- sr-users - kamailio.org

Kamailio 5.0 doesn't forward Authorization (Digest username) to Asterisk
by Marko Tirs 11 Apr '17

11 Apr '17

Hello all, I am trying to use Kamailio 5.0 with Asterisk (separate IP addresses, both port 5060), but Asterisk doesn't register the SIP clients which comes through Kamailio.If I register the same user directly to the Asterisk then it works! 1st REGISTER is also without Authorization (Digest name) but in the 2nd REGISTER it exists and registration goes OK. How to use Kamailio with Asterisk in the way the SIP client is registered on both of them?Thank youMarko SIP-CLIENT DIRECTLY TO ASTERISK:---… [View More]-------------------------------------------------------- 1 2017-04-01 15:20:48.331806 192.168.0.12 192.168.0.2 SIP 544 Request: REGISTER sip:192.168.0.2 (1 binding) | 2 2017-04-01 15:20:48.332998 192.168.0.2 192.168.0.12 SIP 626 Status: 401 Unauthorized | 3 2017-04-01 15:20:48.333814 192.168.0.12 192.168.0.2 SIP 700 Request: REGISTER sip:192.168.0.2 (1 binding) | 4 2017-04-01 15:20:48.335975 192.168.0.2 192.168.0.12 SIP 647 Status: 200 OK (1 binding) | Frame 3: 700 bytes on wire (5600 bits), 700 bytes captured (5600 bits) on interface 0 Ethernet II, Src: AsrockIn_0a:0b:46 (70:85:c2:0a:0b:46), Dst: Raspberr_a1:fd:f7 (b8:27:eb:a1:fd:f7) Internet Protocol Version 4, Src: 192.168.0.12, Dst: 192.168.0.2 User Datagram Protocol, Src Port: 64369, Dst Port: 5060 Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:192.168.0.2 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.0.12:64369;rport;branch=z9hG4bKPj9f8759c0537b4834bb411358d0b0a2ff Max-Forwards: 70 From: "33" <sip:33@192.168.0.2>;tag=4c586be1fcf842359e7d429abcb0b213 To: "33" <sip:33@192.168.0.2> Contact: <sip:74081936@192.168.0.12:64369>;+sip.instance="<urn:uuid:b408367e-d06b-4780-a9a0-291eb99ec83c>" Call-ID: d3c240808c934b77a9bbcb8bd4c1bd8c CSeq: 2 REGISTER Expires: 600 Supported: gruu User-Agent: Blink 3.0.0 (Windows) Authorization: Digest username="33", realm="asterisk", nonce="76ca37d8", uri="sip:192.168.0.2", response="19fc1a634abc72c41dcb516470c34fc2", algorithm=MD5 Authentication Scheme: Digest Username: "33" Realm: "asterisk" Nonce Value: "76ca37d8" Authentication URI: "sip:192.168.0.2" Digest Authentication Response: "19fc1a634abc72c41dcb516470c34fc2" Algorithm: MD5 Content-Length: 0 SIP-CLIENT TO KAMAILIO WITH FORWARD TO ASTERISK:---------------------------------------------------------------------------------------------- 11 2017-04-01 15:21:06.806347 192.168.0.13 192.168.0.2 SIP 459 Request: REGISTER sip:192.168.0.2:5060 (1 binding) | 12 2017-04-01 15:21:06.807514 192.168.0.2 192.168.0.13 SIP 608 Status: 401 Unauthorized | 13 2017-04-01 15:21:06.814819 192.168.0.13 192.168.0.2 SIP 456 Request: REGISTER sip:192.168.0.2:5060 (remove 1 binding) | 14 2017-04-01 15:21:06.815712 192.168.0.2 192.168.0.13 SIP 608 Status: 401 Unauthorized | 15 2017-04-01 15:21:06.821125 192.168.0.13 192.168.0.2 SIP 459 Request: REGISTER sip:192.168.0.2:5060 (1 binding) | 16 2017-04-01 15:21:06.821910 192.168.0.2 192.168.0.13 SIP 608 Status: 401 Unauthorized | Frame 15: 459 bytes on wire (3672 bits), 459 bytes captured (3672 bits) on interface 0 Ethernet II, Src: Micro-St_de:b2:3a (00:24:21:de:b2:3a), Dst: Raspberr_a1:fd:f7 (b8:27:eb:a1:fd:f7) Internet Protocol Version 4, Src: 192.168.0.13, Dst: 192.168.0.2 User Datagram Protocol, Src Port: 5060, Dst Port: 5060 Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:192.168.0.2:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKd101.0ad599b6000000000000000000000000.0 To: <sip:32@192.168.0.2> SIP to address: sip:32@192.168.0.2 From: <sip:32@192.168.0.2>;tag=533cb9e91f4b999cf76861cbb9ed54ed-85c4 SIP from address: sip:32@192.168.0.2 SIP from tag: 533cb9e91f4b999cf76861cbb9ed54ed-85c4 CSeq: 10 REGISTER Call-ID: 41df282f10fdc7b6-2773(a)127.0.0.1 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.0.0 (x86_64/linux)) Contact: <sip:32@192.168.0.13:5060> Contact URI: sip:32@192.168.0.13:5060 Expires: 3600 kamailio.cfg:-------------------- #!define WITH_DEBUG #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK ... #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.0.2" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.0.13" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ... # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { 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(ruri)="sip:" + $au + "@" + $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(); } [View Less]

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Info: mailing lists use lists.kamailio.org
by Daniel-Constantin Mierla 11 Apr '17

11 Apr '17

Hello, all the mailing lists related to the Kamailio project are using now lists.kamailio.org as primary domain, including sr-dev and sr-users. >From now on, the emails should be addressed to: - sr-dev(a)lists.kamailio.org - sr-users(a)lists.kamailio.org The previous domain (lists.sip-router.org) should still work, so existing discussions can go on as usual. If anyone is encountering any problems or notices some information on kamailio.org that needs to be updated, do not hesitate to … [View More]

2 3

Disable DNS
by Diego Nadares 11 Apr '17

11 Apr '17

Hi Guys, Is it possible to disable using of dns in kamailio? The thing is that not all INVITES that arrives to the proxy comes with ip 172.16.205.25/PUB_IP in the ruri. Some gateways sends a string in ruri -> @myproxysip and others sends @other.domain.com Request-Line: INVITE sip:1111111@myproxysip:5060;user=phone SIP/2.0 Request-Line: INVITE sip:1111111@other.domain.com:5060;user=phone SIP/2.0 Sometimes, myproxysip is from a public ip and sometimes is from a private ip. So, it's … [View More]

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Use of variables for force_send_socket
by Jöran Vinzens 11 Apr '17

11 Apr '17

Hi all, i try to re-arrange my config and i try to use a variable declared in modparam pv such like: modparam("pv", "varset", "externalip=s:1.2.3.4") and use it as: force_send_socket($var(externalip)); unfortunately the kamailio does not start with this configuration and telling me bad config file like: line 109, column 36: bad argument, [proto:]host[:port] expected it seems the force_send_socket is not able to get variable. Is there any other way to have static content defined and put … [View More]

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Notice: work on mailing lists server
by Daniel-Constantin Mierla 11 Apr '17

11 Apr '17

Hello, be aware that today I plan to do some maintenance on the mailing lists server, including the shifting to the lists.kamailio.org as primary domain. I expect not to have any relevant downtime, but one never knows. Maybe the archive won't be available for short time due to changes needs to be done to the web server after mailing lists server is upgraded. I will sent another notification when everything is finished. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- … [View More]

1 0

Authentication on Kamailio with MySQL
by Nicolas Pace 10 Apr '17

10 Apr '17

Hi everyone, I'm having some trouble setting up a simple Kamailio installation with user authentication. I've installed a kamailio (4.3.4) base install on an ubuntu server 16.04 (x86/64) on virtualbox, and added this lines at the beginning of the file: ``` #!KAMAILIO #! define WITH_MYSQL #! define WITH_AUTH #! define WITH_USRLOCDB ``` Also configured the db name, host, user and pass for RO and RW and run `kamdbctl create`. Everything I do was following this tutorial: http://www.kamailio.… [View More]

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Introduction
by Nicolas Pace 10 Apr '17

10 Apr '17

Hi Everyone! I just wanted to introduce myself, before blasting the list with questions (just kidding!). My name is Nicolás Pace, I'm a Community Networks advocate from Argentina. I'm working on a community project for Rhizomatica [0], which is an amazing organization that promotes Community Cellular Networks for underserved communities in Latin America (mainly mexican's first nations but also in Nicaragua, Colombia nad Brazil). We are working on a mobile app to connect the Community Cellular … [View More]

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INFO: Shifting mailing lists to kamailio.org domain
by Daniel-Constantin Mierla 10 Apr '17

10 Apr '17

Hello, I think it is time to migrate the mailing lists under kamailio.org domain, respectively use lists.kamailio.org. The lists.sip-router.org will still work, but lists.kamailio.org is intended to become the primary domain for mailing lists. The sip-router.org was the quick solution for integrating Kamailio and SER in one project back in 2008, when kamailio.org and iptel.org still needed to be operated for a while as independent projects. For few years now, Kamailio is the only project … [View More]

1 1

Transaction Lookup Failing
by Colin Morelli 10 Apr '17

10 Apr '17

Hey all, Trying to debug an issue with canceling an invite. I have two different types of clients. On one client, canceling an invite works correctly. With the other client, it t_check_trans fails. Both clients show the same request/responses: On Client A: -> INVITE <- 183 -> PRACK <- 200 -> UPDATE (from successful STUN binding) <- 200 -> CANCEL <- 200 On Client B: -> INVITE <- 183 -> PRACK <- 200 -> UPDATE (from successful STUN binding) <- 200 -… [View More]

2 3

"kamdbctl create" failed
by Tim Makarios 10 Apr '17

10 Apr '17

Hi, Following http://kamailio.org/docs/tutorials/devel/kamailio-install-guide-deb/ I got to the step where I had to run kamdbctl create Unfortunately, this failed, with the following output: > INFO: test server charset > INFO: creating database kamailio ... > INFO: granting privileges to database kamailio ... > INFO: creating standard tables into kamailio ... > ERROR 1101 (42000) at line 2: BLOB/TEXT column 'body' can't have a default value > ERROR: Creating core tables … [View More]

2 8

Carrier Route Module
by Tyler Preder 10 Apr '17

10 Apr '17

I am new to Kamailio. I've got the Carrier Route module up and running, but I'd like to match on scan_prefix length and not just the scan_prefix digits. Stated a little differently, I would like to define the scan_prefix and how many digits to expect after the scan_prefix. I am hoping that wildcards are supported in the scan_prefix field or is there another way to do this? Thanks, Tyler

2 1

Using $var inside textops function
by Olli Attila 07 Apr '17

07 Apr '17

Hello, Is it possible to use variables inside a textops search_hf function? I am trying to compare variable content against From header content and return true or false. I tried the following statement but with no luck: search_hf("From", "$var(MYVARIABLE)", "f") { ... } If this cannot be done, any suggestions on how to achieve this kind of comparsion? The variable content is pulled out from a htable entry. Cheers, --Olli

2 5

Re: [SR-Users] Via header with i attribute
by DanB 07 Apr '17

07 Apr '17

Hey Daniel, Many thanks again for premium support as always ;). Will get back to them since I understand Kamailio is RFC compatible on this one. Have a good one! DanB

3 2

RPCFIFOPATH / DEFINE_FIFO_NAME settings problem
by Ginhoux, Patrick 06 Apr '17

06 Apr '17

In my 'kamctlrc' file : ## path to FIFO file for engine RPCFIFO RPCFIFOPATH="/var/run/kamailio/kamailio_rpc_fifo" #RPCFIFOPATH="/tmp/kamailio_rpc_fifo" In my 'kamailio.cfg' : !!ifndef DEFINE_FIFO_NAME !!define DEFINE_FIFO_NAME "/var/run/kamailio/kamailio_rpc_fifo" !!endif modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 2) modparam("jsonrpcs", "fifo_name", DEFINE_FIFO_NAME) modparam("jsonrpcs", "fifo_mode", 0755) modparam("jsonrpcs", "fifo_group", "kamailio") … [View More]modparam("jsonrpcs", "fifo_user", "kamailio") kamailio doesn't start. It reports 'Permission denied' : Mar 24 17:31:21 localhost /usr/sbin/kamailio[1138]: ERROR: jsonrpcs [jsonrpcs_fifo.c:144]: jsonrpc_init_fifo_server(): Can't create FIFO: Permission denied (mode=493) Mar 24 17:31:21 localhost /usr/sbin/kamailio[1138]: CRITICAL: jsonrpcs [jsonrpcs_fifo.c:489]: jsonrpc_fifo_process(): failed to init jsonrpc fifo server Mar 24 17:31:21 localhost /usr/sbin/kamailio[1120]: ALERT: <core> [main.c:741]: handle_sigs(): child process 1138 exited normally, status=255 Mar 24 17:31:21 localhost /usr/sbin/kamailio[1130]: DEBUG: <core> [core/sr_module.c:920]: init_mod_child(): rank 4: tm Mar 24 17:31:21 localhost /usr/sbin/kamailio[1137]: DEBUG: <core> [core/sr_module.c:920]: init_mod_child(): rank -1: tm Mar 24 17:31:21 localhost /usr/sbin/kamailio[1127]: DEBUG: htable [htable.c:226]: child_init(): rank is (1) Mar 24 17:31:21 localhost /usr/sbin/kamailio[1120]: INFO: <core> [main.c:759]: handle_sigs(): terminating due to SIGCHLD Mar 24 17:31:21 localhost /usr/sbin/kamailio[1139]: DEBUG: <core> [core/sr_module.c:920]: init_mod_child(): rank -2: kex Mar 24 17:31:21 localhost /usr/sbin/kamailio[1130]: DEBUG: tm [callid.c:137]: child_init_callid(): callid: '15b1f0d63a718465-1130(a)129.227.83.108' Mar 24 17:31:21 localhost /usr/sbin/kamailio[1137]: DEBUG: tm [callid.c:137]: child_init_callid(): callid: '15b1f0d63a718465-1137(a)129.227.83.108' Mar 24 17:31:21 localhost /usr/sbin/kamailio[1127]: DEBUG: <core> [core/action.c:1656]: run_child_one_init_route(): attempting to run event_route[core:worker-one-init] Mar 24 17:31:21 localhost /usr/sbin/kamailio[1136]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1135]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1134]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1133]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1132]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1131]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1129]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1128]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1120]: ERROR: ctl [ctl.c:387]: mod_destroy(): ERROR: ctl: could not delete unix socket /var/run/kamailio//kamailio_ctl: Permission denied (13) Mar 24 17:31:21 localhost /usr/sbin/kamailio[1120]: ERROR: jsonrpcs [jsonrpcs_fifo.c:595]: jsonrpc_fifo_destroy(): FIFO stat failed: Permission denied If I replace the values in the 2 files as appropriate : In the 'kamctlrc" to RPCFIFOPATH="/tmp/kamailio_rpc_fifo" In the 'kamailio.cfg" to !!define DEFINE_FIFO_NAME "/tmp/kamailio_rpc_fifo" Then kamailo starts : [root@vm-vse02-siprouter1 ~]# ps -ef |grep kam kamailio 1235 1 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1236 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1237 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1238 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1239 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1240 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1241 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1242 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1243 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1244 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1245 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1246 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1247 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1248 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio root 1251 1165 0 17:37 pts/0 00:00:00 grep --color=auto kam and I can get result from kamctl/kamcmd commands : [root@vm-vse02-siprouter1 ~]# kamctl dispatcher dump which: no gdb in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/) { "jsonrpc": "2.0", "result": { "NRSETS": 1, "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": "sip:cs1-tool-misc.orange-voicemail.net:5060", "FLAGS": "AP", "PRIORITY": 0 } }] } }] }, "id": 1301 } [root@vm-vse02-siprouter1 ~]# kamcmd dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:cs1-tool-misc.orange-voicemail.net:5060 FLAGS: AP PRIORITY: 0 } } } } } Now, if I change the fifo patch and name to "/var/run/kamailio/kamailio_rpc_fifo' and apply the following rights on /var/run/ to: chmod 755 kamalio/ chown + kamailio:kamailio kamailio/ then kamailio starts. Is there a reason for these results ? Thanks in advance for your answer. Cordialement Patrick GINHOUX [View Less]

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RTPengine + kernel module long call RTP drops
by Anthony Joseph Messina 05 Apr '17

05 Apr '17

Recently, I've had to be on a lot of long conference calls and have noticed that after about 18-22 minutes, the audio on the call drops out. There are no errors logged from Asterisk, Kamailio, or RTPengine. The signalling portion of the call is unaffected and continues as normal. My setup is as follows asterisk-14.3.0-7.gita047b25.fc25.x86_64 (14 branch) kamailio-5.0.0-5.git2608015.fc25.x86_64 (5.0 branch) kernel-4.10.8-200.fc25.x86_64 (stable Fedora kernel) rtpengine-5.1.1.1-9.git27af349.… [View More]

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Kamailio v5.0.1 Released
by Daniel-Constantin Mierla 05 Apr '17

05 Apr '17

Hello, Kamailio SIP Server v5.0.1 stable release is out! This is a maintenance release of the latest stable branch, 5.0, that includes fixes since the release of v5.0.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous 5.x.x versions are strongly recommended to be upgraded to v5.0.1. For more details about version 5.0.1 (including links and guidelines to download the tarball or from … [View More]

1 0

nathelper: ping options over tcp
by Abdoul Osséni 05 Apr '17

05 Apr '17

Hi, Is it possible to send ping OPTIONS over tcp or tls? If yes, could you me how? Regards Abdoul.

