pike to limit bruteforce attacks 2. if success I need to save contact to usrloc db table But there is one issue. Multiple registration are allowed on freeswitch boxes and in replies (freeswitch -> kamailio) I see more than one Contact header. Is it an issue or kamailio will handle it without problems? Or I need to save Contact header from request to hash table and retreive it in onreply_route? But I do not know how to save it in usrloc db table then. I thin this is more sophisticated approach. I appreciate your advice. Thank you in advance. -- С уважением, Владислав Захожай

2 4

Help Asterisk with Kamailio unable to register with remote VOIP providers
by Manoj Gupta 03 Jan '17

03 Jan '17

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1: Issue is REGISTER messages sent by asterisk are not responded by kamailio asterisk is constantly doing a re-transmit on REGISTER messages. Retransmitting #1 (NAT) to 10x.3x.6x.236:5060: REGISTER sip:ims.airtel.in SIP/2.0 Via: SIP/2.0/UDP 10x.3x.6x.236:5080;branch=z9hG4bK40ac1ed8;rport Max-Forwards: 70 From: <sip:+91224400@ims.airtel.in>;tag=as565ef860 To: <sip:+91224400@ims.airtel.in> Call-ID: 7dd361ec25cb07344ea9df526e89ca8a(a)10x.3x.6x.236 CSeq: 182 REGISTER User-Agent: Asterisk PBX 11.13.0 Expires: 120 Contact: <sip:+91224400@10x.3x.6x.236:5080> Content-Length: 0 Issue 2: We are not able to see any log messages in /var/log/kamailio.log. We have done all kind of changes suggested on online blogs Why does kamailio not have a pre-set configuration as it is mostly used as sip proxy? Manoj K. Gupta

2 11

Rewrite From domain
by Gonzalo Gasca Meza 03 Jan '17

03 Jan '17

Hi all, I'm using Kamailio to forward calls between 2 Service Providers and I need to rewrite the From header "domain" URI. Example: From: "+18888888888" <sip:+18888888888@sip.sp1.com> to From: "+18888888888" <sip:+18888888888@*sip.sp2.com <http://sip.sp2.com>*> *Call flow:* Phone A --- > SP1 ---> sip ----> (kamailio) SP2 --(LOCATION)-> Phone B When Phone A calls SP2 PhoneB, it contains original sip domain from sp1. ( sip.sp1.com) hence user in SP2 can see call comes from SP1. I would like to rewrite the From domain field in this conditions: a) Calls comes from "sip.sp1.com" AND b) Call is being routed to PhoneB. Right now Im using the following code to find user and send call to B. #!ifdef WITH_ALIASDB # search in DB-based aliases xlog("L_INFO","alias_db_lookup: Call received. $rU\n"); if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif I found this in documentation: $fd - From URI domain if($hdr(From)=~"sip.sp1\.com") { ...} But not sure where is the best place to overwrite the From URI domain header. Thanks

3 7

Help Asterisk with Kamailio unable to register with remote VOIP providers
by Manoj Gupta 02 Jan '17

02 Jan '17

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1: Issue is REGISTER messages sent by asterisk are not responded by kamailio asterisk is constantly doing a re-transmit on REGISTER messages. Retransmitting #1 (NAT) to 10x.3x.6x.236:5060: REGISTER sip:ims.airtel.in SIP/2.0 Via: SIP/2.0/UDP 10x.3x.6x.236:5080;branch=z9hG4bK40ac1ed8;rport Max-Forwards: 70 From: <sip:+91224400@ims.airtel.in>;tag=as565ef860 To: <sip:+91224400@ims.airtel.in> Call-ID: 7dd361ec25cb07344ea9df526e89ca8a(a)10x.3x.6x.236 CSeq: 182 REGISTER User-Agent: Asterisk PBX 11.13.0 Expires: 120 Contact: <sip:+91224400@10x.3x.6x.236:5080> Content-Length: 0 Issue 2: We are not able to see any log messages in /var/log/kamailio.log. We have done all kind of changes suggested on online blogs Why does kamailio not have a pre-set configuration as it is mostly used as sip proxy? Manoj K. Gupta

