- sr-users - kamailio.org

load-balance to registered trunks?
by David Villasmil 03 Sep '18

03 Sep '18

Hello guys, I've got this scenario where some gateways will register on my kamailio and i need to send them calls on a load-balance fashion. is this possible? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

4 6

Dispatcher question
by Igor Olhovskiy 03 Sep '18

03 Sep '18

Hi! I’m having some issues with dispatcher and cannot get why it’s not working Idea is to use ds_next_dst right after ds_select_dst I’m having it like this modparam("dispatcher", "list_file", "/tmp/dispatcher.list") modparam("dispatcher", "flags", 2) # Enable dispather failover modparam("dispatcher", "xavp_dst", "_dispacher_dst_list_") … ds_select_dst(«1», «0»); … # Some extra check on $du here … ds_next_dst(); But get in logs WARNING: dispatcher [dispatch.c:2285]: ds_update_dst(): no xavp uri field in next destination record But contents of /tmp/dispatcher.list is 1 sip:172.28.0.12:5060 1 sip:172.28.0.19:5060 Can I somehow dump xavp_dst var to check if ds_select_dst is correct? kamailio 5.2.0-dev6 Regards, Igor

5 17

handle_ruri_alias() question or issue?
by Wilkins, Steve 02 Sep '18

02 Sep '18

Good Morning All, The following Incoming request came in => SIP Incoming Request: [[BYE sip:2406506175@20.20.20.20:5060;alias=10.10.10.5~55157~2 SIP/2.0 I called handle_ruri_alias(), and expected the destination and port to be set to 10.10.10.5:55157, but it was not. It was left 20.20.20.20:5060. Is my thinking correct or is there another way to set the destination and port to the alias? Thanks All!, -Steve

3 9

Having trouble associating response to message request
by Byran Carlock 31 Aug '18

31 Aug '18

Hello, I'll be the first to admit I'm a complete novice to Kamailio, but I'm learning due to necessity. I'm working on setting up Kamailio as a sip proxy (SBC?). Any help would be massively appreciated. The desired setup would be for Kamailio to as an external connection point to authenticate the connection and reverse proxy/NAT the rest of the traffic to the asterisk server, hiding internal topology as we go. Right now I'm using dispatcher with NAT and ALIAS to push connections to proxy@{PROXY IP} to proxy@{ASTERISK EXTERNAL IP}. I'm running into an issue where a "Message" is being sent, forwarded to the Asterisk server and responded to. Then Asterisk spawns a new call ID "To" the proxy rather than the original requestor which of course gets a 484 response. How can I somehow associate these two messages and complete the route? MESSAGE sip:proxy@{ASTERISK<sip:proxy@%7bASTERISK> EXTERNAL IP} SIP/2.0 Via: SIP/2.0/UDP {PROXY IP}:5060;branch=z9hG4bK0542.4d3d95c66b82f5825d9b90ba7ae88191.0 Via: SIP/2.0/UDP {CLIENT IP}:37916;received={CLIENT EXTERNAL IP};rport=37916;branch=z9hG4bKPj9b82293b-8ac4-4314-9cb8-d76716746e Max-Forwards: 69 From: <sip:{CLIENT IP}<sip:%7bCLIENT%20IP%7d>>;tag=8aabdd54-6312-4ba4-bbe4-e713d2cf55a7 To: <sip:proxy@{PROXY IP}<sip:proxy@%7bPROXY%20IP%7d>> Call-ID: 9d7bc193-55d9-479b-8c34-3663c3a8bf3b CSeq: 50215 MESSAGE Accept: text/plain, application/im-iscomposing+xml User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-AppServer-RequestType: Preconfig X-Digium-AppServer-Preconfig-Id: ca0175e891557c4e9d6f7314cdbf4044 X-Digium-Software-Version: 4.1.1 (2fccfc8f) X-Digium-AppServer-ID: mobileui0000000004 Content-Type: text/plain Content-Length: 182 P-hint: outbound Message Contents MESSAGE sip:{PROXY<sip:%7bPROXY> IP}:5060 SIP/2.0 Via: SIP/2.0/UDP {ASTERISK EXTERNAL IP}:5060;rport;branch=z9hG4bKPj6c1bdda3-2b78-4ae2-ac19-5ec2ca8b8928 From: <sip:proxy@{ASTERISK INTERNAL IP}<sip:proxy@%7bASTERISK%20INTERNAL%20IP%7d>>;tag=79cb0374-5b41-4550-9158-304a18bb1ee6 To: <sip:{PROXY IP}<sip:%7bPROXY%20IP%7d>> Contact: <sip:asterisk@{ASTERISK EXTERNAL IP}:5060<sip:asterisk@%7bASTERISK%20EXTERNAL%20IP%7d:5060>> Call-ID: 94c778a4-d3b3-42c1-ac26-25664d9d37bc CSeq: 4076 MESSAGE X-Digium-AppServer-ResponseType: PreconfigResponse X-Digium-AppServer-Password: ofPnRi4Pt/tGqdSYNn97tiO8ju/maPvAsG7ZeESGyUUWUgvF0xlMuhAUDVzJE866dv0+QeOowFTawa6UQJKQsQ== X-Digium-AppServer-Session: 471800327353850017 X-Digium-AppServer-Message-Salt: 85DA16A6C8F17E46 X-Digium-AppServer-Message-Sequence: 0 X-Digium-AppServer-ID: mobileui0000000004 Max-Forwards: 70 User-Agent: Asterisk PBX 13.1.0-6.7.rl.1516160751.a9f0643 Content-Type: text/plain Content-Length: 600 MESSAGE CONTENT

1 0

Having trouble associating response to message request
by Byran Carlock 31 Aug '18