5 7

Replacing a message body received with a subsequent message sending out of kamailio
by narayan panigrahi 05 Apr '17

05 Apr '17

Hi All, I have a architecture of pjsip,kamailio,freeswitch and one external server. When kamailio is receiving "407"-proxy authentication required from the external server it sends to freeswitch and received "407" from freeswitch.But when replying back to kamailio the freeswitch is adding some extra headers in that "407". Now I need to send the message from kamailio to pjsip the same thing received from external server.So I need to replace the message received from external server with the … [View More]

2 1

Kamailio 5.0.0 KEMI Lua event routes
by José Seabra 05 Apr '17

05 Apr '17

Hi all, I'm starting using kamailio 5 and was trying to convert my actual routing scripts to new KEMI interface using Lua. In my old script i'm using event routes for the evapi, dialog and dispatcher modules, so i was wondering how can i use it on KEMI. It exports some special Lua function to the KEMI framework? Can anyone give an example please. Many thanks José Seabra

2 3

Dialog Timers- Session Refresh
by Yasin CANER 05 Apr '17

05 Apr '17

2 1

Kamailio Issue
by 노형균 05 Apr '17

05 Apr '17

Hello all, I’m newbie to Kamailio and I’m having problem with installation. I tried to install Kamailio through '/usr/local/sbin/kamdbctl create’ but it showed error that database engine not specified. When I checked the kamctlrc file, the DBENGINE was already MYSQL. (DBENGINE=MYSQL) What am I missing here?? Cheers, Hyoung Kyoon Roh

3 2

Kamailio database fail
by 노형균 05 Apr '17

05 Apr '17

Hello all, I’m having problem again. After solving DBENGINE=MYSQL, I’ve been asked to type password but still get this message. I’ve tried to install mysql-server but it showed the same. What am I missing here? root@ubuntu:/home/parallels# kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/… [View More]

2 1

Re: [SR-Users] Via header with i attribute
by DanB 05 Apr '17

05 Apr '17

Hey Daniel, Indeed, call comes in over tcp and goes out over udp. They claim that it is not RFC compatible to add the i. Thanks, DanB

2 1

Planning Kamailio v5.0.1
by Daniel-Constantin Mierla 05 Apr '17

05 Apr '17

Hello, I am considering to release Kamailio v5.0.1 sometime next week, likely on Wednesday, April 5, 2017. Should anyone be aware of issues not listed yet on bug tracker, report them there as soon as possible to try to fix. Soon after, we should release a new version from branch 4.4 and the last one from branch 4.3. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World … [View More]

3 5

Problems with avp_db_load
by Kevin Bushong 04 Apr '17

04 Apr '17

I am using the avpops module to read from a mysql database to look for CFWD information for the call. When a call is placed, it appears the command avp_db_load attempts to read the file but all values come back as NULL. There are no messages in the log and of course the call does not process correctly. We are using a config that is exactly the same on a 4.3.4 version of kamailio using cento 6.7, with a local version of mysql Ver 14.14 Distrib 5.1.73, that works perfectly. When moving the … [View More]code to kamailio 4.4.4 on Debian 8, Jessie, mysql Ver 14.14 Distrib 5.5.53 the avp_db_load returns no information. Here is the snippet of the code we are using. Since I am not used to troubleshooting kamailio, please let me know what other information you need. Thanks, Kevin Code snippet: #!ifdef WITH_CFWD xlog ("$TF Call to $rU made it into WITH_CFWD in route[LOCATION] block\n"); # Call forwarding is enabled, so lets do a lookup to pull all AVP's for possible call forwarding types/destinations # Below line should load all AVP's into memory from DB for the username portion of the TO: field from table usr_preferences xlog("script should load from DB usr_preferences next"); avp_db_load("$tU","s/usr_preferences"); # This line i simple want to print to log when the avp load is complete and what the values are (for debug purposes only and should typically be commented out) xlog("DB load should be complete"); xlog("SCRIPT: $TF Call Forwarding Unconditional set to $avp(Unconditional)\n"); xlog("SCRIPT: $TF Call Forwarding NoAnswer set to $avp(NoAnswer)\n"); xlog("SCRIPT: $TF Call Forwarding Busy set to $avp(Busy)\n"); avp_print(); # Check to see if the AVP's are set and if so put them in log for logging purposes if(is_avp_set("$avp(Unconditional)")) xlog("SCRIPT: $TF Call Forwarding Unconditional set to $avp(Unconditional)\n"); if(is_avp_set("$avp(NoAnswer)")) xlog("SCRIPT: $TF Call Forwarding NoAnswer set to $avp(NoAnswer)\n"); if(is_avp_set("$avp(Busy)")) xlog("SCRIPT: $TF Call Forwarding Busy set to $avp(Busy)\n"); t_set_fr(16000, 15000); # set fr invite timeout to 16s and no response at all to 15s to meet the specs for call fowarding no answer ringtime # If Call fowarding Unconditional is set, we need to change the dialed number and sent it off to the original users trunk from subscriber table # Check to see if unconditional is set if(is_avp_set("$avp(Unconditional)")) # Shove this call into route [PSTN_CFWD] Block route("PSTN_CFWD"); Here is the syslog for the same snippet: 09:29 2017 Call to 15703771055 made it into WITH_CFWD in route[LOCATION] block Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: script should load from DB usr_preferences next Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: DB load should be complete Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: SCRIPT: Wed Mar 29 14:09:29 2017 Call Forwarding Unconditional set to <null> Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: SCRIPT: Wed Mar 29 14:09:29 2017 Call Forwarding NoAnswer set to <null> Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: SCRIPT: Wed Mar 29 14:09:29 2017 Call Forwarding Busy set to <null> Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: INFO: avpops [avpops_impl.c:1473]: ops_print_avp(): p=0x7f440ed86f90, flags=0x0113 Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: INFO: avpops [avpops_impl.c:1477]: ops_print_avp(): #011#011#011name=<direction> Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: INFO: avpops [avpops_impl.c:1485]: ops_print_avp(): #011#011#011val_str=<INCOMING / 8> [View Less]

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Why these 2 items in the same entry for a htable
by Ginhoux, Patrick 04 Apr '17

04 Apr '17

Hi, While working on the project to migrate my kamailio.cfg script from kamailio 3.3.1 to 5.0.x, I discovered that in there are 2 items in the same htable. Below the figure that illustrate this case : 1) Content of the MBXRANGE table : mysql> select * from mbxrange; +-----+-------------+-------------------------------------------------+----- -------+----------+ | id | key_name | key_value | value_type | key_type | +-----+------… [View More]-------+-------------------------------------------------+----- -------+----------+ | 0 | VERSION | 120104-173400 | 0 | 0 | | 101 | 1 | min=0100000000;max=0199999999;node=OPMVTS1VSE02 | 0 | 0 | | 102 | 2 | min=0200000000;max=0299999999;node=OPMVTS1VSE02 | 0 | 0 | | 103 | 3 | min=0300000000;max=0399999999;node=OPMVTS1VSE02 | 0 | 0 | | 104 | 4 | min=0400000000;max=0499999999;node=OPMVTS1VSE02 | 0 | 0 | | 105 | 5 | min=0500000000;max=0599999999;node=OPMVTS1VSE02 | 0 | 0 | | 106 | 6 | min=0600000000;max=0699999999;node=OPMVTS1VSE02 | 0 | 0 | | 107 | 7 | min=0700000000;max=0799999999;node=OPMVTS1VSE02 | 0 | 0 | | 108 | 8 | min=0800000000;max=0899999999;node=OPMVTS1VSE02 | 0 | 0 | | 109 | 9 | min=0900000000;max=0999999999;node=OPMVTS1VSE02 | 0 | 0 | | 199 | maxmbxrange | 9 | 0 | 0 | +-----+-------------+-------------------------------------------------+----- -------+----------+ 11 rows in set (0.00 sec) 2) modparam instruction used in my kamailio.cfg script: modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") 3) Result of < kamctl fifo sht_dump mbxrangeHash > command with kamailio version 3.3.x: [root@op52is4router1 ~]# kamctl fifo sht_dump mbxrangeHash Entry:: 0 6:: min=0600000000;max=0699999999;node=OPMVTS1VSE02 Entry:: 1 7:: min=0700000000;max=0799999999;node=OPMVTS1VSE02 Entry:: 2 4:: min=0400000000;max=0499999999;node=OPMVTS1VSE02 Entry:: 3 5:: min=0500000000;max=0599999999;node=OPMVTS1VSE02 Entry:: 4 2:: min=0200000000;max=0299999999;node=OPMVTS1VSE02 Entry:: 5 3:: min=0300000000;max=0399999999;node=OPMVTS1VSE02 VERSION:: 120104-173400 Entry:: 6 maxmbxrange:: 9 Entry:: 7 1:: min=0100000000;max=0199999999;node=OPMVTS1VSE02 Entry:: 14 9:: min=0900000000;max=0999999999;node=OPMVTS1VSE02 Entry:: 15 8:: min=0800000000;max=0899999999;node=OPMVTS1VSE02 Result of < kamcmd htable.dump mbxrangeHash > command with kamailio version 5.0.x : [root@vm-vse02-siprouter1 ~]# kamcmd htable.dump mbxrangeHash { entry: 0 size: 1 slot: { item: { name: 6 value: min=0600000000;max=0699999999;node=OPMMMS1VSE02 type: str } } } { entry: 1 size: 1 slot: { item: { name: 7 value: min=0700000000;max=0799999999;node=OPMMMS1VSE02 type: str } } } { entry: 2 size: 1 slot: { item: { name: 4 value: min=0400000000;max=0499999999;node=OPMMMS1VSE02 type: str } } } { entry: 3 size: 1 slot: { item: { name: 5 value: min=0500000000;max=0599999999;node=OPMMMS1VSE02 type: str } } } { entry: 4 size: 1 slot: { item: { name: 2 value: min=0200000000;max=0299999999;node=OPMMMS1VSE02 type: str } } } { entry: 5 size: 2 slot: { item: { name: 3 value: min=0300000000;max=0399999999;node=OPMMMS1VSE02 type: str } item: { name: VERSION value: 120104-173400 type: str } } } { entry: 6 size: 1 slot: { item: { name: maxmbxrange value: 9 type: str } } } { entry: 7 size: 1 slot: { item: { name: 1 value: min=0100000000;max=0199999999;node=OPMMMS1VSE02 type: str } } } { entry: 14 size: 1 slot: { item: { name: 9 value: min=0900000000;max=0999999999;node=OPMMMS1VSE02 type: str } } } { entry: 15 size: 1 slot: { item: { name: 8 value: min=0800000000;max=0899999999;node=OPMMMS1VSE02 type: str } } } So, I can see : - The entry number is not incremental 1 by 1 ; there are entry #1 to #7, then #14 and #15 (all have size=1) - For the entry #5, its size is of 2 and it contains 2 items I don't say here that this is a problem, but the way the htable is loaded seems strange to me. So if there people who can explain me if what I see is normal or not, and how the htable load process works, I would appreciate. Thanks in advance. Cordialement Patrick GINHOUX [View Less]

2 4

Via header with i attribute
by DanB 04 Apr '17

04 Apr '17

Hey Guys, Any of you know which module is adding the "i" attribute and maybe if it can be removed safely? The reason would be for some Avaya PBXes discarding the Via completely due to this attribute. Sample header: """ Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKf0d6.72eca17f202d1e73a1f519dbecf7d6a6.0;i=8f1. """ Many thanks, DanB

2 1

nosql database
by Ján Füri 04 Apr '17

04 Apr '17

Hi community, I'd like to ask that witch nosql DB is the most supported by kamalio 5 ? or witch do you guys use in production ? Thanks, Jan

2 1

Re: [SR-Users] ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 04 Apr '17

04 Apr '17

Hi, A second update for people who started to look at this issue. I have identified the real cause of the problem by looking more in the kamailio log with the debug traces. I discover that it is related to the load of the properties table in the htable. I have the following entries in my mysql db : +-------+----------------+---------------------------------------------------------+------------+----------+ | id | key_name | key_value | … [View More]value_type | key_type | +-------+----------------+---------------------------------------------------------+------------+----------+ | 0 | VERSION | 120104-1734 | 0 | 0 | | 1 | mmm-retry | 408=Y;502=Y;503=Y | 0 | 0 | | 2 | mgcf-retry | 408=Y;502=Y;503=Y | 0 | 0 | | 3 | max-mmm-retry | 2 | 0 | 0 | | 4 | max-mgcf-retry | 2 | 0 | 0 | | 5 | target-uri | mgcf=10.10.10.10;mmm-legacy=20.20.20.10;mmm=20.20.20.20 | 0 | 0 | | 6 | pike-status | 503 Service Overload | 0 | 0 | | 7 | mgcf-path | 99 | 0 | 0 | | 16 | mmm-nodeid | OPMMMS1VSE01=1;OPMMMS1VSE02=2 | 0 | 0 | | 17 | mmm-node-path | 1=1;2=2 | 0 | 0 | +-------+----------------+---------------------------------------------------------+------------+----------+ In the kamailio.cfg, the << modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") >> command causes the load of the above table in the htable. In the log we can see, that the load is done in the alphabetic order of the key_name and it fails to load the last entry "VERSION" : Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: DEBUG: <core> [db_val.c:169]: db_str2val(): converting BLOB [VERSION] Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [0] Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [0] Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: DEBUG: <core> [db_val.c:169]: db_str2val(): converting BLOB [120104-1734] Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) If I add a new entry named "WWWWW", then the failure occurs on this entry that is the last alphabetic entry : | 10000 | WWWWW | toto | 0 | 0 | Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: DEBUG: <core> [db_val.c:169]: db_str2val(): converting BLOB [WWWWW] Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [0] Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [0] Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: DEBUG: <core> [db_val.c:169]: db_str2val(): converting BLOB [toto] Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Attached the kamailio log for each test. I hope this help. Have anyone an idea of this problem ? Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : mardi 21 mars 2017 09:41 À : Kamailio users (sr-users(a)lists.sip-router.org) <sr-users(a)lists.sip-router.org> Objet : RE: ht_db_load_table(): key type must be string (type=6) Hi, There was a mistake in the file attached ; the short kamailio.cfg file (that makes kamailio to start) was not the correct one; attached the correct one. Have anyone an idea of this problem ? Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : lundi 20 mars 2017 17:58 À : Kamailio users (sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>) <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Objet : RE: ht_db_load_table(): key type must be string (type=6) Hi, To figure out what causes this issue, I work to shorten my kamailio.cfg to contains loadmodule/modparam instructions for the htable module only. My first attempt (see kamailio.cfg.ok) shows that the problem doesn't occur. In the second attempt (see kamailio.cfg.ko) I add instructions previously removed from the original .cfg file. I have determined that, depending on the instructions that follows the "modparam("htable", "fetch_rows", 1000)" line, the "key type must be string (type=6)" error occurs or not. In this ko file, if I comment out the following sequence, the error occurs : modparam("htable", "fetch_rows", 1000) #!!ifdef FIXED_ROUTING #modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") #!!else #modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") #modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") #!!endif modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") I hope my explanation is clearn and there are people who can help me on this problem that causes kamailio to not start. Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : vendredi 17 mars 2017 15:46 À : Kamailio users (sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>) <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Objet : ht_db_load_table(): key type must be string (type=6) Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX [View Less]

