- sr-users - kamailio.org

LCR - too few dots/colons
by Annus Fictus 11 Jun '17

11 Jun '17

Hello, any explication about these messages? DEBUG: lcr [../../core/resolve.h:258]: str2ip(): str2ip: ERROR: too few dots in [253719602] DEBUG: lcr [../../core/resolve.h:355]: str2ip6(): str2ip6: ERROR: too few colons in [253719602] Thnak you Regards

2 1

openrcs
by Thufir Hawat 09 Jun '17

09 Jun '17

is there a discussion list for openrcs? I couldn't find a mailing list. thanks, Thufir

2 11

route request to multiple pstn gws
by 赵国杰 09 Jun '17

09 Jun '17

Hi all. I need to route calls to different counties to different pstn gateways. But in the config file, i can only set one pstn gw ip:port in pstn section. How should i do it? Can i use carrierroute module to do this? If yes, can anyone paste a config sample? many thanks Jesse

1 0

WITH_NATSIPPING option
by Nguyen Tran Nhan 08 Jun '17

08 Jun '17

I am config rtpproxy with kamailio but I am stuck at - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING It is configure in client side? Thanks, Nhan

2 3

route to different pstn
by zhaoguojie2010＠163.com 08 Jun '17

08 Jun '17

Hi all. I need to route calls to different counties to different pstn gateways. But in the config file, i can only set one pstn gw ip:port in pstn section. How should i do it? Can i use carrierroute module to do this? If yes, can anyone paste a config sample? many thanks Jesse 发自我的华为手机

1 0

Re: [SR-Users] No private memory left
by Vuk Radovanovic 08 Jun '17

08 Jun '17

Its debian 8.8 Regards, Vuk -------- Original message -------- Sender: Daniel-Constantin Mierla <miconda(a)gmail.com> Time: Wed 6/7 12:07 To: Vuk Radovanovic <vuk(a)gripenpro.com>,"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No private memory left >Hello, > >what is your operating system? > >Cheers, >Daniel > > >On 07.06.17 11:26, Vuk Radovanovic wrote: >> Hi Daniel, >> >> So the ps auxw is telling me the starting options for shm and pkg >> memory (-m -M) and options for user and group are not defined when >> starting via init script. >> >> Ive checked the init script in case i made an error when changing the >> memory directive but its fine, as provided in the GIT package. >> >> I am running kamailio 5.0 from git. >> >> Userblacklist trying to load whole db at once sounds scary, my main >> goal was implementing a redis whitelist with around 5mil numbers, now >> I am wondering is it even feasible ? >> >> root 3181 0.1 0.3 222300 12216 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3183 0.0 0.0 222300 3856 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3184 0.0 0.0 222300 3732 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3185 0.0 0.0 222300 3732 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3186 0.0 0.0 222300 3732 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3187 0.0 0.0 222300 3732 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3188 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3189 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3190 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3191 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3192 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3195 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3196 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3199 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3201 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3203 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3204 0.0 0.0 222300 3740 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3206 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3207 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3209 0.0 0.0 222300 3736 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3211 0.0 0.0 222304 1620 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3213 0.0 0.0 222300 1620 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3214 0.0 0.1 222300 5524 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3215 0.0 0.1 222300 5520 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3217 0.0 0.1 222300 5520 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3220 0.0 0.1 222300 5520 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.cfg >> root 3221 0.0 0.1 222300 5520 ? S 11:18 0:00 >> /usr/local/sbin/kamailio -P /var/run/kamailio.pid -f >> /usr/local/etc/kamailio//kamailio.c

3 8

DROUTING Tables
by Annus Fictus 08 Jun '17

08 Jun '17

Hello, I can't find DROUTING Tables on the standard kamailio database. dr_gateways dr_rules dr_groups Have I to create them manually? Thank you Regards