31 Aug '18

Hello, I'll be the first to admit I'm a complete novice to Kamailio, but I'm learning due to necessity. I'm working on setting up Kamailio as a sip proxy (SBC?). The desired setup would be for Kamailio to as an external connection point to authenticate the connection and reverse proxy/NAT the rest of the traffic to the asterisk server, hiding internal topology as we go. Right now I'm using dispatcher with NAT and ALIAS to push connections to proxy@{PROXY IP} to proxy@{ASTERISK EXTERNAL IP}. I'm running into an issue where a "Message" is being sent, forwarded to the Asterisk server and responded to. Then Asterisk spawns a new call ID "To" the proxy rather than the original requestor which of course gets a 484 response. How can I somehow associate these two messages and complete the route? MESSAGE sip:proxy@{ASTERISK EXTERNAL IP} SIP/2.0 Via: SIP/2.0/UDP {PROXY IP}:5060;branch=z9hG4bK0542.4d3d95c66b82f5825d9b90ba7ae88191.0 Via: SIP/2.0/UDP {CLIENT IP}:37916;received={CLIENT EXTERNAL IP};rport=37916;branch=z9hG4bKPj9b82293b-8ac4-4314-9cb8-d76716746e Max-Forwards: 69 From: <sip:{CLIENT IP}>;tag=8aabdd54-6312-4ba4-bbe4-e713d2cf55a7 To: <sip:proxy@{PROXY IP}> Call-ID: 9d7bc193-55d9-479b-8c34-3663c3a8bf3b CSeq: 50215 MESSAGE Accept: text/plain, application/im-iscomposing+xml User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-AppServer-RequestType: Preconfig X-Digium-AppServer-Preconfig-Id: ca0175e891557c4e9d6f7314cdbf4044 X-Digium-Software-Version: 4.1.1 (2fccfc8f) X-Digium-AppServer-ID: mobileui0000000004 Content-Type: text/plain Content-Length: 182 P-hint: outbound Message Contents MESSAGE sip:{PROXY IP}:5060 SIP/2.0 Via: SIP/2.0/UDP {ASTERISK EXTERNAL IP}:5060;rport;branch=z9hG4bKPj6c1bdda3-2b78-4ae2-ac19-5ec2ca8b8928 From: <sip:proxy@{ASTERISK INTERNAL IP}>;tag=79cb0374-5b41-4550-9158-304a18bb1ee6 To: <sip:{PROXY IP}> Contact: <sip:asterisk@{ASTERISK EXTERNAL IP}:5060> Call-ID: 94c778a4-d3b3-42c1-ac26-25664d9d37bc CSeq: 4076 MESSAGE X-Digium-AppServer-ResponseType: PreconfigResponse X-Digium-AppServer-Password: ofPnRi4Pt/tGqdSYNn97tiO8ju/maPvAsG7ZeESGyUUWUgvF0xlMuhAUDVzJE866dv0+QeOowFTawa6UQJKQsQ== X-Digium-AppServer-Session: 471800327353850017 X-Digium-AppServer-Message-Salt: 85DA16A6C8F17E46 X-Digium-AppServer-Message-Sequence: 0 X-Digium-AppServer-ID: mobileui0000000004 Max-Forwards: 70 User-Agent: Asterisk PBX 13.1.0-6.7.rl.1516160751.a9f0643 Content-Type: text/plain Content-Length: 600 MESSAGE CONTENT

1 0

CentOS_7/kamailio_v5.1.x-rpms/ [Errno -1] Package does not match intended download.
by Denys Pozniak 31 Aug '18

31 Aug '18

Hello! Please check repository for Kamailio v5.1.x for CentOS7: http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… I got error: kamailio-debuginfo-5.1.5-1.el7 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not SatisfiableB --:--:-- ETA Trying other mirror. kamailio-outbound-5.1.5-1.el7. FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.1.x-rpms clean metadata Trying other mirror. kamailio-5.1.5-1.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.1.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. kamailio-debuginfo-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. kamailio-outbound-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. -- BR, Denys Pozniak

2 1

Re: [SR-Users] Final reply code is wrong in case of 302 and 486 busy
by Marco Capetta 31 Aug '18

31 Aug '18

Hi Daniel, did you have some time to have a look at the issue? Thank you very much Regards Marco

2 1

How to detect and Hangup a call on network lost at Caller or Callee?
by Hamid Hashmi 31 Aug '18

31 Aug '18

Dear Users Please advise how to detect a network lost at caller or callee side. I have tried timeout_socket<https://kamailio.org/docs/modules/5.1.x/modules/rtpproxy.html#rtpproxy.p.ti…> and keepalives <https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dialog.p.ka_int…> method but nothing worked for me. RTPproxy did not send timeout notification to Kamailio (Issue #1597<https://github.com/kamailio/kamailio/issues/1597>) while there is no options packet within a dialog. Is there any other method to cater such scenario. Regards Hamid R. Hashmi

5 11

Quick way to enable/disable all uac outbound registrations
by Joel Serrano 31 Aug '18

31 Aug '18

Hi, On an active/passive kamailio cluster, we use keepalived to move a virtual IP between the nodes and all signaling goes to that VIP. One of our DID providers requires us to register with every single DID we have from them in order to receive calls for such DID (NOTE: we only want to have the "active" kamailio sending outbound registrations to the carrier). My idea was to have keepalived enable all the outbound registrations when the current node changes to MASTER and disable them when it changes to BACKUP. Is there a quick "enable all" or "disable all" for uac module outbound registrations or do I have to "know in advanced" all of them and enable/disable one by one? In the docs we have: *8.4. uac.reg_disable* *Disable a remote registration record based on a filter. The command has two parameter: attribute and value. The attribute can be: l_uuid, l_username, r_username or auth_username. The value is what should be matcheg against the value of the attribute in the remote registration record.* I don't know if there is an "all" or maybe if a wildcard would work, but from the description it doesn't seem to. Any ideas? Do you guys think this is taking a correct approach? All suggestions are more than welcome!! Thanks, Joel.

3 7

Request Line alases
by Wilkins, Steve 30 Aug '18

30 Aug '18

Hello All, If the below Requests from 40.40.40.40(Asterisk) is received by 10.10.10.10(Kamailio) Request Line: INFO sip:3145553313@10.10.10.10:5060;alias=30.30.30.30~43508~2 SIP/2.0 Shouldn't Kamailio forward this request to 30.30.30.30? so that the '200 OK' can sent back to Asterisk for the response. (In this scenario) Would it be ok to just force a '200 OK' back to Asterisk from Kamailio? Thank you,

2 1

Delay
by Daniel W. Graham 30 Aug '18

30 Aug '18

Recently encountered a delay issue, packets arrive on system but Kamailio doesn't process right away. Here are a few details, any direction on what I can do to narrow down is appreciated. The receive queue normally sits at 0 or has random spike, when this issue occurs receive queue remains high like the below output. Version 5.1.5 ss -4 -n -l | grep 5060 udp UNCONN 321536 0 207.241.130.2:5060 *:* tcp LISTEN 0 128 207.241.130.2:5060 *:* -dan

2 2

IPv6 2 Ipv4
by Karlheinz Knapp 30 Aug '18

30 Aug '18

Hi i serach for a config to convert IPv6 to ipv4. My Provider give me only IPv4 and i have 4000 Customer which have ipv6. Anybody have expierience and a running config ?