1 0

Freezing for release v5.0.0
by Daniel-Constantin Mierla 02 Jan '17

02 Jan '17

Hello, I propose to freeze the development of v5.0.0 on January 4, 2017. Hopefully it is just in time for getting the frozen version in next Debian stable. Meanwhile, I hope we can get rid of mi code to have a proper fresh start with cleaned up code in 5.0.0. The big code tree restructuring was already done, config files in embedded languages is there as well (it needs more exports, but that's only about available time to fill some structures, not new feature development). If anyone wants to make other proposals, reply to mailing lists and let's discuss. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Happy 2017!
by Daniel-Constantin Mierla 01 Jan '17

01 Jan '17

A fresh year ahead, let it be a great one for all Kamailio friends! Wishing a prosperous and safe 2017 everywhere! Happy New Year! -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

TOPOS contact issue on RE-INVITE
by Daren FERREIRA 30 Dec '16

30 Dec '16

Hello, I made lots of tests in order to make Kamailio act as a topology hiding proxy, but i always got problems when using topos to store needed Route and Record-Route informations. I thought that was because of possible error(s) playing with : - socket binding - destination tweaking that may have lead to transaction breaking, so i decided to use NAT mode (even if that only can be temporary). My problem is the same. I have not found any way to influence TOPOS on the domain it use on the « Contact » header. Everything is getting well, iPs on Contact Header are toggled between external and internal sides, until RE-INVITE. From there, Contact domain are no more toggled, and so: - Public IP appear on Private side : i can manage my divide to force it to reply to the sender of the request instead of the contact - Private IP appear on Public side : i can’t do anything to make the external side ignore contact’s domain I wonder if that is a bug, related to RE-INVITE, or if there is any variable i can change to say topos which domain to be advertised on Contact headers? PUBLIC NETWORK SIDE RFC6598 NETWORK SIDE ————————— | | PROVIDER Y.Y.Y.Y———————X.X.X.X—| KAMAILIO ]—100.64.10.1———————— 100.64.10.2 ASTERISK (or anything else) | | ————————— 20:58:46.215911 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 1004 INVITE sip:+0598@100.64.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 246 v=0 o=root 79038203 79038203 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=audio 13564 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv 20:58:46.216642 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 375 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport=5060;received=100.64.10.2 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 0 20:58:46.222391 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 1040 INVITE sip:+0598@Y.Y.Y.Y:5060 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 Max-Forwards: 69 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 264 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=root 79038203 79038203 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=audio 57810 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=nortpproxy:yes 20:58:46.234592 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 290 SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE 20:58:46.548771 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 718 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:46.554903 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 719 SIP/2.0 183 Session Progress From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:47.735919 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:47.741229 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:48.050762 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:48.056175 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.811066 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 774 SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:52.818381 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 775 SIP/2.0 200 OK From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.819315 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 446 ACK sip:atpsh-58656af1-769f-2@100.64.10.1 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK18c31028;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 20:58:52.825894 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 491 ACK sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.d149220aa711a5230da51bfea457d8bc.0 Max-Forwards: 69 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> 20:59:00.388481 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 910 INVITE sip:btpsh-58656af1-769f-1@X.X.X.X <sip:btpsh-58656af1-769f-1@X.X.X.X> SIP/2.0 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 273 Max-Forwards: 69 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103905 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=image 17818 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.390679 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 420 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1;rport=5060 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 0 20:59:00.395723 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 890 INVITE sip:+9997@100.64.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 289 Max-Forwards: 68 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=- 1510281183 168103905 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=image 62130 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.397160 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 580 SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Length: 0 20:59:00.397344 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.405004 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 SIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.896864 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.904049 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 TSIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes And so on until timeout and BYE

3 4

Astersik kamailio integration.
by Deep Pratap Singh Tomar 29 Dec '16

29 Dec '16

Hi I have modified the file so that if i send register request to kamailio it sud register on asterisk also. i have shared you the code but i cant see anyting happening in the asterisk server. please help me ASAP. L&T Technology Services Ltd www.LntTechservices.com<http://www.lnttechservices.com/> This Email may contain confidential or privileged information for the intended recipient (s). If you are not the intended recipient, please do not use or disseminate the information, notify the sender and delete it from your system.