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Re: [SR-Users] sr-users Digest, Vol 143, Issue 5
by Narayan P 04 Apr '17

04 Apr '17

Hi Juha, I mean to say that all of the headers in a SIP message. I need to replace all the headers with the headers received before and send to the client. Thanks, Narayan ________________________________ From: sr-users <sr-users-bounces(a)lists.sip-router.org> on behalf of sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> Sent: Tuesday, April 4, 2017 10:00:02 AM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 143, Issue 5 … [View More]Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Issue in replacing message header in kamailio (Narayan P) 2. Re: Issue in replacing message header in kamailio (Juha Heinanen) ---------------------------------------------------------------------- Message: 1 Date: Tue, 4 Apr 2017 09:02:28 +0000 From: Narayan P <narayana88(a)outlook.com> To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Issue in replacing message header in kamailio Message-ID: <HK2PR0201MB1745D10E07415A9696765176D70B0(a)HK2PR0201MB1745.apcprd02.prod.outlook.com> Content-Type: text/plain; charset="iso-8859-1" Hi Can anybody help me on this.Any API to replace a message header completely in kamailio configuration file and its usage. Thanks, Narayan ________________________________ From: Narayan P Sent: Tuesday, April 4, 2017 6:31:14 AM To: sr-users(a)lists.sip-router.org Subject: Re: Issue in replacing message header in kamailio Hi All, How to replace message header completely that has been received previously in another 200 OK response message. I have used remove_body() and replace_body(re, txt) . Is there any API to replace these. Any suggestion will be highly appreciated. Thanks, Narayan ________________________________ From: Narayan P Sent: Monday, April 3, 2017 1:41:47 PM To: sr-users(a)lists.sip-router.org Subject: Issue in replacing message header in kamailio Hi All, How to replace total message header in another 200 ok response message. I have used remove_body() and replace_body(re, txt) . Is there any API to replace these. Any suggestion will be highly appreciated. Thanks, Narayan [View Less]

3 2

Issue in replacing message header in kamailio
by Narayan P 04 Apr '17

04 Apr '17

Hi All, How to replace total message header in another 200 ok response message. I have used remove_body() and replace_body(re, txt) . Is there any API to replace these. Any suggestion will be highly appreciated. Thanks, Narayan

2 3

Doubt parsing multipart SDP
by Albert Petit 03 Apr '17

03 Apr '17

Hi list, I am having some issues parsing multipart INVITE [1] when calling force_rtp_proxy . I tried to upgrade to Kamailio 5 to see if maybe i was using too old code but seems not the case. Log of error is: Apr 3 18:31:21 mcptt /usr/sbin/kamailio[3232]: INFO: <script>: forcing RTP proxy for INVITE message Apr 3 18:31:21 mcptt /usr/sbin/kamailio[3232]: ERROR: <core> [core/parser/parse_content.c:370]: decode_mime_type(): ERROR:decode_mime_type: parse error near in [… [View More]application/vnd.3gpp.mcptt-info+xml] char[51][3] offset=16 So seems there is some issue using Content Type application/vnd.3gpp.mcptt-info+xml in a multipart SIP message But this content type is accepted at following 3GPP spec: http://www.etsi.org/deliver/etsi_ts/129100_129199/129165/13.06.00_60/ts_129… Checking implementation of parse_content.c I see ". - _ +" characters are accepted so i am not sure what is failing. I did some test and if instead of that Content-Type: application/vnd.3gpp.mcptt-info+xml i use Content-Type: application/mcptt no errors are seen in Kamailio logs and ports are replaced succesfully. So do you see something wrong on what i am doing with attached info? INVITE sip:mcpttparticipating@demo.net SIP/2.0 Record-Route: <sip:10.1.1.123:6074;transport=tcp;lr=on;ftag=i3b01a;nat=yes> Call-ID: db6b10362447354bc9819df9a9046ecb(a)10.1.1.222 CSeq: 1 INVITE From: <sip:blue1@demo.net>;tag=i3b01a To: <sip:ccw@demo.net;poc-group=blues> Via: SIP/2.0/TCP 10.1.1.123:6074 ;branch=z9hG4bK15a9.1c063dec40e772152f44c026a1627082.0;i=f Via: SIP/2.0/TCP 10.1.1.222:5070 ;received=10.1.1.222;rport=55173;branch=z9hG4bK-363135-1750dc6e97437e4788f446f2d818b60a Max-Forwards: 69 Contact: <sip:blue1@10.1.1.222:55173 >;transport=tcp;+g.3gpp.mcptt;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mcptt";isfocus Accept-Contact: *;+g.3gpp.mcptt;require;explicit Accept-Contact: *;+g.3gpp.icsi-ref="urn:urn-7:3gpp-service.ims.icsi.mcptt";require;explicit P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mcptt P-Preferred-Identity: <sip:blue1@demo.net> User-Agent: PoC-client/3GPP-Rel13 DemoClient/v2.03.01/BV6000/SDK23 Session-Expires: 3600 Min-SE: 90 Supported: replaces, 100rel, timer, norefersub, timer, answermode, eventlist Resource-Priority: mcpttp.5 Content-Type: multipart/mixed;boundary=gnkboundary Content-Length: 997 --gnkboundary Content-Type: application/sdp v=0 o=- 3626700268 3626700268 IN IP4 10.1.1.222 s=spimedia c=IN IP4 10.1.1.222 t=0 0 a=X-nat:0 m=audio 10002 RTP/AVP 106 a=rtcp:10003 IN IP4 10.1.1.222 i=speech a=rtpmap:106 AMR-WB/16000 a=fmtp:106 mode-change-period=1; mode-change-capability=2; mode-change-neighbor=0; max-red=0 a=sendrecv a=ptime:100 a=maxptime:400 m=application 10003 udp MCPTT a=fmtp:MCPTT mc_queueing;mc_priority=1 --gnkboundary Content-Type: application/vnd.3gpp.mcptt-info+xml <?xml version="1.0" encoding="UTF-8"?> <mcpttinfo xmlns="urn:3gpp:ns:mcpttInfo:1.0" xmlns:xsi=" http://www.w3.org/2001/XMLSchema-instance"> <mcptt-Params> <session-type>chat</session-type> <mcptt-request-uri type="Normal"> <mcpttURI>sip:ccw@demo.net;poc-group=blues</mcpttURI> </mcptt-request-uri> <mcptt-client-id type="Normal"> <mcpttURI>urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF</mcpttURI> </mcptt-client-id> </mcptt-Params> </mcpttinfo> --gnkboundary-- Thanks! Albert [View Less]

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Doubt about MI Datagram module in Kamailio 5
by Albert Petit 03 Apr '17

03 Apr '17

Dear list, I've recently upgraded to Kamailio 5 in order to avoid issue parsing SDP multipart bodies which think are fixed in the update ( - symbol rejected in content type) . I could upgrade without issues from my kamailio 4 dev deployment but i cannot load anymore MI_Datagram module (seems not present in Kamailio 5 , MI Datagram module i refer to: http://www.kamailio.org/docs/modules/4.3.x/modules/mi_datagram.html ) Can i use MI Datagram module still in Kamailio 5 ? Thanks Albert

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Exclusive: Quitting the fast life of open-source SIP
by Alex Balashov 03 Apr '17

03 Apr '17

(Filed by UC Celeb Talk - 1 Apr 2017.) BERLIN (Tegel), GERMANY--In the now-infamous breakdown episode on the VIP tarmac at Berlin's Tegel Airport, Alex Balashov shocked the IP telecoms celebrity world last month. Shielding his eyes from the punishingly bright daylight of an overcast day in northern Germany, Balashov climbed out of his Gulfstream G650 jet, clutching a half-finished bottle of Woodford Reserve, and announced his sudden retirement from the "fast life" of open-source SIP … [View More]consulting to a waiting gathering of techn business reporters. "I'm not with this 'live fast, die young' VoIP s*** anymore," he reportedly shouted, before collapsing and vomiting profusely on the red carpet. UC Celeb Talk's own Benjamin Brant sat down with Balashov in an exclusive interview to learn more about what happened, even as the telecoms industry continues to digest the shock meltdown. -- BRANT: Everyone's dying to know, what was going through you head at that moment? BALASHOV: Funny how you mention dying. I want to live to see 35, man. I can't handle the SIP biz anymore. BRANT: What do you mean? You seemed to have all the makings of an open-source SIP superstar. BALASHOV: I can't keep up with the drinks anymore, the three day-long parties, the paparazzi, always watchin' my back at ClueCon... BRANT: When'd you realise it might be time to call it? BALASHOV: I'll tell you when. It was Kamailio World '16. Man, those vodka + Club Mate drinks they pour down at Fokus are something else. They'll really get you thinking about things. BRANT: So you had a cocktail and thought better of your entire life as a SIP celebrity? The fans, the adulation, the glory? BALASHOV: When you wake up at 3 PM at the Park Inn on Alexanderplatz with a $31,000 room service bill on your AMEX, a goat wandering around, and reporters in your face, you tell me with a straight face you're not gonna take stock of your life and make some changes. BRANT: That really happened? BALASHOV: That's what I'm trying to tell you! That's Kamailio World, dude. That's the SIP software engineering culture. You wanna ITSP and WebRTC? You're gonna have to bro down! BRANT: Surely not everywhere. What about the IETF? BALASHOV: Those guys drink more than ANYONE. Have you seen the SIP Outbound RFC? BRANT: That's the work-product of an alcohol-fuelled creative orgy? BALASHOV: It's the work product of something-something. I don't know if I'm allowed to identify the uh, substance in question. BRANT: How are you going to live without the private jet? If I had a G6, I wouldn't just turn over the keys. BALASHOV: Yeah, I mean, open-source VoIP consulting has its upsides. It's a fine way to make a living. But it's just not worth it. Besides, hourly operating costs are through the roof. BRANT: But you were an icon, and quite a ladies' man too! How do you walk away from that? BALASHOV: Look man, of course women are gonna mob you when you're a SIP consulting star. That's just part of the open-source tech life in general. But it's fake, man. They don't love you. They don't want to know who you are inside. They just want you 'cause you know all about the SIP transaction state machine, loose vs. strict routing, cross-compiling Kamailio. They just got a crush on you 'cause you know presence and parallel forking race conditions. I want something real. BRANT: But that's part of the social bargain of being an open-source SIP celebrity. Didn't you once boast of having uh, lady friends, "in different geographic and non-geographic operator codes"? BALASHOV: Yeah, and that's cool, but I want to come home to a woman who loves me for me, not Mr. 3... ... 02, Moved Temporarily, but not happy necessarily. Feel me? BRANT: What happened at ClueCon? BALASHOV: You talking about the bullets Acme Packet ventilated me with? BRANT: Five of them, as I heard. BALASHOV: Well I'm still standin'. Took those shots and smiled. BRANT: I know you guys have a kind of omertà about this stuff, but I've got to ask: why? BALASHOV: I ain't worried, kids growing up in today's world need to learn the truth. They act like I was dealing transcoding licences on their block. O.G. 729. BRANT: But that wasn't what gave you a wake-up call that the SIP game is sharper than a razor blade? BALASHOV: Hell no, it wasn't their block. That fake-ass east coast channel partner crew can cash me outside howboudah? BRANT: How exclusive is Kamailio World? BALASHOV: It's one of the most exclusive clubs I've ever hit up. Lines wrapping around blocks. And the bouncers. Damn. BRANT: The bouncers? BALASHOV: Those dudes are harsh. BRANT: On a scale of 1 to Berghain, how bad is it? BALASHOV: All I know is, one time some squirrely guy in the audience asked a moderately new-blood question like, "If a Record-Route header has been added by an intermediate proxy, is it proper of a SIP User Agent to populate the Request URI of subsequent in-dialog requests with the Record-Route URI, or should it be the remote Contact?" Half the room got up like it was ready to curb-stomp that little face. Guy didn't know what he got himself into--a room full of elites. BRANT: So what happened? BALASHOV: Someone was like, "Bounce this fool! He don't know how to act." So these three muscular Kaiser Wilhelm-looking dudes--moustaches and all--in sunglasses roll up and drag this guy through the Atrium, out the doors and dump him straight out onto Anna-Louisa-Karsch-Straße. BRANT: Damn. Ejected out of the saloon. BALASHOV: Yo, that dude was dismissed from the premises. BRANT: So what do you see in your future post-SIP life? BALASHOV: I'm thinking about starting a bakery. A friend threw it out there as a way to kind of atone for all this and get straight with Jesus. Gonna bake really big cookies. You know, huge ones, like a small pizzas? But recognise--they're actually cookies. BRANT: Got any advice for those who bright young upstarts who haven't turned over a new leaf and thrown in the towel yet? BALASHOV: Come on down to Kamailio World #5, May 8-10 2017 in Berlin. You'll see your life in a new way. Registration and more information online at https://www.kamailioworld.com/k05/ [View Less]

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Kamailio 6.0 architecture
by Olle E. Johansson 03 Apr '17

03 Apr '17

Dear Kamailians! We got news for you! The Kamailio 6.0 architecture is finalised! It's something we've been working on for a long time in the project team and you've already seen part of it in the recent Kamailio 5 release - the KEMI interface for LUA. We've realised that we have too many modules, especially all those IMS* modules. And the core is getting slimmer and slimmer. This is what we're going to do: * All RPC, EVENTAPI,AUTH, JSON, TM, HTTP_CLIENT and TLS modules will be integrated … [View More]

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Re: [SR-Users] Exclusive: Quitting the fast life of open-source
by JR Richardson 01 Apr '17

01 Apr '17

Brilliant! Bravo! Bravo! Brovo! Gonna miss you Bro! Get in this neck of the woods, got some dope a$$ hooch and BBQ waitin! Peace out! JR Richardson Engineering for the Masses Chasing the Azeotrope ---------------------------------------------------------------------- Message: 1 Date: Sat, 1 Apr 2017 01:02:58 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Exclusive: Quitting the fast life of open-source SIP BRANT: But … [View More]

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Configuring Kamailio as a "VoIP traffic reflector"
by Tahiro Hashizume 31 Mar '17