3 2

Interesting issue with ACK back from 200 OK INVITE
by Wilkins, Steve 30 Aug '18

30 Aug '18

Hello All, I have had an issue for quite some time where I needed to open two UDP ports (via LISTEN) for some softphone UAC's to stay connected. Funny enough there is one Provider that works with the Single UDP. I think I finally see why I was needing two listening UDP ports. When the ACK is returning from Asterisk to Kamailio, the ACK is coming back as protocol TCP, not SIP, causing the call to drop. I finally realized this but I am not sure what would cause this and why one provider phone works but not the others. Once this is resolved, I may actually say...It works! Thank you, -Steve

1 0

Notify: stateless error reply used: I'm terribly sorry, server error occurred
by Soltanici Ilie 29 Aug '18

29 Aug '18

For some reason, Voicemail Notify messages coming from Asterisk Boxes - are not forwarded to the SIP Extensions by Kamailio. This is the SIP Packet coming to the Kamailio: 2018/08/28 10:59:19.165644 1.1.1.1:5060 -> 2.2.2.2:5060 NOTIFY sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK5f628579;rport Max-Forwards: 70 Route: <sip:2.2.2.2:5060;transport=tcp;ftag=0472e823;lr=on;nat=yes> From: "asterisk" <sip:asterisk@1.1.1.1>;tag=as1833c329 To: <sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2>;tag=0472e823 Contact: <sip:asterisk@1.1.1.1:5060> Call-ID: 93183ZjE3YmI0OGYxMDNiMzhlNzVjZjRjNTc2NTMwZGNmYzk CSeq: 103 NOTIFY User-Agent: NUACOM Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@1.1.1.1 Voice-Message: 0/0 (0/0) And this is the message which I'm receiving in the Kamailio logs: [RELAY] Error when forwarding Request ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) This is the Relay route: # Forwarding Statefully if ( ! t_relay()) { xlogl( "L_ERR" , "[RELAY] Error when forwarding Request \n " ); sl_reply_error(); } exit ;

2 2

Info: Cluecon Weekly about Kamailio, today at 17:00GMT
by Daniel-Constantin Mierla 29 Aug '18

29 Aug '18

Hello, if you haven't seen the news article on kamailio.org, today at 17:00GMT I will be part of the Cluecon Weekly online video conference to present about what's new lately in Kamailio, plans for the future as well as address questions about using Kamailio together with FreeSwitch. Anyone can join via audio or video calls, more details about at: * https://www.kamailio.org/w/2018/08/kamailio-on-cluecon-weekly-aug-29-2018/ * https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 1

Kamailio consuming UDP packets very slowly leading to high RECV - Q
by sagar malam 29 Aug '18

29 Aug '18

Hello, I am using Kamailio as a SIP proxy.So it receives SIP packet from internet and forwards it to FS servers in local network.When i execute "ss" command i see very high value in RECV-Q column.I THINK IT IS NOT NORMAL.PLEASE CORRECT ME IF I AM WRONG. ==================================================== [root@fep-1 proc]# ss -u -a -n -e | grep 5060 UNCONN 0 0 10.50.8.1:5060 *:* ino:6831630 sk:fd <-> UNCONN 0 0 10.50.7.18:5060 *:* ino:6831629 sk:fe <-> UNCONN *1183104* 0 10.50.7.254:5060 *:* ino:6831627 sk:ff <-> UNCONN *84864* 0 2607:f900:1:3::254:5060 :::* ino:6831628 sk:100 v6only:1 <-> ====================================================== Initially i thought that there is something in script which must be causing kamailio to process UDP request slower but i faced same issue with a very simple script where i simply reply with stateless 200 OK for each sip request : request_route { sl_send_reply("200","OK");exit; $avp(uuid) = $rm + "-" + $ci; ........ ......... ......... ....... Server configuration : OS : CENTOS 7 Kernel : 4.16 CPU : 5 X Intel(R) Xeon(R) CPU E5-2680 v2 @ 2.80GHz RAM : 32 GB Please help me debugging this issue.Thanks in advance