2 1

Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 29 Dec '16

29 Dec '16

I implemented full NAT logic as per the sample config. Still unable to resolve the issue. How do I let Kamailio change record_route header to use public ip address? Please help!!! (attached are latest scripts) From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 12:39 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Subject: Kamailio behind NAT, ACK to private IP not advertised public IP. Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

3 4

String split function
by Grant Bagdasarian 29 Dec '16

29 Dec '16

Hi, Is there a string split function built into Kamailio for use in the cfg script? I'd like to split a string of key/value pairs inside the cfg script. For example: Key1=Value1;Key2=Value2; First split on ; Then on = I think I can use the String Transformation {s.select,index,separator}, but I was hoping there is a built in function. {param.valueat,index[, delimiter]} may also be an option. Regards, Grant

2 2

Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 29 Dec '16

29 Dec '16

I am attaching latest config along with kamailio log file and client trace. Please let me know how I can get Kamailio to advertise public ip in record route. I followed all the steps mentioned here - http://www.fredposner.com/1457/kamailio-behind-nat/ Thanks Pranathi From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 5:09 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Cc: Mick McGrellis <MMcGrellis(a)cyracom.com>; 'miconda(a)gmail.com' <miconda(a)gmail.com> Subject: RE: Kamailio behind NAT, ACK to private IP not advertised public IP. I implemented full NAT logic as per the sample config. Still unable to resolve the issue. How do I let Kamailio change record_route header to use public ip address? Please help!!! (attached are latest scripts) From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 12:39 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Subject: Kamailio behind NAT, ACK to private IP not advertised public IP. Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

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Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 28 Dec '16

28 Dec '16

Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

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BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Nihar Ranjan Deb 28 Dec '16

28 Dec '16

Hi All, I am a new user of kamailio. We have setup a kamilio server with RTP proxy. Most of the features are working fine except BYE. Say User A and B are registered in kamailio server. A (caller) connected to B(called), call established. 1. If Caller (A) disconnects the call, everything goes fine. B gets disconnected and call releases. 2. But if Called party (B) disconnects the call, A never gets disconnected. I checked trace as well, in this case caller (A) gets a BYE with B's callid. So A rejects call with "481 call leg transaction doesn't exists". Basically A party and B party has separate dialog established with Kamilio. So they have two separate callids . All messages are exchanged correctly. Only when B party disconnects call, kamailio sends BYE to A party with B party callid. Please help me to resolve this issue. Do let me know if any more information required. ------ INVITE-------------------------- INVITE sip:966268@*** SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport Max-Forwards: 70 Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> To: <sip:966268@***> From: "334031"<sip:334031@****>;tag=4406552c Call-ID: 82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc CSeq: 2 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="334031",realm="*****",nonce="****",uri="sip:966268@***",response=" ***",algorithm=MD5 Supported: replaces User-Agent: X-Lite release 4.9.6 stamp 82158 Content-Length: 334 v=0 o=- 13127393523589215 1 IN IP4 192.168.1.36 s=X-Lite release 4.9.6 stamp 82158 c=IN IP4 192.168.1.36 t=0 0 m=audio 57116 RTP/AVP 9 8 120 0 84 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -------------------------BYE-------------------------------- BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* From: <sip:966268@***>;tag=1665200125 To: "334031" <sip:334031@***>;tag=4406552c Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> Max-Forwards: 69 User-Agent: Yealink SIP-T20P 9.72.0.80 Content-Length: 0 -------------------------- 481--------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.1 64.228.12 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* To: "334031" <sip:334031@***>;tag=4406552c From: <sip:966268@***>;tag=1665200125 Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Accept-Language: en Content-Length: 0 Regards, Nihar

1 0

rtpengine destroys voice branch on multiple branch call
by Yuriy Gorlichenko 27 Dec '16

27 Dec '16

Hi. I have 2 endpoints 1001 and 1002 1001 registered from 1 device 1002 registered from 2 devices at the same time When I called to 1002 kamailio makes 2 branches rtpengine_manage command called from branch rout for handling every branch directly (it can be different endoint types (ws/tls/udp) for each branch) When i picking up at the 1002 on one device server sends CANCEL (answered elswhere) to another devices of 1002 rtpengine_manage reacts at the CANCEL and deletes this brach after 30 seconds My question is: What is solution for use rtpengine_manage and call branches together?