31 Mar '17

Hi there, This e-mail is going to be lengthy and I hope you will not get frustrated (like I am, for different reasons described later) by reading it all from the beginning to the end since I have just warned you :P You have absolutely no need for reading the entire post unless you are crazy enough read about what I do for fun in given circumstances. Basically, I would like Kamailio and Asterisk to run on the same box in such a way that A.)Asterisk listens on eth0 (on 10.1.1.15/24 with … [View More]ProxyARP enabled) for incoming SIP register requests from VoIP-ATAs B.)Kamailio listens on five IP addresses on dummyN which has 10.1.1.1N/32 (N=0-4, so dummy0 has 10.1.1.10/32, for example) making five SIP+RTP sessions to the same remote host to appear as if they are from five different hosts - so that a single instance of Asterisk running on debian may be registered (through Kamailio) on the same remote host by using five different accounts and source IP addresses. I know Asterisk alone cannot do this (even by running multiple instances of it) and that there have to be five instances of both SIP-Proxy and RTP-Proxy so that the source address can be literally spoofed. Should be simple, but so far cannot find good sources of reference. I need your advice. Thanks in advance. << Backgrounds - read only if you are curious >> The so-called frustration is when your needs are not realized due to technological limitations. I have years worth of frustration on the VoIP setup at home and that's totally NOT your fault. Written below for those who happen to be curious should sound interesting, so here is the situation I'm facing; I'm in Japan and our infrastructure is well within the definition of NGN which in my case includes a low-latency VoIP to replace the traditional PSTN. (NTT will end ISDN in 2020, copper-line PSTN in 2025, as announced so far.) The service is provided by NTT (former state-run telecom) of Japan and works on FTTH which carries: a.)PPPoE for those who wish to get IPv4&IPv6 connectivity - which for the most part is so that users get to choose their favorite ISP. b.)WAN-reachable IPv6 traffic c.)WAN-unreachable IPv4 for SIP based services (address blocks reserved and announced by AS9595) Given the fact that b.) and c.) are only applicable (optionally and respectively) to those who signed up for the native-IPv6 (we call it IPoE, oddly) service and to those who signed up for the VoIP service, it's essentially an L2 transport service for PPPoE to your ISP - very much like the way we had our phone-line and ISP (dial-up) separately signed up for in the old days (hence I call it the access-line service). I've made a switch over to VoIP-based PSTN from the copper-line in April 2011 after seeing that copper-line PSTN served no better in case of emergency like 3.11 when people want to call others to see if their loved one are okay - on the evening through the night of March 11 2011, our traditional PSTN stayed at its capacity limits while VoIP-based means of communication (incl. Skype, surprisingly) stayed intact. That was the wake-up call. Now, upon signing up for the VoIP service, the user receives a box so called HGW (stands for Home Gateway) which is very much a SIP+RTP UA on its WAN side and SIP+RTP server on LAN side (up to 5 UAs may register if the UA supports "sip session-timer"). And this HGW is being the pain that drove me to writing this e-mail - it only allows one SIP-register per IP address. Further, looking into the specs requirements for HGW published by NTT (=there's a centralized documentation for it since the designs and manufacturing of HGW is split across several companies), there's a part which basically says: "One SIP register to HGW from a UA on LAN should be able to handle (but not limited to) one data session. An UA on HGW's LAN side may support multiple-data session per SIP register(optional)." - which to me reasonably explains why my HGW only allows one SIP-register per IP address. My gyess is that the unwritten assumption here is "one sip register"="one physical phone", which is quite understandable. Regarding the fact that NTT optionally provides services like: 1.)additional phone numbers 2.)multi-data sessions(RTP for voice, for example) ...on one trunk. This, and also in accordance with the technical specs. requirements for HGW above means supporting multiple RTP sessions either terminated or originated by an Asterisk box which sits on the HGW's LAN side requires that a single instance of Asterisk on a debian box somehow uses multiple IP addresses (with or without the use of other software). The simplest way of doing this (to me) seemd like running five instances of RTP(proxy)+SIP(proxy) which will each listen on one of five IP addresses, to which the Asterisk instance in question could establish one session of SIP+RTP each. The idea seemed simple enough, but there did not seem like good selections of software years ago. I have once given up on the implementation years ago, altered the NTT-supplied HGW with a third-party product (NVR500 by YAMAHA which on its LAN behaves as a MGCP+RTP server) in order to maximize the stability of the phone system. This was due to the fact that the x86-box I was using was hardware-unstable and that the HGW's behavior was something I did not like dealing with (recently solved). I've decided to make a come-back to Asterisk recently since I recently started using hylafax running on a RasPi with RS232 modem hooked up to it but with missing support for CID since the Japanese equivalent of CID (we call it Number-Display) does not follow the Bellcore standards and the modem I use only supports Bellcore standards- going VoIP again and running iaxmodem could solve this issue. -------------------------------------------- Tahiro Hashizume Department of Electrical and Electronic Engineering at Tokyo Denki University 5 Senju-Asahi-cho Adachi-ku Tokyo, JP1208551 Japan -------------------------------------------- [View Less]

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[SIP Transaction] Handle Cancel message
by Hai Bui Duc Ha 31 Mar '17

31 Mar '17

Hi all, I have problem when use module pusher on Kazoo. The user after receive call from pusher can not receive the Cancel message if have another people pick up this phone. I trace log and debug on Kamailio and see the Freeswitch sent Cancel message but Kamailio reply 481 - Call leg/transaction does not exist. The configure on Kamailio: default.cfg ======================================================= *route[PREPARE_INITIAL_REQUESTS]* *{* * if (is_method("CANCEL")) {* * if(… [View More]t_lookup_cancel()) xlog("L_INFO","$ci|log|======== t_lookup_cancel()");* * xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CAPTURED IN MAIN---\n");* * if (t_check_trans()) {* * xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CHECK TRANS TRUE---\n");* * //xlog("L_INFO", "$ci|log|====================================");* * route(RELAY);* * } else {* * xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CHECK TRANS FALSE---\n");* * sl_send_reply("481", "Call leg/transaction does not exist");* * }* * exit();* * } else if (is_method("ACK")) {* * if (t_check_trans()) {* * route(RELAY);* * }* * exit();* * }* ======================================================= As I read on http://kamailio.org/docs/modules/4.3.x/modules/tm.html#tm.f.t_check_trans Kamailio see the cancel message not same the transaction INVITE message. INVITE message: ------------------------------------------------------------------------ INVITE sip:100@quydang.htk.cvoice SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:11000;rport;branch=z9hG4bKK9jyp8mXUZHgS Route: <sip:192.168.1.77:5060> Max-Forwards: 49 From: "02 quy" <sip:102@quydang.htk.cvoice>;tag=04K2SappeyDtj To: <sip:100@quydang.htk.cvoice> Call-ID: 19fa9228-ef95-11e6-b473-1f645951fd7b CSeq: 103030653 INVITE Contact: <sip:mod_sofia@192.168.1.77:11000> User-Agent: Anttel Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-AUTH-IP: 192.168.1.141 X-AUTH-PORT: 64799 X-KAZOO-AOR: sip:100@quydang.htk.cvoice X-KAZOO-PUSHER-Token-Proxy: sip:192.168.1.77:5060 X-KAZOO-PUSHER-Token-ID: 803F0D6610AC5979C6F0513A3A4BE62E0BAF1530DC7A98C2C0DC13D784585FBE X-KAZOO-PUSHER-Token-Type: apple X-KAZOO-PUSHER-Token-App: com.htkinc X-FS-Support: update_display,send_info Remote-Party-ID: "02 quy" <sip:102@quydang.htk.cvoice >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1486708670 1486708671 IN IP4 192.168.1.77 s=FreeSWITCH c=IN IP4 192.168.1.77 t=0 0 m=audio 24764 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ CANCEL message: ------------------------------------------------------------------------ CANCEL sip:100@quydang.htk.cvoice SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:11000;rport;branch=z9hG4bKK9jyp8mXUZHgS Route: <sip:192.168.1.77:5060> Max-Forwards: 49 From: "02 quy" <sip:102@quydang.htk.cvoice>;tag=04K2SappeyDtj To: <sip:100@quydang.htk.cvoice> Call-ID: 19fa9228-ef95-11e6-b473-1f645951fd7b CSeq: 103030653 CANCEL Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 ------------------------------------------------------------------------ How can I know it same transaction ? We have any idea to solve this problem ? Thank for advice ! Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876 [View Less]

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$hep(dst_ip)
by David Villasmil 30 Mar '17

30 Mar '17

Hello guys, the merged https://github.com/kamailio/kamailio/pull/995 is not available on 5.0 and I'm wondering if/when it will be added to 5.0? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Adding two time prefix to $rU
by Yebe diomande Keffa 29 Mar '17

29 Mar '17

Hello I am new with kamailio, and trying to add prefix to Request URI ($rU). I use branch_route, for $rU=”prefix”+$rU, i got two time prefix adding => prefix_prefix_original$rU This is my code small code: request_route { if (is_method("INVITE")) { &… [View More]

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log_custom module question
by Anonim Stefan 29 Mar '17

29 Mar '17

Hi, I'm trying some logging via UDP, using log_custom module. I've noticed that the logs are always sent through the first interface in the config "listen=" rows. Now, is there a way to listen on that interface, in order to log_udp() on it, without being able to send/receive any SIP on it? (i.e. disabling SIP capability and use it only for logging via UDP) Thank you, Stefan

2 1

Nightly build config error
by David Villasmil 28 Mar '17

28 Mar '17

Hello guys, I have in my apt the nightly build repo, which i use fairly regularly for testing. Today i tried installing kamailio 5 and got the following: # apt-get install -y kamailio Reading package lists... Done Building dependency tree Reading state information... Done The following packages were automatically installed and are no longer required: libdbd-mysql-perl libdbi-perl libev4 libevent-2.0-5 libjansson4 libmysqlclient18 libsctp1 libterm-readkey-perl libunistring0 mysql-client … [View More]mysql-client-5.5 mysql-common Use 'apt-get autoremove' to remove them. Suggested packages: kamailio-berkeley-modules kamailio-carrierroute-modules kamailio-cpl-modules kamailio-dbg kamailio-ldap-modules kamailio-lua-modules kamailio-mysql-modules kamailio-perl-modules kamailio-postgres-modules kamailio-presence-modules kamailio-python-modules kamailio-radius-modules kamailio-snmpstats-modules kamailio-tls-modules kamailio-unixodbc-modules kamailio-xml-modules kamailio-xmpp-modules The following NEW packages will be installed: kamailio 0 upgraded, 1 newly installed, 0 to remove and 42 not upgraded. Need to get 0 B/5304 kB of archives. After this operation, 24.5 MB of additional disk space will be used. Selecting previously unselected package kamailio. (Reading database ... 37888 files and directories currently installed.) Preparing to unpack .../kamailio_5.0.0+0~20170322012605.24+jessie_amd64.deb ... Unpacking kamailio (5.0.0+0~20170322012605.24+jessie) ... Processing triggers for systemd (215-17+deb8u4) ... Processing triggers for man-db (2.7.0.2-5) ... Setting up kamailio (5.0.0+0~20170322012605.24+jessie) ... Job for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. invoke-rc.d: initscript kamailio, action "start" failed. dpkg: error processing package kamailio (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for systemd (215-17+deb8u4) ... Errors were encountered while processing: kamailio [master 5dad0af] committing changes in /etc after apt run 16 files changed, 2992 insertions(+) create mode 100644 default/kamailio create mode 100755 init.d/kamailio create mode 100644 kamailio/dictionary.kamailio create mode 100644 kamailio/kamailio-advanced.cfg create mode 100644 kamailio/kamailio-basic.cfg create mode 100644 kamailio/kamailio.cfg create mode 100644 kamailio/kamctlrc create mode 120000 rc0.d/K02kamailio create mode 120000 rc1.d/K02kamailio create mode 120000 rc2.d/S03kamailio create mode 120000 rc3.d/S03kamailio create mode 120000 rc4.d/S03kamailio create mode 120000 rc5.d/S03kamailio create mode 120000 rc6.d/K02kamailio create mode 120000 systemd/system/multi-user.target.wants/kamailio.service Counting objects: 1321, done. Compressing objects: 100% (749/749), done. Writing objects: 100% (1321/1321), done. Total 1321 (delta 127), reused 1320 (delta 126) E: Sub-process /usr/bin/dpkg returned an error code (1) Checking up: root@homer-dev ~/homer-docker# systemctl status kamailio.service * kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled) Active: failed (Result: exit-code) since Mon 2017-03-27 16:58:09 UTC; 7s ago Mar 27 16:58:09 homer-dev kamailio[9438]: ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command t_precheck_trans (params 0) Mar 27 16:58:09 homer-dev kamailio[9438]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 479,...oadmodule? Mar 27 16:58:09 homer-dev kamailio[9438]: ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command t_check_trans (params 0) Mar 27 16:58:09 homer-dev kamailio[9438]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 480,...oadmodule? Mar 27 16:58:09 homer-dev kamailio[9438]: ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command t_check_trans (params 0) Mar 27 16:58:09 homer-dev kamailio[9438]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 483,...oadmodule? Mar 27 16:58:09 homer-dev systemd[1]: kamailio.service: control process exited, code=exited status=255 Mar 27 16:58:09 homer-dev systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Mar 27 16:58:09 homer-dev systemd[1]: Unit kamailio.service entered failed state. Mar 27 16:58:09 homer-dev kamailio[9438]: ERROR: bad config file (46 errors) Hint: Some lines were ellipsized, use -l to show in full. All modules are there, the problem is the startup script can't find where the modules are installed? If i just un-comment the "mpath" in the config file, kamailio starts normally. /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" I'm not 100% sure "mpath" was set before, but i don't remember having to set that up in the near past... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ [View Less]

2 2

Kamailio Registration load Issue
by Jade SZ 28 Mar '17

28 Mar '17

Hi Guys, I am running a simple REGISTER load test on: 1) Kamailio sever with 2 cores - mem 5G 2) Kamailio server with 4 cores - mem 16G Both are EC2 instances. At -r = 500 i.e. 500 reg/sec sipP test works fine with very few re-trans. But when i increase it to 800 reg/sec it starts retransmissions in bulk. I don't see server's CPU, load-avg or memory shooting. Running everything by default, even using kamctl to start the instance. So I have not tuned any params yet. My main concern is how … [View More]

4 6

kamailio 4.4.1 crash on bad sip response
by David Escartín Almudévar 28 Mar '17

28 Mar '17

hello all we have experienced a crash and tracing the logs and the core, seems it was because a sip response from an endpoint. a UDP receiver (26248) crashed and the last message we see on it is a 487 quite bad formed Mar 24 01:58:02 mia-proxy-inout-1-stby /usr/local/kamailio/sbin/kamailio[26248]: ERROR: tm [t_lookup.c:1055]: t_check_msg(): ERROR: reply doesn't have a via, cseq or call-id header Mar 24 01:58:17 mia-proxy-inout-1-stby /usr/local/kamailio/sbin/kamailio[26230]: ALERT: <core&… [View More]gt; [main.c:739]: handle_sigs(): child process 26248 exited by a signal 11 the backtrace of the core (gdb) backtrace #0 0x0000000000457ab9 in get_hash1_case_raw (s=0x0, len=0) at hashes.h:210 #1 0x000000000045b472 in _dns_hash_find (name=0x7f6906943188, type=1, h=0x7fff120793cc, err=0x7fff120793ac) at dns_cache.c:535 #2 0x0000000000461285 in dns_hash_get (name=0x7f6906943188, type=1, h=0x7fff120793cc, err=0x7fff120793ac) at dns_cache.c:762 #3 0x0000000000467194 in dns_get_entry (name=0x7f6906943188, type=1) at dns_cache.c:2102 #4 0x0000000000468a05 in dns_a_get_he (name=0x7f6906943188) at dns_cache.c:2432 #5 0x0000000000468bb9 in dns_get_he (name=0x7f6906943188, flags=1) at dns_cache.c:2505 #6 0x00000000004696c4 in dns_srv_sip_resolvehost (name=0x7f6906943188, port=0x7fff120795e2, proto=0x7fff120795e1 "\001\330\023") at dns_cache.c:2679 #7 0x000000000046aa37 in dns_sip_resolvehost (name=0x7f6906943188, port=0x7fff120795e2, proto=0x7fff120795e1 "\001\330\023") at dns_cache.c:2849 #8 0x000000000049519e in update_sock_struct_from_via (to=0x7fff12079708, msg=0x7f69069a1dd8, via=0x7f69068f82a8) at forward.c:704 #9 0x0000000000495ee5 in do_forward_reply (msg=0x7f69069a1dd8, mode=0) at forward.c:766 #10 0x00000000004970af in forward_reply (msg=0x7f69069a1dd8) at forward.c:849 #11 0x00000000005197ef in receive_msg ( buf=0xab0d80 "SIP/2.0 487 Request Terminated\r\nFrom: \"8888888888 \"<sip:8888888888@7.7.7.7;user=phone>;tag=B7jgc8jQ4m5pB\r\nTo: <sip:555555555@8.8.8.8:5060>;tag=e0d50be-13c4-58d47cba-a2ed9808-36fa\r \nl\337K\016"..., len=367, rcv_info=0x7fff12079a10) at receive.c:299 #12 0x0000000000627b43 in udp_rcv_loop () at udp_server.c:495 #13 0x00000000004b107a in main_loop () at main.c:1600 #14 0x00000000004b842f in main (argc=13, argv=0x7fff12079fb8) at main.c:2616 i have tried to duplicate the issue, but i dont know how to translate l \337K\016 to a xml notation i guess this is some weird that cannot be processed for kamailio could you please take a look and let me know if you know how to duplicate and fix this crash? thanks a lot and regards david [View Less]

2 9

new keepalive module ?
by Guillaume Bour 28 Mar '17

28 Mar '17

Hi all, I would like to implement destinations keepalive in drouting module (as it is done in dispatcher). But instead of duplicating what's implemented in dispatcher, I think it would be more clever to create a new module dedicated to pinging destinations, and to plug drouting, dispatcher and other modules to this new one What's you opinion about that ? Cheers, Guillaume