4 9

Using SEMS in IMS-ZTE Network
by Mojtaba 28 Aug '18

28 Aug '18

Hello, I am working with IMS (ZTE vendor), I want to use SBC in front of IMS network for some security policy (Hide Topology), I used SEMS as SBC in this regards, (like what we have in IMS/SEMS in kamailio by Mr Carsten Bock). When the Sebscriber want to register, The Register request received by sems and the forward to IMS(ZTE), But the "503 Service Unavailable" is received from IMS. I enable "enable_reg_caching=yes" in register.sbcprofile''s configuration file in sems. If i use kamailio, instead of SEMS, The Registration request is done successfully.In Kamailio i just forward the Register request to IMS.like this: route { if (is_method("REGISTER")) { $du = "sip:172.23.1.1:5080"; route(RELAY); } } Let me know do the issue is from myself or from IMS(ZTE)? In the fillowing, I paste all SIP Register signalling. First in kamailio, second with SEMS. ================Kamailio================== 172.23.1.252.4080 > 172.23.1.1.5080: [bad udp cksum 0x5e58 -> 0x4193!] UDP, length 787 E../....@.]...............^XREGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bK018b.e091260b1b7395975a2ff1be0b9bec60.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-7c208b9d587de818-1---d8754z- Max-Forwards: 69 Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=e664e178fc12691e;transport=UDP> From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 08:20:53.114198 IP (tos 0x0, ttl 255, id 25839, offset 0, flags [none], proto UDP (17), length 685) 172.23.1.1.5080 > 172.23.1.252.4080: [udp sum ok] UDP, length 657 E...d......$..............l.SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bK018b.e091260b1b7395975a2ff1be0b9bec60.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-7c208b9d587de818-1---d8754z- To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipTR9m3KdZ*3-5-16648*daei.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 1 REGISTER Content-Length: 0 WWW-Authenticate: Digest realm="r-kh.com",domain="sip:scscf1.r-kh.com",nonce="f42b36e39bdbb33b51599924b304445a",opaque="aW1zLmNvbS5jbg==",stale=TRUE,algorithm=MD5,qop="auth" 08:20:53.120142 IP (tos 0x10, ttl 64, id 48874, offset 0, flags [none], proto UDP (17), length 1137) 172.23.1.252.4080 > 172.23.1.1.5080: [bad udp cksum 0x5f9a -> 0xcc62!] UDP, length 1109 E..q....@.\V.............]_.REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bKd08b.4d2858f41d496b471c01a28e76190390.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-a4d19146375d56f6-1---d8754z- Max-Forwards: 69 Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=e664e178fc12691e;transport=UDP> To: <sip:+985137054518@r-kh.com;transport=UDP> From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Authorization: Digest username="+985137054518(a)r-kh.com",realm="r-kh.com",nonce="f42b36e39bdbb33b51599924b304445a",uri="sip:r-kh.com;transport=UDP",response="9469852a06c01efeabf3a7a5dd1a9f3b",cnonce="79d36600575648a1cc7221d8e1b5d53d",nc=00000001,qop=auth,algorithm=MD5,opaque="aW1zLmNvbS5jbg==" Allow-Events: presence, kpml Content-Length: 0 08:20:53.203871 IP (tos 0x0, ttl 255, id 25842, offset 0, flags [none], proto UDP (17), length 1118) 172.23.1.1.5080 > 172.23.1.252.4080: [udp sum ok] UDP, length 1090 E..^d......p.............J.5SIP/2.0 200 OK Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bKd08b.4d2858f41d496b471c01a28e76190390.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-a4d19146375d56f6-1---d8754z- To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipjRydzgMa*3-5-16648*daej.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 2 REGISTER Contact: <sip:+985137054518@192.168.122.1:64407;transport=UDP;rinstance=e664e178fc12691e>;expires=1800 Contact: <sip:ZTEAGCF000f425704be720000@agcf2.r-kh.com:5079;appid=2-7;UserRegInfo=002-02-00-0b80-04-172.30.48.143;BACaddr=172.30.48.143;BACport=2944>;expires=1800;+sip.instance="<urn:uuid:172.30.35.41-000>";reg-id=1 P-Associated-URI: <tel:+985137054518> P-Associated-URI: <sip:+985137054518@r-kh.com> Date: Sat, 25 Aug 2018 12:25:17 GMT Authentication-Info: nextnonce="00cfbd64b136c12a4e27b67fd28d19fb",qop=auth,rspauth="91ca956c5eb6ee9f1934e2e770750077",cnonce="79d36600575648a1cc7221d8e1b5d53d",nc=00000001 Content-Length: 0 ============================SEMS======================== 172.23.1.252.5080 > 172.23.1.1.5080: [bad udp cksum 0x5e16 -> 0xe11e!] UDP, length 721 E...C.@.@.................^. REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:5080;branch=z9hG4bK~0uk0ahN;rport From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=34384541-5B814A44000AE296-4B5F5700 To: <sip:+985137054518@r-kh.com;transport=UDP> CSeq: 10 REGISTER Call-ID: 37B52BC0-5B814A44000AE592-4B5F5700 Route: <sip:172.23.1.1:5080;lr> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=868560f8328d39e9;transport=UDP>;expires=3600 Content-Length: 0 08:23:32.790414 IP (tos 0x0, ttl 255, id 35790, offset 0, flags [none], proto UDP (17), length 593) 172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 565 E..Q.....................=..SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bK~0uk0ahN To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipDEmH2AWT*3-5-16648*egei.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=34384541-5B814A44000AE296-4B5F5700 Call-ID: 37B52BC0-5B814A44000AE592-4B5F5700 CSeq: 10 REGISTER Content-Length: 0 WWW-Authenticate: Digest realm="r-kh.com",domain="sip:scscf1.r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",opaque="aW1zLmNvbS5jbg==",stale=TRUE,algorithm=MD5,qop="auth" 08:23:32.855067 IP (tos 0x0, ttl 64, id 17364, offset 0, flags [DF], proto UDP (17), length 1071) 172.23.1.252.5080 > 172.23.1.1.5080: [bad udp cksum 0x5f58 -> 0xf553!] UDP, length 1043 E../C.@.@................._X REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:5080;branch=z9hG4bKeAfKkaZa;rport From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5A64C27D-5B814A44000CD034-4B6F6700 To: <sip:+985137054518@r-kh.com;transport=UDP> CSeq: 10 REGISTER Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700 Route: <sip:172.23.1.1:5080;lr> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Authorization: Digest username="+985137054518(a)r-kh.com",realm="r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",uri="sip:r-kh.com;transport=UDP",response="017efb986e17506a82bb01e8eb136b90",cnonce="3764fd5eb2ea9539d61ec14196a2bf00",nc=00000001,qop=auth,algorithm=MD5,opaque="aW1zLmNvbS5jbg==" Allow-Events: presence, kpml Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=868560f8328d39e9;transport=UDP>;expires=3600 Content-Length: 0 08:23:32.860417 IP (tos 0x0, ttl 255, id 35804, offset 0, flags [none], proto UDP (17), length 508) 172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 480 E...........................SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bKeAfKkaZa To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipxxsD3FR2VG8e9zxR_1dx*1-1-16648*ihad.1 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5A64C27D-5B814A44000CD034-4B6F6700 Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700 CSeq: 10 REGISTER X-ZTE-Cause: "CSCF-1.1.BC004016-BC004288-0-0.pcscf1.r-kh.com" Retry-After: 30 Content-Length: 0 -- --Mojtaba Esfandiari.S

2 3

Parmas in lcr_gw
by Ashutosh chaubey 28 Aug '18

28 Aug '18

Hi, How to give username and password for the gateway in params column in lcr_gw? Regards, Ashutosh

2 1

regular expression in kamailio
by Wilkins, Steve 25 Aug '18

25 Aug '18

Hi All, I am trying get the 10digit number called in on using the following $var(caller) = $(fu{re.subst,[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]}); I thought I could use a regular expression for the expression in re.subst,expression Thank you, -Steve

2 8

Listner problems
by Wilkins, Steve 25 Aug '18

25 Aug '18

Hi Joel, Actually, I have a strange issue. If I have a single UDP IP:Port listening, most Providers/Phones have two-way Audio/Video, however, with some Providers/Phones, I need to have two UDP IP:Port listners in order to get two-way Audio/Video. Here is the strange thing. If I enable two, the Providers/Phones that worked when I had a single listener only have one-way Audi/Video. I use Kamailio 5.1 and Asterisk 15.3 (pjsip). This behavior is so strange, Thank you, -Steve

2 2

add listen dynamically
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hi All, Is it possible to add or remove "listen" dynamically? Thank you, -Steve

2 2

Struggling with RTPEngine
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hi Florian, Thank you for your response. I checked and direct-media is off. Just as a recap, here is where I am I just can't get rtpengine to work. I have tried multiple configurations, but to no avail. Note that calls work good if rtpengine is disabled. Here is my setup => Public IP: 20.20.20.20 Private IP 10.10.10.10 flow => webrtc client <-> kamailio+rtpengine <-> asterisk <-> kamailio <-> legacy sip phone rtpenngine startup (I have tried a few different startups) => rtpengine --interface=int/10.10.10.10 --interface=ext/10.10.10.10\!20.20.20.20 --listen-ng=127.0.0.1:12221 --pidfile=/var/run/rtpengine --dtls-passive -f -m 10000 -M 20000 -E kamailio => Invites: rtpengine_manage("trust-address replace-origin replace-session-connection direction=ext direction=int ICE=remove RTP/AVP"); Reply's: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); I have tried direction ext ext; and many other combinations, each producing its own incorrect behavior. Thanks again, Steve