1 0

set_body_multipart and append_body_part
by Diego Nadares 26 Dec '16

26 Dec '16

Hi guys, I'm trying to append a content type to a message that has only one content type. I tried using just set_body_multipart and now I'm using set_body multipart and append_body_part with no success too. The result is a message with multipart but has "two" sdp and no gtd. It has the old sdp with no ip changed for rtpproxy and the new sdp with ip changed BUT with no content type. And the message has not the end boundary (--unique-boundary-1.) Kamailio log has an info and a warning Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: INFO: <core> [msg_translator.c:1693]: get_boundary(): Content-Type hdr has no params <application/sdp> Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: WARNING: <core> [msg_translator.c:1959]: build_req_buf_from_sip_req(): check_boundaries error kamailio.cfg ...... route[RELAY]{ $avp(ch) = 1 .... if ($avp(ch)){ set_body_multipart(); msg_apply_changes(); $var(gtd) ="IAM,\r\nGCI,asdafsdfasd\r\n\r\n"; append_body_part("$var(gtd)", "application/gtd", "signal;handling=optional"); } .... } Ngrep U 2016/12/26 11:36:22.496634 172.16.213.21:5060 -> 172.16.213.38:5060 ... Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. U 2016/12/26 11:36:22.688458 172.16.213.38:5060 -> 172.16.208.11:5060 .... Content-Type: multipart/mixed;boundary="unique-boundary-1". Mime-Version: 1.0. Remote-Party-ID: <sip:1152565064@172.16.213.38:5060 >;party=calling;privacy=off;screen=no. . --unique-boundary-1. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. . --unique-boundary-1. v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 58724 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=sendrecv. a=rtcp:58725. a=ice-ufrag:hhXDwrco. a=ice-pwd:5temIXP8veOGuD8Dsllm5HXkWU. Any help will be very very appreciated! Thanks in advance. Diego

2 4

help configure dispatcher in Amazon ec2
by huan nguyen duy 26 Dec '16

26 Dec '16

Hi, I’m very appreciated if someone can figure out this problem for me. My aim is to setup HA Kamailio using Dispatcher module in Amazon EC2, system like this: soft phone 1(Linphone) —————> | Load balancer —————> | Sip Server huancomputer | private IP: 172.31.11.35 | 172.31.13.113 soft phone 2(Linphone) —————> | public IP: 35.166.179.244 | 35.166.152.104 huannguyen | | The dispatcher.lst config file in /usr/local/etc/kamailio as below: # gatewayss 1 sip:172.31.13.113:5060;transport=udp <sip:172.31.13.113:5060;transport=udp> 0 0 abc=xyz User records subscriber table: ——————————————————————————————————————————————————— id | username | domain ——————————————————————————————————————————————————— 70 | nguyenhuan | nguyenhuan(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> 71 | huancomputer | huancomputer(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> Location records from location table: ——————————————————————————————————————————————————— username | domain | contact ——————————————————————————————————————————————————— huancomputer | NULL | sip: huancomputer(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:64166;transport=udp nguyenhuan | NULL | sip: nguyenhuan(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:33267;transport=udp But soft phone 1 cannot make a call to soft phone 2, below is data is extracted from wireshark: Load balancer: -------------------------------- No. Time Source Destination Protocol Length Info 177 4.463129 14.161.41.70 172.31.11.35 SIP/SDP 1433 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 179 4.463367 172.31.11.35 14.161.41.70 SIP 452 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 180 4.463421 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1215 32.552942 14.161.41.70 172.31.11.35 SIP 1105 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1216 32.553170 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1219 32.555588 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1220 32.555706 172.31.11.35 14.161.41.70 SIP 593 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1609 42.603150 14.161.41.70 172.31.11.35 SIP 1113 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1610 42.603364 172.31.11.35 172.31.13.113 SIP 1269 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1613 42.605800 172.31.13.113 172.31.11.35 SIP 749 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1614 42.605932 172.31.11.35 14.161.41.70 SIP 599 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1688 44.425237 172.31.11.35 14.161.41.70 SIP 471 Status: 408 Request Timeout | No. Time Source Destination Protocol Length Info 1689 44.425285 172.31.11.35 172.31.13.113 SIP 491 Request: CANCEL sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1691 44.426018 172.31.13.113 172.31.11.35 SIP 519 Status: 200 canceling | Data from Sip Server: ------------------------------- No. Time Source Destination Protocol Length Info 237 6.242928 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 243 6.244409 172.31.13.113 172.31.11.35 SIP 602 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 244 6.244541 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 288 6.710414 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 328 7.710454 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 396 9.710411 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 547 13.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 688 17.710406 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 835 21.710381 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 985 25.710385 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1130 29.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1279 33.710400 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1303 34.332395 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1312 34.334415 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1459 37.710402 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1609 41.710395 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | Huan Nguyen, --------------------------------------- Mobile app developer Phan Thi Company Sai Gon, VietNam Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