2 5

Presence + RLS + XCAP
by Paul Smith 27 Mar '17

27 Mar '17

Hi I am struggling to figure out how to build and test an integrated presence server with Resource-List (rls.so) and XCAP (xcap_server.so). I am confused about what is meant to happen and how to debug it. Not sure if I am suffering from incompatibilities, configuration errors, or bugs… I have not found an up to date guide on configuring an integrated presence, rls, xcap server so I have used old tutorials, module documentation and some course code review … but I’m pretty confused now! I am … [View More]using Snom phones and Jitsi (on Mac) to test. I have a kamailio Registrar which proxies all presence messages to the separate kamailio 4.2 Presence server. REQUIREMENT : Contact Presence ============================= I would like to get a SUBSCRIBE from the Snom phone to subscribe to a contact-list resource list, and to receive a NOTIFY with all of the presence states for those contacts. Relevant snom settings? Contact List URI: this generates a SUBSCRIBE with event:presence I believe that: ========== presence module should be managing the individual user presence and dialog states xcap module should allow me to use HTTP / curl to edit the resource lists rls module should be managing the mapping from a resource list subscription to the individual presence states and generating the aggregated NOTIFY messages. each device should then be able to SUBSCRIBE to a resource list uri and then receive an aggregated NOTIFY for all of the watched entities. Current status: =========== SUBSCRIBE sip:org1a-contacts@*****.co.uk SIP/2.0 v: SIP/2.0/UDP 192.168.1.15:59594;branch=z9hG4bK-qllzlgs3xf43;rport f: <sip:org1a@*****.co.uk>;tag=wwasqc6lmb t: <sip:org1a-contacts@****.co.uk> i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 1 SUBSCRIBE Max-Forwards: 70 User-Agent: snomD375/8.9.3.46 m: <sip:org1a@192.168.1.15:59594>;reg-id=1 Event: presence Accept: application/pidf+xml, application/rlmi+xml, multipart/related Supported: eventlist Expires: 3600 l: 0 … auth handshake SIP/2.0 202 OK Record-Route: <sip:10.210.160.13;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes> Record-Route: <sip:**pubip**;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes> v: SIP/2.0/UDP 192.168.1.15:59594;received=**recvip**;branch=z9hG4bK-8wufiv1ua3ne;rport=60554 f: <sip:org1a@****.co.uk>;tag=wwasqc6lmb t: <sip:org1a-contacts@*****.co.uk>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-d30e i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 2 SUBSCRIBE Expires: 3600 Contact: <sip:10.210.160.16:5060;transport=udp> Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 … and then the snom receives an empty NOTIFY. NOTIFY sip:org1a@192.168.1.15:58325 SIP/2.0 Via: SIP/2.0/UDP 109.104.108.18;branch=z9hG4bKc54a.20db45097fb44595ccd5a30827ca2a62.0 Via: SIP/2.0/UDP 10.210.160.16;rport=5060;branch=z9hG4bKc54a.40150886000000000000000000000000.0 To: <sip:org1a@****.co.uk>;tag=sx4ne9jpcl From: <sip:org1a-contacts@*****.co.uk>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-ed89 CSeq: 2 NOTIFY Call-ID: 313439303237323938323235353133-2du221e6okvf Content-Length: 0 User-Agent: kamailio (4.2.0 (x86_64/linux)) Max-Forwards: 69 Event: presence Contact: <sip:10.210.160.16:5060;transport=udp;alias=10.210.160.16~5060~1> Subscription-State: active;expires=3600 The Subscription is showing up in the watchers table, not the rls_watchers table, so I believe it is being rejected by the rls_handle_subscribe() and then parsed by the presence module. Is that correct? There is noting in rls_watchers or rls_presentity tables. Should there be? I have a rows in the XCAP table for the Resource-List curl -d @/tmp/contacts-resource-list.xml -vvv -X PUT http://109.104.108.23:5060/xcap-root/resource-lists/users/sip:org1a-contact… relevant row in xcap table: | 3 | org1a-contacts | *****.co.uk | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="team"><entry uri="sip:1002@****.co.uk"/><entry uri="sip:1001@****.co.uk"><display-name>bob</display-name></entry><entry uri="sip:1003@****.co.uk"><display-name>test1</display-name></entry><entry uri="sip:1006@****co.uk"><display-name>test1006</display-name></entry><entry uri="sip:1007@****co.uk"><display-name>test1007</display-name></entry><entry uri="sip:1008@h****.co.uk"><display-name>test1008</display-name></entry><entry uri="sip:1009@****.co.uk"><display-name>test1009</display-name></entry> <entry uri="sip:1005@****.co.uk"><display-name>testing1005</display-name></entry> <entry uri=“sip:org1a@****.co.uk"><display-name>testingorg1a</display-name></entry> <entry uri=“sip:org1c@****.co.uk"><display-name>testingorg1c</display-name></entry></list></resource-lists> | 4 | sr-1490270411-10965-2 | 0 | /xcap-root/resource-lists/users/sip:org1a-contacts@****co.uk/index | 0 | I have rows in the presence table for the entities that I want to track e.g. | | | 2889 | org1c | ****.co.uk | presence | a.1490270411.10942.42.1083 | 1490277099 | 1490273499 | <?xml version="1.0 " encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:im="urn:ietf:params:xml:ns:pidf:im" entity=“pres:org1c@*****.co.uk"> <tuple id="snom370-000413260935"> <status><basic>open</basic> <im:im>Available</im:im> </status> <contact priority="1.00">sip:org1c@****.co.uk</contact> <note xml:lang="en">Available</note> </tuple></presence> I am running kamaiio 4.2 Question : Should I have an resource-services row too? Or does rls.so infer the relationship using the URI? When I use jitsi to manipulate its contact list it only inserts 2 rows into the xcap table and I cannot get a SIP Subscribe to connect to that resource list either. Question : What should the curl PUT commands and XML docs look like to build a valid RLS service (is there a good example anywhere)? Question : Is there an example config or doc anywhere showing how to build integrated presence, rls, xcap? Question : Has anyone got this working with Snom and or other handsets… the standards and implementations seem vague / inconsistent? Note: There is another Snom setting (Extension Monitoring Call Pickup List URI:) which produces a SUBSCRIBE with Event:dialog but which should allow a button/lamp/callpickup/BLF for the buddy contact. I’d like to get that working too, but recognise that rls.so does not handle dialog messages by default, so I’m starting with Presence... Paul [View Less]

3 2

Important problem with this test : if ($var(i:maxRangeInx)==$null)
by Ginhoux, Patrick 27 Mar '17

27 Mar '17

Hi, I'm continue my job to 'migrate' my 3.3.1 kamailio.cfg script on a RHEL 6.2 server to the 5.0.x kamailio version on a RHEL 7.1 server, and I encounter a major problem with the '== $null' test in the script. My original script (3.3.1) contains this sequence : $var(i:rangeInx)=0; $var(i:maxRangeInx) = $sht(mbxrangeHash=>"maxrange"); xlog(" imbxRangeInx value before test : '$var(i:maxRangeInx)' "); if ($var(i:maxRangeInx)==$null) { $var(i:… [View More]

2 2

Module cnxcc
by m9rt8ii 27 Mar '17

27 Mar '17

Please can anybody give me basic steps how to set up cnxcc module on kamailio 4.4 ? I will be very very grateful. http://kamailio.org/docs/modules/4.4.x/modules/cnxcc.html Thank you in advance -- View this message in context: http://sip-router.1086192.n5.nabble.com/Module-cnxcc-tp157107.html Sent from the Users mailing list archive at Nabble.com.

2 1

Cannot yum download from download.opensuse.org, no mirrors
by Shadowy Man 27 Mar '17

27 Mar '17

This is only happening for one particular block of IP's. I can download from other blocks. I suspect it is being done on the download.opensuse.org end for some reason. It appears to be the entire xx.xx block of which I only use a few addresses from. I will probably have to try contact them about it. What I would like to ask is why Kamailio does not have any mirrors? If there were then this would not be a problem. It would just failover.

2 1

wiresharking TLS problem
by Rex Lin (林昱頡) 27 Mar '17

27 Mar '17

hi all wondering if any of you has the experience of TLS decrypting, SIP client has problem while calling each other, but we are using TLS for communication protocol.....facing problem debugging it..... Best Regards, Rex Lin

3 3

Regex module having trouble with large regex expressions in kamailio 4.4.5
by Tim Balmer 24 Mar '17

24 Mar '17

Hi, I discovered some strange behavior from the regex module yesterday (kamailio version 4.4.5). For example we have the phone number 0033755500381 that we try to match with the following kamailio code: #!define REGEX_FF_NEG_INTERNATIONAL "6" ... if (pcre_match_group("$rU", REGEX_FF_NEG_INTERNATIONAL)) { # Do something } When the content of our regex groups file is like this: [6] ^(\+|00)3(0690|0693|0694|0695|0697|0698|0699|36|37|4160|46|47|51160|511691|511693|511696|511891|… [View More]

2 2

Re: [SR-Users] Question on the debug change level command with 5.0.x
by Federico Cabiddu 24 Mar '17

24 Mar '17

No, you don't. The debugger module, among other things, allows to set the debug level per module. Regards, Federico On 23 Mar 2017 6:31 pm, "Ginhoux, Patrick" <patrick.ginhoux(a)fr.unisys.com> wrote: Hi, Thanks you so much; it works fine as expected. Now have I to keep the ‘debugger’ module lines in my config ? Cordialement Patrick GINHOUX *De :* sr-users [mailto:sr-users-bounces@lists.sip-router.org] *De la part de* Federico Cabiddu *Envoyé :* jeudi 23 mars 2017 17:38 … [View More]*À :* Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> *Objet :* Re: [SR-Users] Question on the debug change level command with 5.0.x Hi, you have to load the cfg_rpc module (http://www.kamailio.org/docs/ modules/5.0.x/modules/cfg_rpc.html). Then you can run kamcmd cfg.set_now_int core debug <LEVEL>. Regards, Federico On Thu, Mar 23, 2017 at 5:24 PM, Ginhoux, Patrick < patrick.ginhoux(a)fr.unisys.com> wrote: Hi, Previously, in my kamailio.cfg 3.3.1, the debug level is set using the ‘debug’ option. Then with the MI_FIFO module it was simple to change debug level on the flight using the simple command ‘kamctl fifo debug <new value>’. Moving to kamailio 5.0.x, there is no more MI_FIFO module. In the kamailio.cfg, I still have the ‘debug’ option set; so I’m looking for the command to use to change the debug level. I look at the debugger module that seems the solution. So I add the following lines in y script: debug=4 loadmodule "debugger.so" modparam("debugger", "log_level", 1) modparam("debugger", "mod_hash_size", 5) modparam("debugger", "mod_level_mode", 1) Then I try to use the following commands : This one retrieves the debug level set by the debug=4) : [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level core 4 This one changed the debug level for the core module [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level core 2 200 ok Same commands can be used for each of the module loaded: [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level tm 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level tm 2 200 ok [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level dispatcher 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level dispatcher 2 200 ok Each command only applies to a specific module. I don’t find a ‘global’ command that I could use in the same manner as the old ‘kamctl fifo debug’ command. Is there something I missed in my config ? Cordialement Patrick GINHOUX _______________________________________________ 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 _______________________________________________ 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 [View Less]

2 1

Question on the debug change level command with 5.0.x
by Ginhoux, Patrick 23 Mar '17

23 Mar '17

Hi, Previously, in my kamailio.cfg 3.3.1, the debug level is set using the 'debug' option. Then with the MI_FIFO module it was simple to change debug level on the flight using the simple command 'kamctl fifo debug <new value>'. Moving to kamailio 5.0.x, there is no more MI_FIFO module. In the kamailio.cfg, I still have the 'debug' option set; so I'm looking for the command to use to change the debug level. I look at the debugger module that seems the solution. So I add the following … [View More]

2 2

Presence + RLS + XCAP
by Paul Smith 23 Mar '17

23 Mar '17

Longer version of this got blocked by the mail list earlier today… and I have dug a bit deeper since then… I am trying to SUBSCRIBE to a resource-list (rls.so) but the rls_handle_subscribe() is returning with a “list not found" Mar 23 16:36:11 presence kamailio[12536]: DEBUG: rls [subscribe.c:211]: rls_get_service_list(): No rl document found Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:81]: db_free_columns(): freeing 2 columns Mar 23 16:36:11 presence kamailio[… [View More]12536]: DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[0] at 0x7f70b50ebee8 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[1] at 0x7f70b50afb90 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:94]: db_free_columns(): freeing result names at 0x7f70b50eaf98 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:99]: db_free_columns(): freeing result types at 0x7f70b50cc0a0 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:136]: db_free_result(): freeing result set at 0x7f70b50cc2d0 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: rls [subscribe.c:634]: rls_handle_subscribe(): list not found - searched for uri <sip:org1a-contacts@**MYDOMAIN**> I have tried various permutations of xml docs in the xcap table for the rls-service and resource-list but nothing have not yet managed to get a success from the rls_handle_subscribe() Latest version of data in my mysql database / xcap table: | 3 | org1a-contacts | **MYDOMAIN** | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="team"><entry uri="sip:1002@**MYDOMAIN**"/><entry uri="sip:1001@**MYDOMAIN**"><display-name>bob</display-name></entry><entry uri="sip:1003@**MYDOMAIN**"><display-name>test1</display-name></entry><entry uri="sip:1006@**MYDOMAIN**"><display-name>test1006</display-name></entry><entry uri="sip:1007@**MYDOMAIN**"><display-name>test1007</display-name></entry><entry uri="sip:1008@**MYDOMAIN**"><display-name>test1008</display-name></entry><entry uri="sip:1009@**MYDOMAIN**"><display-name>test1009</display-name></entry> <entry uri="sip:1005@**MYDOMAIN**"><display-name>testing1005</display-name></entry> <entry uri="sip:org1a@**MYDOMAIN**"><display-name>testingorg1a</display-name></entry> <entry uri="sip:org1c@**MYDOMAIN**"><display-name>testingorg1c</display-name></entry> <entry uri="sip:dnd1002-6@presence.byphone.co.uk"><display-name>1002 dnd</display-name></entry></list></resource-lists> | 4 | sr-1490270411-10960-2 | 0 | /xcap-root/resource-lists/users/sip:org1a-contacts@**MYDOMAIN**/index | 0 | | 4 | org1a-contacts | **MYDOMAIN** | <?xml version="1.0" encoding="UTF-8" standalone="no"?><rls-services xmlns="urn:ietf:params:xml:ns:rls-services"><service uri="sip:org1a-contacts@**MYDOMAIN**"><list name="org1a-contacts"><rl:entry uri="1000_6@**MYDOMAIN**"/><rl:entry uri="org1c@**MYDOMAIN**"/></list><packages><package>presence</package></packages></service></rls-services> | 8 | sr-1490285288-12551-1 | 0 | /xcap-root/rls-services/users/sip:org1a-contacts@**MYDOMAIN**/index | 0 | Is there a reference doc somewhere with an example of a successful SUBSCRIBE and associated XCAP docs for a resource-list? Thanks Paul [View Less]

1 0

failed to find append_branch
by Anuradha Selvarajan 23 Mar '17

23 Mar '17

Hello, I am a new user to kamailio. I am trying to use the fork.cfg to send sip registration to 2 destination. I get the following error but I have added loadmodule tm.so, why am I still getting this error? Mar 22 15:53:48 emps116 kamailio[14065]: ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command append_branch (params 1) Mar 22 15:53:48 emps116 kamailio[14065]: : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/fork.cfg, line 45, … [View More]