1 0

Struggling with RTPProxy and RTPEngine
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hello all, I can't seem to get either RTPProxy or RTPEngine to work correctly. I have decided to concentrate on RTPEngine because I have read that it works the best. I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine being hit but I continually get the error "Call-ID not found", calls work but media traffic doesn't go through RTPEngine. I have tried many combination of the flags but I seem to always get the same error. When I look at the tcpdump log, I see that the media offer is not on the RTPEngine port. Is there a common mis-configuration error that can cause this? I know I can send all my logs and configurations but I really want to try and resolve this as a learning experience. Thanks all, -Steve

8 17

Send a BYE for the call
by Nicolas Breuer 23 Aug '18

23 Aug '18

Hello, Any ideas how to cut a call in progress ? How to send a forced "BYE" to all parties ? Thanks

2 2

Kamailio 5.1.4 crashes
by Abdulaziz Alghosh 23 Aug '18

23 Aug '18

Hi Everyone, I am trying to start kamailio 5.1.4 with two own modules. After /etc/init.d/kamailio start , I am getting one error of the dispatcher module and bunch of warrnings because of missed reply and failure routes. The first error before crashing is: ERROR: dispatcher [dispatcher.c:814]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Later on, when I am trying to register a user (from a soft phone) i am getting the following: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) Aug 22 10:02:52 sgw1a /sbin/kamailio[6960]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 20 Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ALERT: <core> [main.c:746]: handle_sigs(): child process 6942 exited by a signal 11 Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ALERT: <core> [main.c:749]: handle_sigs(): core was not generated Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: INFO: <core> [main.c:771]: handle_sigs(): terminating due to SIGCHLD Aug 22 10:02:52 sgw1a /sbin/kamailio[6959]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6958]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6957]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6956]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6955]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6954]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6953]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6950]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6941]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6944]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6948]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6951]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6946]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6952]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6943]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6949]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6947]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6938]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6937]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6940]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6939]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a kernel: [296301.271457] kamailio[6942]: segfault at 0 ip (null) sp 00007fffb3ddf368 error 14 in kamailio[400000+442000] Aug 22 10:02:52 sgw1a /sbin/kamailio[6945]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6960]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6936]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ERROR: jsonrpcs [jsonrpcs_fifo.c:594]: jsonrpc_fifo_destroy(): cannot delete the fifo (/tmp/kamailio_jsonrpc.fifo): Operation not permitted Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ERROR: jsonrpcs [jsonrpcs_sock.c:511]: jsonrpc_dgram_destroy(): cannot delete the socket (/tmp/kamailio_rpc.sock): Operation not permitted Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Regarding to the first error of the dispatcher module i could not find out which calibration i can follow with its parameters to get rid of this error. Regarding the jasonrpcs and the last two erros, here are my configuration in kamailio.cfg: #/* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_jsonrpc.fifo") modparam("jsonrpcs", "fifo_mode", 0755) modparam("jsonrpcs", "fifo_group", "kamailio") modparam("jsonrpcs", "fifo_user", "kamailio") #/* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") modparam("jsonrpcs", "dgram_mode", 0755) modparam("jsonrpcs", "dgram_user", "kamailio") modparam("jsonrpcs", "dgram_group", "kamailio") I think the root cause is even beyond the dispatcher and jsonrpcs modules setting. Please check the attached where you can find my complete log file after starting Kamailio. Best regards Abdulaziz

2 4

Dispatcher - Force one end point active
by Marrold 23 Aug '18

23 Aug '18

Hi, We are currently using the dispatcher module and probing end points with OPTIONS, and making them active / inactive depending on the response: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;code=404") modparam("dispatcher", "ds_ping_from", "[REDACTED]") modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_probing_mode", 1) One of our suppliers endpoints doesn't respond to SIP OPTIONS, is it possible to force a single endpoint as active, but continue probing the rest? I've tried various settings - both globally and flags for the individual end point, but I couldn't seem to find the correct combination. Thanks Matthew

2 2

Re: [SR-Users] Fail to reload htable
by Henning Westerholt 23 Aug '18

23 Aug '18

Am Montag, 20. August 2018, 20:58:43 CEST schrieb Soltanici Ilie: > In the logs, I see these errors: > > ERROR: <core> [db.c:463]: db_use_table(): invalid parameter value > Aug 20 19:56:41 kamailio[12716]: ERROR: <core> [db.c:463]: db_use_table(): > invalid parameter value Aug 20 19:56:41 kamailio[12716]: ERROR: htable > [ht_db.c:211]: ht_db_load_table(): failed to use_table Hi, please keep the list in CC. This error is related to a database issue. Please double check if there are any issues during startup in the log. Do you see any other error there? A similar (not 100% identical) issue was discussed recently here: https://lists.kamailio.org/pipermail/sr-users/2018-August/102393.html Maybe this helps you as well. Best regards, Henning -- Henning Westerholt https://skalatan.de/blog/

4 7

Re: [SR-Users] Struggling with RTPProxy and RTPEngine
by Wilkins, Steve 23 Aug '18

23 Aug '18

Good morning Daniel, If RTPEngine is on the same server as Kamailio (Asterisk being on another server), and RTP traffic is sent to and from RTPEngine, then the provider only needs to whitelist one IP-Address. I thought with RTPEngine that all RTP traffic would go through it and then it would pass it on to the correct destination. Is this correct? Thank you -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Daniel Tryba Sent: Thursday, August 23, 2018 4:36 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine MITRE WARNING: Do not open unexpected password-protected attachments. >>>Email originates from a non-MITRE system. Use caution.<<< On Wed, Aug 22, 2018 at 05:05:02PM +0000, Wilkins, Steve wrote: > The SIP traffic is working this way for me but I still see RTP traffic going directly from Asterisk to the UAC, which means they need to whitelist asterisk IP. Am I missing something? In what sense do they need whitelisting? In a common NATed solution where is no white/blacklist needed. UA gets RTP endpoints from SDP, starts sending packets to ip/port and the destination will send back packets to the source ip/port, the router/firewall will just send this to the actual UA. I have yet to find an UA that cares about where the RTP stream is coming from with regards to the SIP traffic.