1 1

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 24 Dec '16

24 Dec '16

While waiting for Santa, it's a good time to also express my thanks and greetings to all the friends, developers and community members that made 2016 another amazing year for Kamailio project. Enjoy the winter holidays together with your beloved ones! Merry Christmas! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Merry Christmas to all Kamailians
by Olle E. Johansson 24 Dec '16

24 Dec '16

Friends, Trying to calm down and prepare for Christmas with the family. It’s been another great year for Kamailio and I’m proud to be part of the Kamailio development community. We’ve made great releases, had a great conference and overall done good stuff :-) For all of you that celebrate Christmas - a Merry Christmas! To the rest of you: Happy Holidays! A big Thank You to all developers and a special giga-Thank You to Daniel for all the time you spend working with the code, helping out in discussions on the lists and in the bug tracker. And to all of you in the community - Thank You for being part of this! Greetings from a cold Sweden, currently without snow in my area just north of Stockholm. We celebrate on Christmas Eve (tomorrow) so there’s a lot of food being prepared right now and the last minute shopping is just about to start... Keep your Kamailio running! /Olle PS. If you are not currently following the project on Twitter - it’s time! Find us as @kamailioproject

3 2

baresip version 0.5.0 released
by Alfred E. Heggestad 23 Dec '16

23 Dec '16

Hi, my apologies for the off-topic posting but I think this could be interesting for the SIP Community. Baresip version 0.5.0 is now released, you can find the announcement email here: http://lists.creytiv.com/pipermail/re-devel/2016-December/001154.html Merry Xmas from Norway! /alfred

1 0

Re: [SR-Users] rtjson + mhomed and record route (corrected)
by Diego Nadares 23 Dec '16

23 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The call is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header? ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # 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(); } .... Thanks in advance. Diego 2016-12-22 12:08 GMT-03:00 Diego Nadares <dnadares(a)gmail.com>: > Hi list, > > We have two eth, one with a private ip and the other with a public ip. We > have MHOMED configured. The call comes from private network and then is > routed to a public network via rtjson routing. The is routed but the thing > is that the record route is being set with the private one (It's the same > when calls come from public and is routed to a private network, record > route has the public and not the private). > > With mhomed configured shouldn't Record Route be updated as VIA header > > ".... When activated, sip-router will select a socket that can reach the > destination (to be able to connect to the remote address). (sip-router > opens a UDP socket to the destination, then it retrieves the local IP which > was assigned by the operating system to the new UDP socket. Then this > socket will be closed and the retrieved IP address will be used as IP > address in the Via/Record-Route headers)..." > > This is part of ngrep > > U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 > INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. > To: <sip:1111111111@172.16.213.38:5060>. > Call-ID: 1-13024(a)172.16.213.21. > CSeq: 1 INVITE. > Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. > .... > > U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 > SIP/2.0 100 trying -- your call is important to us. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > .... > > U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 > INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. > *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* > Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f > Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 > > This is my cfg > > mhomed=1 > > request_route { > > > # per request initial checks > route(REQINIT); > > # NAT detection > #route(NATDETECT); > > # CANCEL processing > if (is_method("CANCEL")) { > if (t_check_trans()) { > route(RELAY); > } > exit; > } > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # handle retransmissions > if(t_precheck_trans()) { > t_check_trans(); > exit; > } > t_check_trans(); > > # authentication > route(AUTH); > > # 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(); > } > > > Thanks in advance. > > Diego. > >