3 3

Presence + RLS + XCAP
by Paul Smith 23 Mar '17

23 Mar '17

Hi I am struggling to figure out how to build and test an integrated presence server with Resource-List (rls.so) and XCAP (xcap_server.so). I am confused about what is meant to happen and how to debug it. Not sure if I am suffering from incompatibilities, configuration errors, or bugs… I have not found an up to date guide on configuring an integrated presence, rls, xcap server so I have used old tutorials, module documentation and some course code review … but I’m pretty confused now! I am … [View More]using Snom phones and Jitsi (on Mac) to test. I have a kamailio Registrar which proxies all presence messages to the separate kamailio 4.2 Presence server. REQUIREMENT : Contact Presence ============================= I would like to get a SUBSCRIBE from the Snom phone to subscribe to a contact-list resource list, and to receive a NOTIFY with all of the presence states for those contacts. Relevant snom settings? Contact List URI: this generates a SUBSCRIBE with event:presence I believe that: ========== presence module should be managing the individual user presence and dialog states xcap module should allow me to use HTTP / curl to edit the resource lists rls module should be managing the mapping from a resource list subscription to the individual presence states and generating the aggregated NOTIFY messages. each device should then be able to SUBSCRIBE to a resource list uri and then receive an aggregated NOTIFY for all of the watched entities. Current status: =========== SUBSCRIBE sip:org1a-contacts@*****.co.uk <sip:org1a-contacts@*****.co.uk> SIP/2.0 v: SIP/2.0/UDP 192.168.1.15:59594;branch=z9hG4bK-qllzlgs3xf43;rport f: <sip:org1a@*****.co.uk <sip:org1a@*****.co.uk>>;tag=wwasqc6lmb t: <sip:org1a-contacts@****.co.uk <sip:org1a-contacts@****.co.uk>> i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 1 SUBSCRIBE Max-Forwards: 70 User-Agent: snomD375/8.9.3.46 m: <sip:org1a@192.168.1.15:59594 <sip:org1a@192.168.1.15:59594>>;reg-id=1 Event: presence Accept: application/pidf+xml, application/rlmi+xml, multipart/related Supported: eventlist Expires: 3600 l: 0 … auth handshake SIP/2.0 202 OK Record-Route: <sip:10.210.160.13;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes <sip:10.210.160.13;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes>> Record-Route: <sip:**pubip**;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes <sip:**pubip**;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes>> v: SIP/2.0/UDP 192.168.1.15:59594;received=**recvip**;branch=z9hG4bK-8wufiv1ua3ne;rport=60554 f: <sip:org1a@****.co.uk <sip:org1a@****.co.uk>>;tag=wwasqc6lmb t: <sip:org1a-contacts@*****.co.uk <sip:org1a-contacts@*****.co.uk>>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-d30e i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 2 SUBSCRIBE Expires: 3600 Contact: <sip:10.210.160.16:5060;transport=udp <sip:10.210.160.16:5060;transport=udp>> Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 … and then the snom receives an empty NOTIFY. NOTIFY sip:org1a@192.168.1.15:58325 <sip:org1a@192.168.1.15:58325> SIP/2.0 Via: SIP/2.0/UDP 109.104.108.18;branch=z9hG4bKc54a.20db45097fb44595ccd5a30827ca2a62.0 Via: SIP/2.0/UDP 10.210.160.16;rport=5060;branch=z9hG4bKc54a.40150886000000000000000000000000.0 To: <sip:org1a@****.co.uk <sip:org1a@****.co.uk>>;tag=sx4ne9jpcl From: <sip:org1a-contacts@*****.co.uk <sip:org1a-contacts@*****.co.uk>>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-ed89 CSeq: 2 NOTIFY Call-ID: 313439303237323938323235353133-2du221e6okvf Content-Length: 0 User-Agent: kamailio (4.2.0 (x86_64/linux)) Max-Forwards: 69 Event: presence Contact: <sip:10.210.160.16:5060;transport=udp;alias=10.210.160.16~5060~1 <sip:10.210.160.16:5060;transport=udp;alias=10.210.160.16~5060~1>> Subscription-State: active;expires=3600 The Subscription is showing up in the watchers table, not the rls_watchers table, so I believe it is being rejected by the rls_handle_subscribe() and then parsed by the presence module. Is that correct? There is noting in rls_watchers or rls_presentity tables. Should there be? I have a rows in the XCAP table for the Resource-List curl -d @/tmp/contacts-resource-list.xml -vvv -X PUT http://109.104.108.23:5060/xcap-root/resource-lists/users/sip:org1a-contact… <http://109.104.108.23:5060/xcap-root/resource-lists/users/sip:org1a-contact…> relevant row in xcap table: | 3 | org1a-contacts | *****.co.uk <http://co.uk/> | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="team"><entry uri="sip:1002@****.co.uk <http://co.uk/>"/><entry uri="sip:1001@****.co.uk <http://co.uk/>"><display-name>bob</display-name></entry><entry uri="sip:1003@****.co.uk <http://co.uk/>"><display-name>test1</display-name></entry><entry uri="sip:1006@****co.uk <http://co.uk/>"><display-name>test1006</display-name></entry><entry uri="sip:1007@****co.uk <http://co.uk/>"><display-name>test1007</display-name></entry><entry uri="sip:1008@h****.co.uk <http://co.uk/>"><display-name>test1008</display-name></entry><entry uri="sip:1009@****.co.uk <http://co.uk/>"><display-name>test1009</display-name></entry> <entry uri="sip:1005@****.co.uk <http://co.uk/>"><display-name>testing1005</display-name></entry> <entry uri=“sip:org1a@****.co.uk <http://co.uk/>"><display-name>testingorg1a</display-name></entry> <entry uri=“sip:org1c@****.co.uk <http://co.uk/>"><display-name>testingorg1c</display-name></entry></list></resource-lists> | 4 | sr-1490270411-10965-2 | 0 | /xcap-root/resource-lists/users/sip:org1a-contacts@****co.uk/index <http://co.uk/index> | 0 | I have rows in the presence table for the entities that I want to track e.g. | | | 2889 | org1c | ****.co.uk <http://co.uk/> | presence | a.1490270411.10942.42.1083 | 1490277099 | 1490273499 | <?xml version="1.0 " encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:im="urn:ietf:params:xml:ns:pidf:im" entity=“pres:org1c@*****.co.uk <http://co.uk/>"> <tuple id="snom370-000413260935"> <status><basic>open</basic> <im:im <im:im>>Available</im:im> </status> <contact priority="1.00">sip:org1c@****.co.uk <http://co.uk/></contact> <note xml:lang="en">Available</note> </tuple></presence> I am running kamaiio 4.2 Question : Should I have an resource-services row too? Or does rls.so infer the relationship using the URI? When I use jitsi to manipulate its contact list it only inserts 2 rows into the xcap table and I cannot get a SIP Subscribe to connect to that resource list either. Question : What should the curl PUT commands and XML docs look like to build a valid RLS service (is there a good example anywhere)? Question : Is there an example config or doc anywhere showing how to build integrated presence, rls, xcap? Question : Has anyone got this working with Snom and or other handsets… the standards and implementations seem vague / inconsistent? Note: There is another Snom setting (Extension Monitoring Call Pickup List URI:) which produces a SUBSCRIBE with Event:dialog but which should allow a button/lamp/callpickup/BLF for the buddy contact. I’d like to get that working too, but recognise that rls.so does not handle dialog messages by default, so I’m starting with Presence... Paul Paul Smith 1 Lanyon Quay, Belfast, BT1 3LG UK: 02890270050 RoI: 1800855111 paul.smith(a)claritytele.com <mailto:paul.smith@claritytele.com%0b> www.claritytele.com <http://claritytele.com/> This email and any attachment, is confidential. If you have received this in error, please delete it from your system, do not use or disclose the information in any way, and notify me immediately. The contents of this message may contain personal views which are not the views of Clarity Telecom, unless specifically stated. > >> >> [View Less]

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Status of db_cassandra
by Markus Bönke 23 Mar '17

23 Mar '17

Hello, We are thinking about to use kamailio as a sip registrar with cassandra as db backend. As I can see in the documentation for kamailio 5.0 it is only tested with Casandra 1.1.6 and 1.0.1, current version is 3.10 in the meantime. Is anyone using this module with newer versions of Cassandra? Is it stable? Thanks and regards Markus

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looks like dp_translate with pvs in repl_exp is badly broken
by Juha Heinanen 23 Mar '17

23 Mar '17

when match_exp of dp 100 is ^#test and repl_exp is sip:$rU@$rd this code: xlog("L_INFO", "****** ru = $ru\n"); dp_translate(100, "$rU/$var(res)"); xlog("L_INFO", "****** res = $var(res)\n"); produces: Mar 22 08:21:17 lohi /usr/bin/sip-proxy[7543]: INFO: ****** ru = sip:#test@test.tutpro.com Mar 22 08:21:17 lohi /usr/bin/sip-proxy[7543]: INFO: ****** res = sip:$rU@$rd -- juha

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getting source IP from Contact header
by Ali Taher 22 Mar '17

22 Mar '17

Hello everyone, I'm trying to get the host part from the contact header in the sip invite packet. Following is the contact header format: Contact: <sip:user_name@192.168.1.1:5060> What I want as result is 192.168.1.1 I searched among the pseudovairables and found $ct variable which reference to the body of contact header. (<sip:user_name@192.168.1.1:5060>) Is there a way to get directly the host (192.168.1.1) without applying transformation at the level of $ct. … [View More]

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Alternative of SIPp for WebRTC2SIP load-test
by Jade SZ 21 Mar '17

21 Mar '17

Hi Guys, SIPp is great in testing pure SIP udp/tcp/tls but I was wondering if there is a similar tool for stress testing Kamailio websocket2sip. I am trying to load-test webrtc2sip kamailio's implementation closest to the real scenario. One way would be to fork multiple process of webrtc2sip client in loop. But would really like to hear how are kamailio webrtc users/community load-testing it for benchmarking their servers. Also roughly how much CPS they have achieved using single kamailio … [View More]

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do we need high available IP for S-CSCF?
by Valentin Christoph 21 Mar '17

21 Mar '17

Hi all, Maybe I need a small hint. I have got a configuration of 2 S-CSCF instances (both kamailio 4.4.5 with S-CSCF configuration, each on a separate virtual machine) in load sharing mode (50:50). Both instances work on the same "scscf" data base, so that it is only one S-CSCF from a logical point of view. Now I get into troubles with registration: * P-CSCF sends REGISTER to first S-CSCF * First S-CSCF answers with 401 Unauthorized an challenges the UE. So far everything OK. *… [View More]

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Siremis 4.3.0 with Apache 2.4
by kai.ohnacker＠cbc.de 21 Mar '17

21 Mar '17

Hello Community, I installed the Kamailio 4.4 and try to install the Siremis GUI based on apache 2.4. If I try to start the Siremis Wizard, the Webserver display the following line: "location.href='$url'"; exit; } ?>". The starting page of apache 2.4 is working fine, but the /siremis page doesn't. Due the installation I take the dedicated apache 2.4 commands. Are there some further requirements? Some ideas? OS: Ubuntu 16.04 Thanks a lot. Cheers, Kai

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No cdr after Kamailio 5.0 restart
by Kordován Szabolcs 21 Mar '17

21 Mar '17

Hi, I use acc module for create cdr into db and also make acc log into db. After restart Kamailio cdr won't be created but acc log will be inserted into db. In the syslog also I see just the acc log. The dialog will be remove from db. I have read many threads. I know that it should work. Thanks for help! Regards, Szabolcs >From my configs: #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 modparam("acc", "db_url", DBURL) modparam("acc", "db_flag", FLT_ACC) modparam("… [View More]acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") modparam("acc", "log_level", 3) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "early_media", 1) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "acc_prepare_always", 1) modparam("acc", "time_mode", 1) modparam("acc", "time_attr", "seconds") modparam("acc", "time_exten", "microsecs") modparam("acc", "reason_from_hf", 1) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdr_extra", "callid=$ci;src_user=$dlg_var(fU);src_domain=$dlg_var(fd);dst_ouser=$dlg_var(tU);dst_user=$dlg_var(rU);dst_domain=$dlg_var(rd);src_ip=$dlg_var(si);peer_in=$dlg_var(peerin);peer_out=$dlg_var(peerout)") request_route { <------> <------>$var(loose) = loose_route(); <------>$var(check_trans) = t_check_trans(); <------>sip_trace(); <------>setflag(FLT_SIPTRACE); <------>dlg_manage(); <------># handle retransmissions <------>if(t_precheck_trans()) { <------> xlogl("LOG_LOCAL0", "L_ALERT", "================== kamaty $rm route $ci ==== $rm ==== precheck_trans exit\n"); <------> exit; <------>} <------>if(is_method("OPTIONS")) {. <------> # send reply for each options request. <------> sl_send_reply("200", "ok");. <------> exit();. <------>} <------># CANCEL processing <------>if (is_method("CANCEL")) { <------><------>if (t_check_trans()) { <------><------> route(RELAY); <------><------>} <------><------>exit; <------>} <------>if ( is_method("ACK") ) { <------><------>if ( $var(check_trans) ) { <------><------> route(RELAY); <------><------>} <------><------>exit; <------>} <------>if (is_method("BYE")) { <------> if ($var(loose)) { <------><------>setflag(FLT_ACC); # do accounting ... <------><------>setflag(FLT_ACCFAILED); # ... even if the transaction fails <------><------>setflag(FLT_ACCMISSED); <------><------>route(RELAY); <------> <-->exit; <------> } <------> exit; <------>} etc. [View Less]

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Re: [SR-Users] ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 21 Mar '17

21 Mar '17

Hi, There was a mistake in the file attached ; the short kamailio.cfg file (that makes kamailio to start) was not the correct one; attached the correct one. Have anyone an idea of this problem ? Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : lundi 20 mars 2017 17:58 À : Kamailio users (sr-users(a)lists.sip-router.org) <sr-users(a)lists.sip-router.org> Objet : RE: ht_db_load_table(): key type must be string (type=6) Hi, To figure out what causes this issue, I work to … [View More]shorten my kamailio.cfg to contains loadmodule/modparam instructions for the htable module only. My first attempt (see kamailio.cfg.ok) shows that the problem doesn't occur. In the second attempt (see kamailio.cfg.ko) I add instructions previously removed from the original .cfg file. I have determined that, depending on the instructions that follows the "modparam("htable", "fetch_rows", 1000)" line, the "key type must be string (type=6)" error occurs or not. In this ko file, if I comment out the following sequence, the error occurs : modparam("htable", "fetch_rows", 1000) #!!ifdef FIXED_ROUTING #modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") #!!else #modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") #modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") #!!endif modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") I hope my explanation is clearn and there are people who can help me on this problem that causes kamailio to not start. Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : vendredi 17 mars 2017 15:46 À : Kamailio users (sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>) <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Objet : ht_db_load_table(): key type must be string (type=6) Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX [View Less]

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Re: [SR-Users] ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 20 Mar '17

20 Mar '17

Hi, To figure out what causes this issue, I work to shorten my kamailio.cfg to contains loadmodule/modparam instructions for the htable module only. My first attempt (see kamailio.cfg.ok) shows that the problem doesn't occur. In the second attempt (see kamailio.cfg.ko) I add instructions previously removed from the original .cfg file. I have determined that, depending on the instructions that follows the "modparam("htable", "fetch_rows", 1000)" line, the "key type must be string (type=6)" … [View More]error occurs or not. In this ko file, if I comment out the following sequence, the error occurs : modparam("htable", "fetch_rows", 1000) #!!ifdef FIXED_ROUTING #modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") #!!else #modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") #modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") #!!endif modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") I hope my explanation is clearn and there are people who can help me on this problem that causes kamailio to not start. Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : vendredi 17 mars 2017 15:46 À : Kamailio users (sr-users(a)lists.sip-router.org) <sr-users(a)lists.sip-router.org> Objet : ht_db_load_table(): key type must be string (type=6) Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX [View Less]

1 0

ACC - Cdr enabled
by Diego Nadares 20 Mar '17

20 Mar '17

Hi guys, I have configured to use cdr with extra fields modparam("acc", "cdr_extra", * "sip_code=$rs;sip_reason=$rr;"* "hangup_src=$dlg_var(hangup_disposition);" "prefix=$dlg_var(prefix);gos=$dlg_var(gos);endpoint_type=$ dlg_var(endpoint_type);" "ip_a=$dlg_var(ip_a);ip_b=$dlg_var(ip_b);" Is this the right way to always save the hangup cause? Or is another better way? Because, when call is finished with a bye both fields are empty. The other thing is that I need … [View More]

2 5

Is File Sharing between two Extensions Possible?
by Shahid Mehmood 20 Mar '17

20 Mar '17

Dear all, I want to share (any kind of) files between two extensions. Is it possible? If it is, then how ??? please elaborate with example. -- Shahid Mehmood

1 0

cisco spa502g issues
by ben＠wtf.com.au 19 Mar '17

19 Mar '17

Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@<our domain> Contact:: sip:5546@<OUR OFFICE IP>:1032 Q= Expires:: 180 … [View More]