2 1

SIP and RTP Proxy for multiple Servers
by sewlist＠gmail.com 23 Aug '18

23 Aug '18

Im very new to kamailio, we have some ptime issues which causes one way audio due to varies PBX vendors connecting to our class 5 softswitches I would like to use Kamailio as the SIP/RTP proxy in front of our class 5 soft switches I followed the example of freeswitch and kamailio. However the config is still hard for me below setup works, however RTP not being proxied, Looks like its its not, because im not using NAT, so RTP goes direct https://freeswitch.org/confluence/display/FREESWITCH/Kamailio+basic+setup+a… I would like to to ensure RTP proxy is always used(possible re-framing ptime) and we have multiple soft switches Is it possible to setup kamailio.cfg so my customers can point the endpoint to kamailio and domain names contains the server we want to use to register with? eg, softswitch 1 = 1.1.1.2 softswitch 2 = 1.1.1.3 softswitch 3 = 1.1.1.4 etc etc my current config #!KAMAILIO #!define IPADDRESS "1.1.1.1" #!define SWITCH_IPADDRESS "1.1.1.2" #!define FLAG_FROM_SWITCH 1 #!define FLAG_FROM_USER 2 # ------------------ module loading ---------------------------------- loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "path.so" loadmodule "rtpproxy.so" # ----------------- setting module-specific parameters --------------- modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") # ------------------------- request routing logic ------------------- #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif # main routing logic route { # per request initial checks route(SANITY_CHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } route(CHECK_SOURCE_IP); ################################## ### HANDLE SEQUENTIAL REQUESTS ### route(WITHINDLG); ############################### ### HANDLE INITIAL REQUESTS ### t_check_trans(); if (is_method("INVITE|REFER")) { record_route(); } if (is_method("REGISTER")) { add_path(); } if (isflagset(FLAG_FROM_SWITCH)) { # don't send INVITE from SWITCH back to SWITCH, set reply route to handle NAT and forward them along t_on_reply("EXTERNAL_REPLY"); } else { # set destination to your SWITCH $du = "sip:1.1.1.2:5060"; } route(RELAY); } route[SANITY_CHECK] { if (!sanity_check()) { #xlog("L_WARN", "$ci|end|message is insane"); exit; } if (!mf_process_maxfwd_header("10")) { #xlog("L_WARN", "$ci|end|too much hops, not enough barley"); send_reply("483", "Too Many Hops"); exit; } if ($ua == "friendly-scanner" || $ua == "sundayddr" || $ua =~ "sipcli" ) { #xlog("L_WARN", "$ci|end|dropping message with user-agent $ua"); exit; } if ($si == IPADDRESS) { #xlog("L_WARN", "$ci|end|dropping message"); exit; } } route[CHECK_SOURCE_IP] { if ($si == SWITCH_IPADDRESS) { setflag(FLAG_FROM_SWITCH); } else { setflag(FLAG_FROM_USER); } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(RELAY); } else { if (is_method("NOTIFY")) { route(RELAY); } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard #xlog("ACK without matching transaction ... ignore and discard"); exit; } } sl_send_reply("404","Not here"); } exit; } } onreply_route[EXTERNAL_REPLY] { route(NAT_TEST_AND_CORRECT); } route[NAT_TEST_AND_CORRECT] { if (nat_uac_test("3")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } force_rport(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { #setbflag(FLB_NATB); route(RTPPROXY); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif route(RTPPROXY); /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; # if (!t_relay()) { # sl_reply_error(); # } # exit; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { #unforce_rtp_proxy(); force_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; }

2 1

RTPEngine wow's
by Wilkins, Steve 22 Aug '18

22 Aug '18

Hello all, I am still trying to get RTPEngine to work. At this point, when I make a call, I do see the offer in the rtpengine log, I also see the update of the SDP (c) changing it to 20.20.20.20. However, after that, there are no logs other than a bunch of "timer run time = 0.nnnnnnn sec", and the delete when I hang up the call. There is no RTP traffic what so ever in the rtpengine log (however, the call is completely successful, other than the RTP traffic not going through RTPEngine). Here is a breakdown of my configs (I borrowed this format from Maxim Fedorov because I thought it was very concise) Starting rtpengine => rtpengine --interface=internal/21.21.21.21--interface=external/21.21.21.21\!20.20.20.20 --listen-ng=127.0.0.1:12221 --dtls-passive -f -m 10000 -M 20000 -E -L 7 --log-facility=local1 RTPEngine Public IP: 20.20.20.20 RTPEngine Private IP: 21.21.21.21 kamailio.cfg modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:12221") listen=tcp: 21.21.21.21:5060 advertise 20.20.20.20:5060 listen=udp: 21.21.21.21:5060 advertise 20.20.20.20:5060 rtpengine_manage call rtpengine_manage("trust-address replace-origin replace-session-connection external external media-address=20.20.20.20"); (I have placed this rtpengine_manage call at the beginning of all routes). I have tried many variations of rtpengine_manage parameters, and placement of the call itself, but I never see RTP traffic in the rtpengine log. Thank you to all who have looked at this, -Steve

1 0

Error at startup
by Nicolas Breuer 22 Aug '18

22 Aug '18

Hello, How can i clear this error ? Aug 22 13:05:56 proxy1 kamailio: ERROR: <core> [core/pvapi.c:228]: pv_table_add(): pvar [rtpstat] already exists Aug 22 13:05:56 proxy1 kamailio: ERROR: <core> [core/pvapi.c:513]: register_pvars_mod(): failed to register pseudo-variable <rtpstat> for module rtpproxy

2 2

Releasing Kamailio v5.1.5
by Daniel-Constantin Mierla 22 Aug '18

22 Aug '18

Hello, with the summer holidays in the northern hemisphere approaching the end, I am considering to release v5.1.5 out of Kamailio git branch 5.1 -- there were several fixes done since 5.1.4 and it is time to package another minor release. The propose date is next week on Wednesday, August 22, or slightly after in case there are unexpected events delays it. Should there be any issues you are aware of and not yet reported on the bug tracker, do it asap in order to have a chance to be fixed: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

1 2

Encoding mechanism in UAC Module
by Duarte Rocha 21 Aug '18

21 Aug '18

Greetings, Is there a way to choose the encoding algorithm that builds the vsf and vst parameters in the Record-Route? What is the default method that Kamailio uses for this ? Thanks in advance, Duarte Rocha

2 1

invalid version 8 for table aliases found, expected 9
by Abdulaziz Alghosh 21 Aug '18

21 Aug '18

Hi everyone, i am trying to start kamailio 5.1.4 after the migration from Kamailio 3.0.3 but getting newly the following error: ERROR: <core> [db.c:450]: db_check_table_version(): invalid version 8 for table aliases found, expected 9 (check table structure and table "version") Allegedlly, this error is because of a wrong version of "aliases" table. I followed the Upgrade process starting from 3.0.3 till 5.1.x and I altered the structure of my aliase table accordinglly. The last new version of aliases is 8 as it is described at the upgrade from 4.2.x to 4.3.0 Even though, the error here says that the newest version is 9: May someone help me to find out if the aliases table's structure was renewed in between ?? Is there any modification in the data types? Thanks in advance Abdulaziz