1 1

kamailio dispatcher call flow example
by huan nguyen duy 23 Dec '16

23 Dec '16

Hi, I want to load balance multiple kamailio server using dispatcher module but not success at configuration. I’m not clearly how it work. Someone can show me an example call flow like below i’m very appreciated. Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| | | | 407 F2 | | | |<---------------| | | | ACK F3 | | | |--------------->| | | | INVITE F4 | | | |--------------->| INVITE F5 | | | 100 F6 |--------------->| INVITE F7 | |<---------------| 100 F8 |--------------->| | |<---------------| | | | | 180 F9 | | | 180 F10 |<---------------| | 180 F11 |<---------------| | |<---------------| | 200 F12 | | | 200 F13 |<---------------| | 200 F14 |<---------------| | |<---------------| | | | ACK F15 | | | |--------------->| ACK F16 | | | |--------------->| ACK F17 | | | |--------------->| | Both Way RTP Media | |<================================================>| | | | BYE F18 | | | BYE F19 |<---------------| | BYE F20 |<---------------| | |<---------------| | | | 200 F21 | | | |--------------->| 200 F22 | | | |--------------->| 200 F23 | | | |--------------->| | | | | Huan Nguyen, ——————————————————— Mobile app developer Phan Thi Company Sai Gon, VietNam

2 1

rtpproxy/rtpengine. Help to understand.
by Igor Olhovskiy 22 Dec '16

22 Dec '16

Hi! Issue I can’t figure out. Or all working ok and that’s just me who not understands. I have situation softPhone (111.222.3.2) -> Kamailio w. rtpproxy (111.222.3.3) -> PBX (111.222.3.4) All addresses are on same public network. rtpproxy is running with -l 111.222.3.3 -A 111.222.3.3 I want all media goes through rtpproxy. So, in INVITE and ACK I call rtpproxy_manage(). But according to sip packets on PBX (10.0.0.4) I got media address 10.0.0.2 in SDP, but not 10.0.0.3 If I use scheme, where client in some other NATted network (like 192.168.1.100), all is replaced correctly (means rtpproxy_manage() is working) So, I can’t understand, rtpproxy_manage() changes only rfc1918 addresses or I’m missing something? If any - can provide pcap’s with both cases. Regards, Igor

3 5

rtjson + mhomed and record route
by Diego Nadares 22 Dec '16

22 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # 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(); } Thanks in advance. Diego.

1 0

Socket in rtjson module
by Diego Nadares 22 Dec '16

22 Dec '16

Hi Guys, Anybody knows if each 'socket' specified in routes struct should be use to send the messages? The socket is configured in kamailio. In kamailio.cfg listen=udp:172.16.213.38:5060 json struct "routes": [ { "uri": "sip:127.0.0.1:5080", "dst_uri": "sip:127.0.0.1:5082", "path": "<sip:127.0.0.1:5084>, <sip:127.0.0.1:5086>", ----> "socket": "udp:172.16.213.38:5060",<---- "headers": { "from": { "display": "Alice", "uri": "sip:alice@127.. In our kamailio has two eth and I want to select which one to use but I can't make it work through rtjson. In the code of the module I can't find where is being assigned the socket. Thanks in advance! Diego

2 3

kamailio dispatcher can not forward call in Amazon EC2
by huan nguyen duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information. Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

1 0

kamailio dispatcher can not forward call in Amazon EC2
by Huân Nguyễn Duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information.

1 0

Memory leak in Kamailio 4.4.4 TLS Module
by raja raja 22 Dec '16

22 Dec '16

Hi, shared memory not freed once TLS connection close means shm available memory decreasing for new connection that leads to the following error *tls: ssl bug #1491 workaround: not enough memory for safe operation* *TestCase:* 1. start kamailio 2. use sample program to establish tls connection and close imediately. 3. Kamailio Log is For first TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63604096 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63545192 For second TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63545192 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63489352 *Memory Leak:* for tls connection connection 63604096 - 63545192 = 58904 bytes not freed. for tls connection connection 63545192 - 63489352 = 55840 bytes not freed. I tested same scenario on kamailio - 4.2.4, 4.3.4 versions result is same. Could you please provide the fix? Thanks & Regards, Raja Chegondi.

2 1

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