3 4

Generate extra cdr per dialog
by Diego Nadares 18 Mar '17

18 Mar '17

Hi Guys, I'm using acc module for accounting with the following conf: .... modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_facility", "LOG_LOCAL1") modparam("acc", "cdr_end_id", "end_time_dt") modparam("acc", "cdr_extra", "sip_code=$rs;sip_reason=$rr;" "hangup_src=$dlg_var(hangup_disposition);" "prefix=$dlg_var(prefix);gos=$dlg_var(gos);endpoint_type=$dlg_var(endpoint_type);" … [View More]

1 1

f_malloc WARNINGS in syslog
by Diego Nadares 17 Mar '17

17 Mar '17

Hi Guys, I'm seeing a lot of warnings in my syslog. Is this normal? Or maybe I am doing something wrong in my cfg? .... Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:444]: fm_malloc(): fm_malloc(0x7f68e5199010, 8) called from nathelper: nathelper.c: nh_timer(2078) Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:527]: fm_malloc(): fm_malloc(0x7f68e5199010, 8) returns address 0x7f68e52a2d50 Mar 16 17:11:47 dwsipm03 ./kamailio[… [View More]

2 2

ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 17 Mar '17

17 Mar '17

Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 … [View More]15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX [View Less]

1 0

Error on <ds_probing_threshhold> of type <2> not found in module <dispatcher> and SCTP API Not initialized
by Ginhoux, Patrick 17 Mar '17

17 Mar '17

Hi, I'm migrating my kamailio.cfg from the 3.3.1 release to 5.0.0 and encounter the following errors : Mar 16 12:56:45 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: Not starting : invalid configuration file! Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: 0(6652) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <ds_probing_threshhold> of type <2… [View More]

2 3

Configuration to fetch SIP response body.
by vivek＠advaitamtech.com 17 Mar '17

17 Mar '17

Hi, Is there any configuration to fetch SIP response body received?. My requirement is to fetch the SIP response body and save it into database. Thanks, Vivek

3 3

Search for kamailio PERL module for 5.0 level
by Ginhoux, Patrick 17 Mar '17

17 Mar '17

Hi, I'm a beginner in the use of kamailio product, so I try below to explain my problem with my words: 1) PERL module issue : Some years ago a SIP Router proxy running RedHat 6.2 was developped using the kamailio version 3.3 and the following packages used : kamailio-3.3.2-1.1.1.x86_64.rpm kamailio-snmpstats-3.3.2-1.1.1.x86_64.rpm kamailio-mysql-3.3.2-1.1.1.x86_64.rpm kamailio-perl-3.3.2-1.1.1.x86_64.rpm We have the project to migrate this SIP Proxy to RedHat 7.1 and used the … [View More]

2 6

(no subject)
by Alberto Llamas 17 Mar '17

17 Mar '17

Hi Guys, Who knows and can recommend a company a company where I could order Toll free numbers for EU?. I really appreciate any hint and sorry for the noise. Regards, -- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA *"Internet is all about share"*

2 1

cisco spa502g issues
by Ben Kramer 17 Mar '17

17 Mar '17

Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@<our domain> Contact:: sip:5546@<OUR OFFICE IP>:1032 Q= Expires:: 180 … [View More]

1 0

send_all blocks indefinitely
by Robert 16 Mar '17

16 Mar '17

H all, Recently, when I'm trying to load test Kamailio with a single proxy, it hung at around 15000 TPS. I attached gdb to one of the kamailio processes and I saw this: (gdb) where #0 0x00007fa8872366ad in __libc_send (fd=13, buf=0x7ffe6e0e5350, n=16, flags=-2027723091) at ../sysdeps/unix/sysv/linux/x86_64/send.c:27 #1 0x00000000004eb635 in send_all (socket=13, data=0x7ffe6e0e5350, data_len=16) at pass_fd.c:133 #2 0x00007fa4155a1b69 in tcp_close_connection (conn_id=28022) at tcp_funcs.c:… [View More]

1 0

Cdr on shutdown
by Diego Nadares 16 Mar '17

16 Mar '17

Hi Guys, Is it possible to generate cdrs or to save dialogs to db when kamailio receives a kill signal from terminal? Thanks in advance. Diego

3 5

siptrace duplicate both incoming and outgoing messages
by Grant Bagdasarian 16 Mar '17

16 Mar '17

Hi, Is it possible to instruct Kamailio to duplicate both incoming and outgoing messages automatically? If not, if I set the flag and explicitly call sip_trace, will both messages be duplicated to the capture server? Regards, Grant Bagdasarian CM

2 1

mi_fifo: No module matching <jsonrpcs> found
by Mostafa Zare 15 Mar '17

15 Mar '17

hi guys i'm newbie to kamailio. i've installed kamailio version 5 (and all the modules) from pkgs on a jessie debian. i have the below config on my kamailio.cfg when i try to start kamailio service it throws the following error. what's the problem? - systemctl status kamailio: kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled) Active: failed (Result: exit-code) since Wed 2017-03-15 19:01:17 IRST; 10s ago … [View More] Process: 9377 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255) Main PID: 1576 (code=exited, status=0/SUCCESS) Mar 15 19:01:17 kamailio-debian kamailio[9377]: ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <jsonrpcs> found Mar 15 19:01:17 kamailio-debian kamailio[9377]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 103, column 72: Can't set module parameter Mar 15 19:01:17 kamailio-debian kamailio[9377]: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <ds_probing_threshhold> of type <2> not found in module <dispatcher> Mar 15 19:01:17 kamailio-debian kamailio[9377]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 135, column 50: Can't set module parameter Mar 15 19:01:17 kamailio-debian kamailio[9377]: ERROR: bad config file (4 errors) Mar 15 19:01:17 kamailio-debian kamailio[9377]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 15 19:01:17 kamailio-debian kamailio[9377]: loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Mar 15 19:01:17 kamailio-debian systemd[1]: kamailio.service: control process exited, code=exited status=255 Mar 15 19:01:17 kamailio-debian systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Mar 15 19:01:17 kamailio-debian systemd[1]: Unit kamailio.service entered failed state. - kamailio.cfg: # *** To run in debug mode: # - define WITH_DEBUG # #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 # log_facility=LOG_LOCAL0 # Ds increasing debug... debug=2 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:0.0.0.0:5060 # listen=127.0.0.1:5060 sip_warning=no ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- # modparam("dispatcher", "db_url", DBURL) # modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") ####### Routing Logic ######## # main request routing logic route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } # dispatch destinations route(DISPATCH); # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; sl_send_reply("404", "No registrar"); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { # Destination types: # http://kamailio.org/docs/modules/4.2.x/modules/dispatcher.html#dispatcher.f… # 4 = round robin # 9 = weight-based # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "9")) { send_reply("404", "No destination"); exit; } t_set_fr(0,2000); t_on_failure("ROUTE_ADVANCE"); route(RELAY); exit; } event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: $rm $ru ($du)\n"); } event_route[dispatcher:dst-up] { xlog("L_ERR", "Destination up: $rm $ru\n"); } failure_route[ROUTE_ADVANCE] { if (t_is_canceled()) { exit; } xlog("L_NOTICE", "Media server $du failed to answer, selecting other one!\n"); if (t_check_status("503") || (t_branch_timeout() and !t_branch_replied())) { ds_mark_dst("ip"); # select the new destination if(ds_next_dst()) { # again set local timeout for reply # xlog("L_NOTICE", "Selecting next route upon failure\n"); t_set_fr(0,2000); t_on_failure("ROUTE_ADVANCE"); route(RELAY); exit; } else { #last available node failed to reply, no other destinations available send_reply("404", "No destination"); xlog("L_NOTICE","No destination, sent 404\n"); exit; } } else { xlog("L_NOTICE","Unhandled failured route!\n"); } } [View Less]

2 1

SIP contact header question
by Jack Davis 15 Mar '17

15 Mar '17

Greetings, I have a general question about the usage of SIP contact headers in the context of using Kamailio as a SIP proxy. ________________________________ [ A ] --> [ Kamailio B ] ---> [ C ] Node A originates a SIP invite, containing a valid via header and URI while setting the contact address to a user@itself and delivers it to Kamailio B which is acting as a SIP proxy. Kamailio B then uses dispatcher routing to direct the Invite to node C, adding a via line with its own … [View More]

5 12

Kamailio does'nt respond to packets from specific addresses
by עדי ויזגן 15 Mar '17

15 Mar '17

Hi everyone, I have the following problem: I have a Kamailio server that is connected to several Asterisk servers for load balacing in a LAN network with the IP's of 192.168.16.X. I recently connected with another company that is supposed to send me calls through the Kamailio server. The problem is this: I am connected to then with a cable from their network to mine. My port has the address 10.9.18.198 and they come from the address 172.18.177.18. When I get packets from their address (172...) … [View More]

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Kamailio World 2017 - A selection of presentations
by Daniel-Constantin Mierla 15 Mar '17

15 Mar '17

Hello, I would like to announce that many of the presentations at Kamailio World 2017 (May 8-10, in Berlin, Germany) are now listed at: - https://www.kamailioworld.com/k05/schedule/ Just several are not published yet, working to get all their details in place and expect to have the first draft of the schedule very soon. This edition takes place again at the same location in the famous city center of Berlin, the number of seats is limited, last year the event was fully booked. Do not delay … [View More]

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Re: [SR-Users] rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hi Daniel, yes, I hear the audio going both ways ok. In the meantime I succeeded to record something by rtpproxy but in best case only one direction, I can hear just the callee, but not the caller. In the meantime I have used start_recording in route[NATMANAGE] but before rtpproxy_manage("co"): -------------------------------------------------- if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); -------------------------------------------------- … [View More] After reading of your message I tested also with start_recording after rtpproxy_manage("co"): -------------------------------------------------- rtpproxy_manage("co"); if (is_method("INVITE") and (status=="200")) { start_recording(); } -------------------------------------------------- But the result was the same: I have 2 SIP users: 31 (X-Lite on Windows, static address 192.168.0.11) 35 (Android Zoiper, IP address 192.168.0.29 (by DHCP)) Kamailio has static address 192.168.0.13. When 35 calls 31 then I hear 1 direction in the recorded rtp file, the voice of callee. When 31 calls 35 then I don't hear anything in the recorded file. I don't understand why this difference when user 31 or 35 initiates a call!? But I hear audio conversation in both directions, just from the recorded file in one direction, or even nothing. I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp How many files should rtpproxy produce for one call? Should also a file or two for 2nd channel be generated? How can I record both audio ways? Thank you Marko The whole function route[NATMANAGE]: # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } ----- Original Message ----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Marko Tirs <marko.tirs(a)yahoo.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 14, 2017 5:23 PM Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording Hello, I think you can call the start recording function just after rtpproxy_manage(). If you don't use the start recording function, is the audio going both ways without problems? Cheers, Daniel [View Less]

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Re: [SR-Users] rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hi Daniel, yes, I hear the audio going both ways ok. In the meantime I succeeded to record something by rtpproxy but in best case only one direction, I can hear just the callee, but not the caller. In the meantime I have used start_recording in route[NATMANAGE] but before rtpproxy_manage("co"): -------------------------------------------------- if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); -------------------------------------------------- … [View More] After reading of your message I tested also with start_recording after rtpproxy_manage("co"): -------------------------------------------------- rtpproxy_manage("co"); if (is_method("INVITE") and (status=="200")) { start_recording(); } -------------------------------------------------- But the result was the same: I have 2 SIP users: 31 (X-Lite on Windows, static address 192.168.0.11) 35 (Android Zoiper, IP address 192.168.0.29 (by DHCP)) Kamailio has static address 192.168.0.13. When 35 calls 31 then I hear 1 direction in the recorded rtp file, the voice of callee. When 31 calls 35 then I don't hear anything in the recorded file. I don't understand why this difference when user 31 or 35 initiates a call!? But I hear audio conversation in both directions, just from the recorded file in one direction, or even nothing. I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp How many files should rtpproxy produce for one call? Should also a file or two for 2nd channel be generated? How can I record both audio ways? Thank you Marko The whole function route[NATMANAGE]: # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } ----- Original Message ----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Marko Tirs <marko.tirs(a)yahoo.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 14, 2017 5:23 PM Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording Hello, I think you can call the start recording function just after rtpproxy_manage(). If you don't use the start recording function, is the audio going both ways without problems? Cheers, Daniel [View Less]

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rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hello experts, I'm newbie in Kamailio and have a problem with my 1st installation. I just want to control the calls between SIP clients registered by Kamailio and record the conversations. I installed rtpproxy and started with rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy -d DBUG:LOG_LOCAL0 After start_recording() in the function onreply_route[MANAGE_REPLY] rtpproxy creates the files <call-id>.rtp and <call-id>.rtcp but they remain at zero length. 2nd … [View More]problem is that the media stream is broken in one direction after start_recording() has been executed, the callee doesn't hear the caller. How is the right Kamailio configuration for recording conversations? Thank you Regards Marko Modified kamailio-basic.cfg partly: #!KAMAILIO #!define WITH_NAT #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!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 ####### Routing Logic ######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (is_method("INVITE") and (status=="200")) { xlog("START_RECORDING #1\n"); start_recording(); xlog("START_RECORDING #2\n"); } else if(status=~"[12][0-9][0-9]") route(NATMANAGE); } [View Less]

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Is kamailio support load balancing for websockets
by SureshTummala 14 Mar '17

14 Mar '17

Hi, I want to use kamailio as a load balancer for web sockets. Is kamailio supports LB functionality for web-sockets? I can able to use kamailio as a proxy and can make calls with websockets. Now i want to add LB functionality. Setup: UA1 -->LB -->Proxy UA2 <--LB <--Proxy LB is not forwarding the packets from Proxy to UA Please help me Regards,Suresh Tummala

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Several Asterisk on this same IP
by przeqpiciel 14 Mar '17

14 Mar '17

I would like to ask you how you deal with several asterisks and kamailio on that same IP address, I have installation where i route 5060 to internal server with kamailio and I would like to route RTP to asterisks, but is any way to get around a problem with RTP ports collisions?