3 4

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed
by Abdulaziz Alghosh 21 Aug '18

21 Aug '18

Greetings, i am at the final stages of emigrating from kamailio 3.0.3 to 5.1.0 including our own module. Unluckily, after writing the command " kamctl start", the following error is popping up: * PID file /var/run/kamailio.pid does not exist -- Kamailio start failed* Thus, I used the command " kamailio -M 8 -E -e -dd " to get more debugging Info and the output is detailing the following: 0(10334) ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on '192.168.1.207' (4) 0(10334) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool 0(10334) ERROR: ABDUL_functions [ABDUL_chk.c:1069]: init_db(): failed to connect database URL=mysql://ABDUL:ABDUL@192.168.1.207/ABDUL 0(10334) ERROR: ABDUL_functions [ABDUL_functions_mod.c:296]: mod_init(): Module ABDUL_functions not correktly initialized, pid[10334] 0(10334) ERROR: <core> [core/sr_module.c:991]: init_mod(): Error while initializing module ABDUL_functions (//lib64/kamailio/modules/ABDUL_functions.so) Seemingly the error relates to old databases url ( "URL=mysql:// ABDUL:ABDUL@192.168.1.207/ABDUL" ) inside an old Kamailio.cfg file from my 3.0.3 package which I put under another different directory. This old Kamailio.cfg file #includes my own module, so he is exporting this url. Consequently, my module is trying to connect again to this databases. I do not know why both kamailio.cfg (new: /etc/kamailio/kamailio.cfg and old one out of /etc/kamailio) are both seen for kamctl or (/sbin/kamailio). I know that my new url inside the new /etc/kamailio/kamailio.cfg is read and connected but again not sure why my hidden old kamailio.cfg is still being read as well. the attached shows my init file for "SUSE Linux Enterprise Server 11 (x86_64)" and my used kamailio *default *file under /etc/default. Thanks in advance and best regards Abdulaziz

2 2

help
by Pravin . 21 Aug '18

21 Aug '18

Hello Team, I want to install Kamailio SIP server on centOS server...do we need to download kamailio ISO file and install on server..pls provide the installation procedure/guidelines. Regards Pravin

6 14

Dialplan priority
by igor.potjevlesch＠gmail.com 21 Aug '18

21 Aug '18

Hello, I'm not sure to understand in which situation "pr" value is used. For example: if I have two identical rules except pr value: 1 with 1 and the second with 2, then the rule with pr 2 is never used. Someone can help me to understand? Thank you. The finality I'd like to accomplish is having some kind of load balacing between two values into "repl_exp" with the same inputs conditions. Thank you. Regards, Igor.

2 2

Call Transfer Scenario Using Freeswitch
by Sunil More 21 Aug '18

21 Aug '18

Hi All, I am using kamailio and freeswitch in my setup. While all users are registered on Kamailio, calls are routed to Freeswitch back to kamailio and then upstream providers. I can make user to user calls but in cases of transfer I am stuck. I am using a Polycom IP phone to test this scenario. While I initiate a attended transfer and click on transfer all the legs hangup. When I register all users on Freeswitch this works well. I am missing something on Kamailio side. Please point me in the right direction. Thanking You, Sunil More Ph : 9503338275

2 1

Enabling tcp_write_buf
by Gholamreza Sabery 21 Aug '18

21 Aug '18

Hi, In the documentation of "tcp_wq_max" option it is written that : > It has no effect if *tcp_write_buf *is not enabled. > > The interesting fact is that, there is not description of what it is and how one can enable it. There is even a question about it in the mailing lists: https://lists.kamailio.org//pipermail/sr-dev/2011-June/011521.html It seems that it is an inaccuracy in the documentation. If there is such an option how to enable it? Can anyone tell? Regards

3 2

Best practice/routing block for TCP connection errors to UAC?
by Anthony Alba 21 Aug '18

21 Aug '18

HI SR-Users Scenario: clients registered by TCP/TLS dropping off the network, kamailio stil thinks the client is registered and relays the INTIVE. The error occurs at the TCP layer so no failure route is called. ERROR: <core> [core/tcp_main.c:2690]: tcpconn_1st_send(): connect 127.0.0.1:48338 failed (RST) Connection refused ERROR: <core> [core/tcp_main.c:2700]: tcpconn_1st_send(): 127.0.0.1:48338: connect & send for 0x7f5494e906c0 failed: Connection refused (111) ERROR: tm [../../core/forward.h:251]: msg_send_buffer(): tcp_send failed WARNING: tm [t_fwd.c:1567]: t_send_branch(): sending request on branch 0 failed ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) I want to send the caller instead a graceful failure route type message as if the callee is not registered (404/486/603/408 type) but don't understand how to hook a routing block for TCP errors. Anthony

1 0

Fail to reload htable
by Soltanici Ilie 20 Aug '18

20 Aug '18

I'm trying to reload a hash table but receiving an error: error: 500 - Mtree reload failed [root@kamailio]# /usr/local/sbin/kamcmd htable.listTables { name: auth dbtable: dbmode: 0 expire: 0 updateexpire: 1 size: 256 dmqreplicate: 0 } [root@kamailio]# /usr/local/sbin/kamcmd htable.dump auth { entry: 69 size: 1 slot: { { name: username::8.8.8.8 value: 0 type: int } } } [root@kamailio]# /usr/local/sbin/kamcmd htable.reload auth error: 500 - Mtree reload failed [root@kamailio]# /usr/local/sbin/kamailio -v version: kamailio 5.1.4 (x86_64/linux) e8b2bf

2 1

Kamailio 5.1 Nathelper sipping on different transport protocols.
by Володимир Іванець 20 Aug '18

20 Aug '18

Hello everyone! I have system with Kamailio v5.1 which needs to send OPTIONS packets to all active clients. Here is full Nathelper module configuration: *modparam("nathelper", "received_avp", "$avp(s:received)")* *modparam("nathelper", "natping_interval", 30)* *modparam("nathelper", "ping_nated_only", 0)* *modparam("nathelper", "sipping_bflag", FLB_SIPPING)* *modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org <sip%3Apinger(a)kamailio.org>")* It works fine if client is connected over UDP. But as soon as I will switch to TCP or register with client that uses WS, Kamailio stops generating OPTIONS. Does anyone know if it is correct behavior or something is wrong? Thanks a lot!

