- sr-users - kamailio.org

RFC: Kamailio tutorials/cookbooks/docs - wiki or markdown
by Daniel-Constantin Mierla 31 Jan '17

31 Jan '17

Hello, I want to start a discussion about some of the kamailio tutorials from the wiki, mainly those related to core cookbook, variables and transformations. They are now in dokuwiki at: - https://www.kamailio.org/wiki/#cookbooks The wiki served us pretty well so far, however the contributions there are not that active. From time to time we still get spammers, even we added restriction that only registered and authenticated users can write. I am thinking that maybe people don't like creating yet another account just add some example or rephrase for clarity. On the other hand, there are over 400 registered users, many of them I expect to be either made because if misunderstanding the purpose or by spammers that guessed the (sip) captcha (plenty of random usernames). Besides that I think the wiki adds overhead when releasing a new version as each time we have to copy and paste old version content to a new set of pages and update the version number. With the above in mind, I wonder if won't be better to store those tutorials in markdown format and use mkdocs.org or gitbook (https://github.com/GitbookIO/gitbook) tools to generate the html and host it in kamailio.org. The wiki will still be used to index them. Among the benefits I see: * store in github.com (a new repo, like: kamailio-docs) and all devs can contribute directly, other users can make pull requests * create branches for each major version of kamailio and backport across them with git whenever is applicable * the edit can be done directly via github.com website (similar to the wiki right now) and even read it directly from github in html transformed from markdown by github * one can get them offline by cloning the git repo, use them or enhance when offline Many if the tutorials will stay in the wiki, but I think those that are related to documenting per kamailio version should be migrated to gihub storage + markdown. If there is a strong pro opinion for this change, I can try to convert one of the cookbooks very soon so people can feel better the difference. If we get to a consensus, then I think we can have those tutorials in the new format for v5.0. Should anyone have comments, other suggestions or improvements, let's put them on debate. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

2 7

serial forking calls using TSILO
by Tomas Zanet 30 Jan '17

30 Jan '17

Hello, I'm using Kamailio 4.4.4 with TSILO module in order to support Push Notifications used by our voip app on Apple ios10. So far, everything works fine: Kamailio can get an incoming call, suspend it, then send push notification with an external script, receive a new app registration and then call it after resuming the invite. To do so, thanks to Mr, Cabiddu, I used all the functions described here http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-… Right now, I would like to support serial forking calls using push notification because it was already supported with legacy voip. To do so, I used failure_route function where, after getting the call destinations and setting them into an avp, I set the new SIP request uri $ru. Here is my failure_route: failure_route[MANAGE_FAILURE] route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) t_reply("404","Not found"); exit; #!endif if (is_avp_set("$avp(group_members_db)")) $ru ="sip:"+$avp(group_members_db)+"@"+$fd; $avp(group_members_db) = $null; route(LOCATION); exit; I am troubleshooting this scenario and this is what I see: 1) Kamailio receives incoming call 2) Suspend it 3) Send Push notification to account1 4) Kamailio receives account1 registration -> INVRESUME route is now called 5) Then call account1 6) Nobody answers the call 7) failure_route[MANAGE_FAILURE] is now called -> set $ru with next sipaccount: account2 8) suspend invite 9) send push notification to account2 10) account2 sends its sip registration -> INVRESUME route is now called 11) Kamailio calls account1 instead of account2 -> this is my issue On step 10: despite account2 is registered, I checked it using "kamctl ul show", the second call is forwarded all'account1 and not to account2. I also checked the TSILO logs and everything seems ok. Here is the logs: First call: suspended transaction [53945:1648394094] asterisk => account1 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account1 53945:1648394094) second call: suspended transaction [53945:1648394094] asterisk => account2 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account2 53945:1648394094) In order to find out the issue I put some xlogs, printing $ru value: what I see is the $ru value is set correctly on failure_route but as soon as the t_continue is called, the ru overwritten back to account1 Can anyone address me to find out the solution? Thanks in advance T. Tomas Zanet Software Design Department tzanet(a)came.com

3 7

Re: [SR-Users] FOSDEM 2017
by DanB 30 Jan '17

30 Jan '17

Hey Daniel, if anything to be reserved, please count us also in. Will come back regarding number of seats but just to be aware that we are interested to join the dinner. Thanks! DanB

8 7

mhomed opened sockets
by Diego Nadares 30 Jan '17

30 Jan '17

Hi Guys, We have two network interfaces. One public and one private. We had enabled mhomed and works great BUT if we do a netstat -naup | grep 5060 we see that some sockets keeps opened also when the call is finished. The source port in netstat it's not 5060 so I suppose that kamailio use another one for something else. Is this the right behavior? How long does this sockets remain opened? Are we missing something in our cfg? 6 2017-01-25 15:52:04.986651 172.16.213.38 172.16.200.159 SIP/SDP 146 Request: INVITE sip:xxx@172.16.200.159:5060 34 2017-01-25 15:52:19.636895 172.16.213.38 172.16.200.159 SIP 716 Request: ACK sip:xxx@172.16.200.159:5060 | 37 2017-01-25 15:54:38.947831 172.16.208.111 172.16.213.38 SIP 736 Request: BYE sip:127.0.0.8:5060;line=sr-987jhlk* | 40 2017-01-25 15:54:38.950899 172.16.213.38 172.16.200.159 SIP 660 Request: BYE sip:172.16.208.111:5060 | [root@dwrfsd01 kamailio]# netstat -naup | grep 5060 udp 0 0 172.16.213.38:*56086* <http://172.16.213.38:56086> 172.16.200.159:5060 ESTABLISHED 17015/kamailio udp 0 0 172.16.213.38:5060 0.0.0.0:* 17006/kamailio udp 0 0 172.16.213.38:*41971* <http://172.16.213.38:41971> 172.16.200.159:5060 ESTABLISHED 17051/kamailio udp 0 0 172.16.213.38:*54651* <http://172.16.213.38:54651> 172.16.200.159:5060 ESTABLISHED 17008/kamailio Thanks in advance! Diego.

2 3

Presence Module do not load db_url
by Vu. Minh Cao 29 Jan '17

29 Jan '17

Hi, I got error when loading presense module in Kamailio 4.4: [Configure] loadmodule "presence.so" modparam("presence", "db_url", "mysql://kamailio:kamailiorw@127.0.0.1/kamailio") [Error] ERROR: presence [presence.c:608]: fixup_presence(): Bad config - you can not call 'handle_publish' function (db_url not set) Which is really strange since according to the source code this error just occurs if “db_url” is not set. I already checked my database also, I can access mysql with about username/password. Yours sincerely, — Vu

1 0

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

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"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

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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

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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

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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.

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