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how to unregister an remote registration at uac module
by José Lopes 13 Mar '17

13 Mar '17

Hello, I am using the module uac to do remote registration. But sometimes i need to do unregister, to control or debug the registration process. I tried using kamcmd with uac.reg_disable, but this doesn't unregister. There is any way to do it? Best regards, Jose Lopes

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Segmentation fault - dialog RPC
by pars3c 13 Mar '17

13 Mar '17

Dear all, many times, when i call dialog_list on dialog module with jsonrpc, kamailio crash. The version i use is: 4.4.2 Thanks for any suggestion. #0 0x00007f2a706eb00a in internal_rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940, profile_name=0x7fffb9fa8b10, value=0x0) at dialog.c:1666 1666 if ((!value || (STR_EQ(*value, ph->value))) && ph->dlg) { The dump , with bt full is this : #0 0x00007f2a706eb00a … [View More]in internal_rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940, profile_name=0x7fffb9fa8b10, value=0x0) at dialog.c:1666 profile = 0x7f2a5098fb40 ph = 0x0 i = 10 #1 0x00007f2a706eb6e4 in rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940) at dialog.c:1772 profile_name = {s = 0x1c78b00 "outbound", len = 8} value = {s = 0x0, len = 0} #2 0x00007f2a4fbe45c6 in jsonrpc_dispatch (msg=0x7fffb9fa9f10, s1=0x0, s2=0x0) at jsonrpc-s_mod.c:1091 rpce = 0x7f2a75609360 ctx = 0x7f2a4fdf9940 ret = 0 nj = 0x1c76af0 val = {s = 0x1c77b20 "dlg.profile_list", len = 16} __FUNCTION__ = "jsonrpc_dispatch" #3 0x000000000041f4ad in do_action (h=0x7fffb9fa9e30, a=0x7f2a75584700, msg=0x7fffb9fa9f10) at action.c:1054 ret = -5 v = 4288256 dst = {send_sock = 0x1b9fa9b23, to = {s = {sa_family = 14080, sa_data = "Xu*\177\000\000\b7Xu*\177\000"}, sin = { sin_family = 14080, sin_port = 30040, sin_addr = {s_addr = 32554}, sin_zero = "\b7Xu*\177\000"}, sin6 = { sin6_family = 14080, sin6_port = 30040, sin6_flowinfo = 32554, sin6_addr = {__in6_u = { __u6_addr8 = "\b7Xu*\177\000\000S \205\000\060\000\000", __u6_addr16 = {14088, 30040, 32554, 0, 8275, 133, 48, 0}, __u6_addr32 = {1968715528, 32554, 8724563, 48}}}, sin6_scope_id = 26}}, id = 0, proto = 16 '\020', send_flags = { f = 155 '\233', blst_imask = 250 '\372'}} tmp = 0x7fffb9fab700 "\t" new_uri = 0x0 end = 0x0 crt = 0x7fffb9fa92d0 "" cmd = 0x7f2a754adcc0 len = 0 user = 0 uri = {user = {s = 0x12c <Address 0x12c out of bounds>, len = 0}, passwd = {s = 0x7fffb9fa92d0 "", len = 1969341184}, host = { s = 0x7f2a75475010 "\001", len = 4742114}, port = {s = 0x0, len = 0}, params = { s = 0x98206f7473 <Address 0x98206f7473 out of bounds>, len = 1969341256}, sip_params = { s = 0x7fffb9fa91e0 "`\222\372\271\377\177", len = 6538286}, headers = {s = 0x1c2b6f0 "\300\300\302\001", len = 0}, port_no = 50838, proto = 30049, type = 32554, flags = 0, transport = {s = 0x0, len = -940795965}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x6222286e00000000 <Address 0x6222286e00000000 out of bounds>, len = 577270895}, maddr = { s = 0x3a746e65696c6309 <Address 0x3a746e65696c6309 out of bounds>, len = 0}, method = {s = 0x75c5f0 "core", len = -1174760864}, lr = {s = 0x764661 "rval_destroy", len = 7718524}, r2 = {s = 0x7f2a7561c348 "\002", len = 1967607824}, gr = {s = 0x7fffb9fa91b8 "H\303au*\177", len = 1967607824}, transport_val = {s = 0x7f2a7561c300 "\300\004", len = 1969341256}, ttl_val = {s = 0x7f2a7557e200 "\020", len = -1174760316}, user_param_val = {s = 0x7f2a75583700 "\002", len = 1969341256}, maddr_val = {s = 0x416f00 "1\355I\211\321^H\211\342H\203\344\360PTI\307\300\060\336q", len = -1174751488}, method_val = { ---Type <return> to continue, or q <return> to quit--- s = 0x0, len = 0}, lr_val = {s = 0x7fffb9fa9260 "Ð\372\271\377\177", len = 5581478}, r2_val = {s = 0x0, len = 0}, gr_val = { s = 0x3a28d <Address 0x3a28d out of bounds>, len = 1969177728}} next_hop = {user = {s = 0x1c2b6f0 "\300\300\302\001", len = 1969330344}, passwd = { s = 0x69692c22 <Address 0x69692c22 out of bounds>, len = 29537760}, host = { s = 0x400000000 <Address 0x400000000 out of bounds>, len = -1174761304}, port = { s = 0x71657228 <Address 0x71657228 out of bounds>, len = 1969342102}, params = {s = 0x0, len = 1}, sip_params = {s = 0x0, len = -1174761328}, headers = {s = 0x1 <Address 0x1 out of bounds>, len = 2}, port_no = 40672, proto = 47610, type = 32767, flags = 3120206016, transport = {s = 0x7a6563697220616c <Address 0x7a6563697220616c out of bounds>, len = 1701736297}, ttl = { s = 0x7f2a75580848 "\001", len = -1174760828}, user_param = {s = 0x7fffb9fa9f10 "\b\005", len = -1174757840}, maddr = { s = 0x65722e2e223a7069 <Address 0x65722e2e223a7069 out of bounds>, len = 1702129517}, method = {s = 0x7f2a7561c696 "/sip", len = 29855024}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x500000004 <Address 0x500000004 out of bounds>, len = 4}, transport_val = {s = 0x400000004 <Address 0x400000004 out of bounds>, len = 6552774}, ttl_val = { s = 0x7fffb9fa9060 "", len = 1967607824}, user_param_val = {s = 0x1000001 <Address 0x1000001 out of bounds>, len = 1967607824}, maddr_val = {s = 0x7fffb9fa90b0 "\020PGu*\177", len = 1956545302}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14985000}, r2_val = { s = 0x165e80 <Address 0x165e80 out of bounds>, len = 1792216}, gr_val = {s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 8}} u = 0x7f2a75583f30 port = 0 dst_host = 0x7f2a75583700 i = 32767 flags = 0 avp = 0x7fffb9fa9e30 st = {flags = 151652617, id = 29193, name = {n = 0, s = {s = 0x6e65090a00000000 <Address 0x6e65090a00000000 out of bounds>, len = -1174761256}, re = 0x6e65090a00000000}, avp = 0x7fffb9fa9000} sct = 0x7fffb9fa94d0 sjt = 0x7fffb9fa9f10 rve = 0x416f00 mct = 0x7fffb9fab700 rv = 0x92922726f727265 rv1 = 0x0 c1 = {cache_type = 3120205776, val_type = 32767, c = {avp_val = {n = -940803561, s = { s = 0x31c7ec7a17 "\213D$8H\201\304\070\002", len = -1174757616}, re = 0x31c7ec7a17}, pval = {rs = { s = 0x31c7ec7a17 "\213D$8H\201\304\070\002", len = -1174757616}, ri = -1174757840, flags = 32767}}, i2s = "vp(s:inserver_erPor)\","} s = {s = 0x1c2b6f0 "\300\300\302\001", len = 0} srevp = {0x0, 0x0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, ---Type <return> to continue, or q <return> to quit--- select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" #4 0x000000000042c218 in run_actions (h=0x7fffb9fa9e30, a=0x7f2a75584700, msg=0x7fffb9fa9f10) at action.c:1549 t = 0x7f2a75584700 ret = -1 mod = 0x0 ms = 0 __FUNCTION__ = "run_actions" #5 0x000000000041f416 in do_action (h=0x7fffb9fa9e30, a=0x7f2a75585b10, msg=0x7fffb9fa9f10) at action.c:1045 ret = 1 v = 1 dst = {send_sock = 0x7fffb9fa9b50, to = {s = {sa_family = 64710, sa_data = "c\000\000\000\000\000\060\233\372\271\377\177\000"}, sin = {sin_family = 64710, sin_port = 99, sin_addr = {s_addr = 0}, sin_zero = "0\233\372\271\377\177\000"}, sin6 = { sin6_family = 64710, sin6_port = 99, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = "0\233\372\271\377\177\000\000Ù¨bu*\177\000", __u6_addr16 = {39728, 47610, 32767, 0, 43225, 30050, 32554, 0}, __u6_addr32 = {3120208688, 32767, 1969400025, 32554}}}, sin6_scope_id = 366}}, id = 10, proto = -24 '\350', send_flags = {f = 234 '\352', blst_imask = 98 'b'}} tmp = 0x1467562aa20 <Address 0x1467562aa20 out of bounds> new_uri = 0x7a0a02 "parse_headers" end = 0x79db5c "core: parser/msg_parser.c" crt = 0x40 <Address 0x40 out of bounds> cmd = 0x7f2a75475010 len = 1967607824 user = 32554 uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x7fffb9fa9b70 "@\234\372\271\004", len = 6529985}, host = { s = 0x416f00 "1\355I\211\321^H\211\342H\203\344\360PTI\307\300\060\336q", len = -1174751488}, port = {s = 0x0, len = 0}, params = {s = 0x7fffb9fa9ba0 "p\234\372\271\377\177", len = 6661320}, sip_params = {s = 0x7fffb9fa99f0 "", len = 1967607824}, headers = {s = 0x0, len = 1967607824}, port_no = 39488, proto = 47610, type = 32767, flags = 6552774, transport = { s = 0x7fffb9fa9a20 "", len = 1967607824}, ttl = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 1967607824}, user_param = {s = 0x7fffb9fa9a70 "\240\232\372\271\377\177", len = 1956545302}, maddr = { s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 0}, method = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14986448}, lr = {s = 0x1658d8 <Address 0x1658d8 out of bounds>, len = 1790768}, r2 = { s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 8}, gr = {s = 0x12e <Address 0x12e out of bounds>, len = 1967607824}, transport_val = {s = 0x14600000000 <Address 0x14600000000 out of bounds>, len = 1969417808}, ttl_val = { s = 0x7fffb9fa9aa0 "\240\233\372\271\377\177", len = 4742114}, user_param_val = {s = 0x0, len = 1967607824}, maddr_val = { s = 0x0, len = 0}, method_val = {s = 0x7fffb9fa9ba0 "p\234\372\271\377\177", len = 6534031}, lr_val = {s = 0x79d630 "core", len = 6552774}, r2_val = {s = 0x7fffb9fa9b00 "n\001", len = 1969417880}, gr_val = {s = 0x7f2a7562aa20 "", len = 1969399988}} next_hop = {user = {s = 0x7fffb9fa9820 "", len = 1967607824}, passwd = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 1967607824}, host = {s = 0x7fffb9fa9870 "\240\230", <incomplete sequence \372\271>, len = 1956545302}, port = { ---Type <return> to continue, or q <return> to quit--- s = 0x1c588e1 "\311\305\001", len = 0}, params = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14986216}, sip_params = {s = 0x1659c0 <Address 0x1659c0 out of bounds>, len = 1969416936}, headers = {s = 0x7f2a7562aa20 "", len = 1969399993}, port_no = 301, proto = 0, type = ERROR_URI_T, flags = 0, transport = { s = 0x7f2a7562a8d9 "Content-Type: application/json-rpc\r\nCache-Control: no-cache\r\nPragma: no-cache\r\nUser-Agent: Java/1.8.0_121\r\nHost: \r\nAccept: text/html, image/gif, image/jpeg, *; q=.2, */*; q=.2\r\nConn"..., len = 0}, ttl = { s = 0xb9fa98a0 <Address 0xb9fa98a0 out of bounds>, len = 1969416936}, user_param = {s = 0x0, len = 1967607824}, maddr = { s = 0x0, len = 0}, method = {s = 0x7fffb9fa99a0 "\240\233\372\271\377\177", len = 6534031}, lr = {s = 0x79d630 "core", len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x7fffb9fa9e7c "\260\f\222\017", len = -1174757736}, transport_val = { s = 0x79d630 "core", len = -1174757704}, ttl_val = {s = 0x7a0a10 "get_hdr_field", len = 7986012}, user_param_val = { s = 0xe8 <Address 0xe8 out of bounds>, len = 1967607824}, maddr_val = {s = 0x79d630 "core", len = 1969416864}, method_val = { s = 0x7a0a10 "get_hdr_field", len = 7986012}, lr_val = {s = 0xe8 <Address 0xe8 out of bounds>, len = 1967607824}, r2_val = { s = 0x7f2a75475010 "\001", len = 1969416864}, gr_val = {s = 0x7f2a7562eaa0 <incomplete sequence \350>, len = 1969399988}} u = 0x7f2a7562ee50 port = 0 dst_host = 0x80067e961 i = 0 flags = 0 avp = 0x7f2a75475010 st = {flags = 3120207856, id = 32767, name = {n = 1967607824, s = {s = 0x7f2a75475010 "\001", len = 1969239958}, re = 0x7f2a75475010}, avp = 0x7f2a75475010} sct = 0x4b9fa9c40 sjt = 0x79d630 rve = 0x7f2a75582f20 mct = 0x7fffb9fab700 rv = 0x63fcc6 rv1 = 0x0 c1 = {cache_type = 43, val_type = RV_NONE, c = {avp_val = {n = 0, s = {s = 0x0, len = 7718384}, re = 0x0}, pval = {rs = {s = 0x0, len = 7718384}, ri = 26, flags = 0}}, i2s = "\t\000\000\000\000\000\000\000huTu*\177\000\000\000\000\000\000\000"} s = {s = 0x7fffb9fa99f0 "", len = 1969239512} srevp = {0x3000000018, 0x7fffb9fa9ab0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" [View Less]

2 2

unable to receive response from freeswitch
by ysuryas . 13 Mar '17

13 Mar '17

I have configured kamailio (192.168.56.102) and freeswitch(192.168.56.103) in separate virtualboxes when making a call, the invite goes to proxy and it is forwarded to freeswitch but response from freeswitch is "unreachable port" attached the wireshark trace: I don't why 10.0.2.15 is making the Invite to freeswitch, i was expecting 192.168.56.102 to do that. UA -> 192.168.56.102 -> 192.168.56.103 Any suggestions on this? [image: Inline image 1] Thanks, Surya

2 1

rtpproxy doesn't record after start_recording
by Marko Tirs 11 Mar '17

11 Mar '17

Hello experts, I'm newbie in Kamailio and have a problem with my 1st installation. I just want to control the calls between SIP clients registered by Kamailio and record the conversations. I installed rtpproxy and started with rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy -d DBUG:LOG_LOCAL0 After start_recording() in the function onreply_route[MANAGE_REPLY] rtpproxy creates the files <call-id>.rtp and <call-id>.rtcp but they remain at zero length. 2nd … [View More]problem is that the media stream is broken in one direction after start_recording() has been executed, the callee doesn't hear the caller. How is the right Kamailio configuration for recording conversations? Thank you Regards Marko Modified kamailio-basic.cfg partly: #!KAMAILIO #!define WITH_NAT #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!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 ####### Routing Logic ######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (is_method("INVITE") and (status=="200")) { xlog("START_RECORDING #1\n"); start_recording(); xlog("START_RECORDING #2\n"); } else if(status=~"[12][0-9][0-9]") route(NATMANAGE); } [View Less]

1 0

Dispatcher Module - Hava limit?
by Rodrigo Moreira 11 Mar '17

11 Mar '17

Hello, I am experiencing the following problem: my dispatcher module is losing many calls when the scenario for balancing is about 175 CPS or more. Is there a limit that the module can do? Where do I find this information? Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069

3 6

Multiple parallel branches, one invalid
by Nathan Ward 10 Mar '17

10 Mar '17

Hi, I have a scenario I’m not sure of the best way to solve. I use alias_db and have multiple destinations for a single number, and have append_branches set to 1. I then use lookup_branches to get details from the location table for each branch. I then do t_load_contacts and t_next_contacts, then t_relay to parallel fork the calls to the destination, and all the phones ring at once. This all works great if all the destinations from the dbaliases table are online. However, if one of them is … [View More]

2 1

404 not here for BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 … [View More]Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/ Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95: F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+ rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. 3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487;branch=z9hG4bK-524287-1--- 1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060;rport=5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. [View Less]

2 1

sip6 BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 … [View More]Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95:F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;branch=z9hG4bK-524287-1---1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060 ;rport=5060;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. [View Less]

3 2

Bounty for topos fix
by Trevor Peirce 09 Mar '17

09 Mar '17

Hello, I would like to offer a $100 USD bounty if a developer is able to provide a fix for the two topos problems I have experienced. I will add another $50 if this can be solved within 48 hours of the time stamp of this message. The bug report is at: https://github.com/kamailio/kamailio/issues/1005 You may find the most relevant information in my last few updates. Please refer to mar1-nat.txt and mar1-reinvite.txt for the most recent SIP captures, and the post below that for example … [View More]

1 1

Changes in LCR Target List doesn't takes place immediately
by Nitesh Lohchab 09 Mar '17

09 Mar '17

Hello, I believe LCR Target list is stored in the database and any changes made here should affect the behaviour of the calls imedialtely. I have changed one of my Target List Rule , where I changed my gateway Id to another gateway. But the calls are still being routed via previous gateway. Is there something I can do to make the changes in production hours. Sometimes one of carriers delivers bad call quality I would like to switch the carriers. -- Regards, Nitesh Lohchab

2 1

rtpengine sending rtp to wrong endpoint after reinvite
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, One of our customers is using a SEMS box to place two outbound calls using our sip trunk. Once the first call is connected a second call is placed and when the second call answers their server sends a re-invite to switch audio ports so the rtp traffic doesn't flow through their server anymore but is routed inside our platform. Basically, they just switch SDP's of both calls. It seems like a random issue, and is not really reproducible, except for placing multiple calls and sometimes both … [View More]

2 3

Access avp between sip transactions
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, Is it possible to access an avp variable from a different sip transaction? On an INVITE is set an avp, but on ReINVITE I would like to access this avp again and get the value. Currently the value is null, which makes sense. Not sure if this is possible and if it violates any rules inside the Kamailio engine? I guess I could use a htable, but I'm just wondering if there is a built-in way for doing this. Regards, Grant Bagdasarian CM

2 1

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