3 3

remove_hf function in INVITE message
by Mojtaba 20 Aug '18

20 Aug '18

Today i encounter with strange issue in SIP signalling in Kamailio, In INVITE message, I just change Contact Header and relay message to destination, But the message is like this: 192.168.122.245.4080 > 192.168.122.174.5060: [bad udp cksum 0x7b5e -> 0x3a4d!] SIP, length: 1104 INVITE sip:8000@r-kh.com;transport=UDP SIP/2.0 Record-Route: <sip:192.168.122.245:4080;lr;ftag=8e47e95e;nat=yes> Via: SIP/2.0/UDP 192.168.122.245:4080;branch=z9hG4bK14d7.dc974309aba269b24213204d3ea085e2.0 Via: SIP/2.0/UDP 192.168.122.1:36853;rport=36853;branch=z9hG4bK-d8754z-423b431a3ebc8a22-1---d8754z- Max-Forwards: 69 sip:7000@192.168.122.1:36853;transport=UDP;alias=192.168.122.1~36853~1To: <sip:8000@r-kh.com;transport=UDP> From: <sip:7000@r-kh.com;transport=UDP>;tag=8e47e95e Call-ID: YTZmYTJmYzg0ZDNiMjM2OGUxZGYwNzg4OWY1ZmIxYzc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 241 <SDP Body> The code in Kamailio is: #!ifdef WITH_SBC_NASIM #if the Invite request is MO, forward it to Asterisk . if (is_method("INVITE") && $si!="192.168.122.245") { xlog("L_INFO", "The Invite message from $si:$sp, Forwart to Asterisk. <192.168.122.174:5060>.\n"); remove_hf("Contact"); append_hf("Contact:sip:$fU@sbc.r-kh.com\r\n"); $du = "sip:192.168.122.174:5060"; } #!endif Be notice that, i have the same procedure in REGISTER message, But it is work correct, if(is_method("REGISTER")) { xlog("L_INFO", "forward register to Asterisk\n"); remove_hf("Contact"); append_hf("Contact:sip:$fU@sbc.r-kh.com\r\n"); rewriteuri("sip:r-kh.com"); $du = "sip:192.168.122.174:5060"; route(RELAY); } Let me know what is the problem? -- --Mojtaba Esfandiari.S

2 2

Upgrade from Kamailio 4.4>5
by Nicolas Breuer 20 Aug '18

20 Aug '18

Hello, After upgrading, i cannot start anymore the service. Aug 19 10:19:57 proxy1 kamailio: ERROR: <core> [core/pvapi.c:828]: pv_parse_spec2(): error searching pvar "NULL" Aug 19 10:19:57 proxy1 kamailio: ERROR: <core> [core/pvapi.c:1032]: pv_parse_spec2(): wrong char [L/76] in [$NULL] at [4 (0)] Aug 19 10:19:57 proxy1 kamailio: ERROR: <core> [core/rvalue.c:2551]: mk_rval_expr_v(): memory allocation failure Any ideas ?

2 2

JSONRPC-S and async RPC commands
by Juha Heinanen 20 Aug '18

20 Aug '18

JSONRPC-S module README has: 2. Limitations This module does not implement asynchronous RPC commands. It is unlikely that asynchronous RPC commands will be executed from an JSON-RPC over HTTP client. Does this mean that, for example, tm.t_uac_wait command cannot be executed using JSON-RPC over HTTP? If so, why "it is unlikely that asynchronous RPC commands will be executed from an JSON-RPC over HTTP client", since those commands are currently commonly executed from XMLRPC client using xmlrpc module? It would be nice if JSONRPC-S module could be used as replacement of xmlrpc module. -- Juha

2 8

domain-based routing
by David Villasmil 19 Aug '18

19 Aug '18

Hello all, I need to set up a multi-domain kamailio, i think in terms of accepting invites to different domains is pretty simple by using the "alias" parameter, that much i have clear. My doubt comes from the fact that I need to route outgoing calls also based on the domain. As of today, i have single-domain kamailios using dispatcher to load-balance and failover said calls. I think i should be able to use dispatcher here as well, separating the dispatcher list by GID, so that Domain1->gid:1 Domain2->gid: 2 Domain3->gid: 3 and so on... I don't have registrations on this server and none are expected. So i'm thinking i will received the invites like: sip:12345@domain1.com sip:12345@domain1.com sip:12345@domain1.com and so on... How would you guys go about associating the domain with the gid list in dispatcher? Thanks a lot for your help! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

Re: [SR-Users] Mysql 5.7 vs 8.0
by Henning Westerholt 17 Aug '18

17 Aug '18

Am Freitag, 17. August 2018, 16:37:23 CEST schrieb Wilkins, Steve: > Yes, I did read that post, thank you for that; my problem is that installing > Kamailio is one of the last things I do in a script so I was hoping to find > a way to tell Kamailio to point to a different version. I guess I will > just need to go another route. Hi Steve, (add sr-users) kamailio by itself will not force the installation of a mysql version. I think this is done from the distribution package management and from the respective kamailio package. There are ways to adapt the installed dependencies during this installation, but you need to investigate it, as its depend on your linux distribution. Best regards, Henning -- Henning Westerholt https://skalatan.de/blog/

1 0

Mysql 5.7 vs 8.0
by Wilkins, Steve 17 Aug '18

17 Aug '18

Hi All, I would like Kamailio to use MySQL 5.7, however when Kamailio installs and I say I want to use MySQL, it installs 8.0. Can I someway direct Kamailio to install version 5.7? Thank you, -Steve

2 1

Registered User limit
by Philip Fleck 17 Aug '18

17 Aug '18

Is there a limit to the number of users which can be registered on Kamailio at any one time, or a configuration file which sets this. I have just installed Kamailio in a lab environment for testing purposes but I seem to only be able to register 1024 users, all remaining Register requests are ignored. I am not using a database to authenticate users or anything, just want to be able to register upto 10,000 users. regards Philip

6 6

Bogus event 8 in state 1 for dlg
by Phillman25 Kyriacou 17 Aug '18

17 Aug '18

Hi! I keep receiving the below logs on Kamailio: CRITICAL: dialog [dlg_hash.c:955]: log_next_state_dlg(): bogus event 8 in state 1 for dlg 0x7f59862cd440 [3685:10496] I understand that state 1 indicates an unconfirmed dialog, can anyone tell me what bogus event 8 is? The following is the call flow: User A ===> Kamailio ==> Provider >From the PCAP trace an INVITE is sent to the provider, the provider answers with 100 Trying, 180 Ringing and immediately an OPTIONS message with the same call id Kamailio Provider INVITE =================> 100 Trying <================= 180 Ringing <================ OPTIONS <================ Im assuming bogus event 8 is the OPTIONS message? Is an OPTIONS message normal in this case? Thanking you very much in advance for your help! Regards Phillip

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