3 3

KAMAILIO STARTUP ERROR
by christopher kamutumwa 08 Jun '17

08 Jun '17

hello all, I am new to kamailio on centos 7 and i have finished installation following steps below; http://computingforgeeks.com/how-to-install-latest-kamailio-sip-server-on-c… but when i try to startup service i get fail with below how can this be resolved? [root@li1496-213 ~]# 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; bad; vendor preset: disabled) Active: failed (Result: exit-code) since Wed 2017-06-07 12:16:26 UTC; 1min 59s ago Docs: man:systemd-sysv-generator(8) Process: 26015 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command handle_publish (params 0) Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 715, column 18: unknown command, missing loadmodule? Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command handle_subscribe (params 0) Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 718, column 20: unknown command, missing loadmodule? Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: ERROR: bad config file (7 errors) Jun 07 12:16:26 li1496-213.members.linode.com kamailio[26015]: 0(26023) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 07 12:16:26 li1496-213.members.linode.com systemd[1]: kamailio.service: control process exited, code=exited status=1 Jun 07 12:16:26 li1496-213.members.linode.com systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Jun 07 12:16:26 li1496-213.members.linode.com systemd[1]: Unit kamailio.service entered failed state. Jun 07 12:16:26 li1496-213.members.linode.com systemd[1]: kamailio.service failed. regards chris

2 2

Docs: database structure
by Daniel-Constantin Mierla 07 Jun '17

07 Jun '17

Hello, I regenerated recently the document with the details for Kamailio's database structure, it is available at: * https://kamailio.org/docs/db-tables/kamailio-db-devel.html If you haven't read it so far, it is a good source to understand the purpose of the fields needed by kamailio from database. Contributions to the document are more than appreciated! You can update the xml files from src/lib/srdb1/schema/ and then make a pull request via github. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

No private memory left
by Vuk Radovanovic 07 Jun '17

07 Jun '17

Hello, I am having trouble with userblacklist module. I am using mysql db, when globalblacklist table is empty or has couple of records kamailio loads fine. When I populate the table with cca 50k entries I get an error no private memory error as pasted below. I have edited both the init.d script and /etc/default cfg file to to have 1GB pkg and 2 GB SHM memory with no change. Init script change: SHM_MEMORY=2048 PKG_MEMORY=1024 Init script output. Jun 07 01:03:45 km systemd[1]: Starting Kamailio SIP Server... Jun 07 01:03:45 km kamctl[1377]: ** INFO: Starting Kamailio : Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: <core> [db_row.c:114]: db_allocate_row(): no private memory left Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: db_mysql [km_row.c:57]: db_mysql_convert_row(): could not allocate row Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: db_mysql [km_res.c:188]: db_mysql_convert_rows(): error while converting row #41249 Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: db_mysql [km_res.c:217]: db_mysql_convert_result(): error while converting rows Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: db_mysql [km_dbase.c:261]: db_mysql_store_result(): error while converting result Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: <core> [db_query.c:139]: db_do_query_internal(): error while storing result Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: userblacklist [db.c:129]: db_reload_source(): error while executing query on db table 'globalblacklist' Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: userblacklist [userblacklist.c:610]: reload_sources(): cannot reload source from 'globalblacklist' Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: <core> [core/sr_module.c:923]: init_mod_child(): Error while initializing module userblacklist (/usr/local/lib64/kamailio/modules/userblacklist.so) Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: ERROR: <core> [core/pt.c:330]: fork_process(): init_child failed for process 1, pid 1405, "udp receiver child=0 sock=127.0.0.1:5060" Jun 07 01:03:46 km /usr/local/sbin/kamailio[1405]: CRITICAL: <core> [main.c:1606]: main_loop(): Cannot fork Jun 07 01:03:46 km /usr/local/sbin/kamailio[1403]: ALERT: <core> [main.c:741]: handle_sigs(): child process 1405 exited normally, status=255 Jun 07 01:03:48 km kamctl[1377]: ** ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed Jun 07 01:03:48 km systemd[1]: kamailio.service: control process exited, code=exited status=1 Jun 07 01:03:48 km systemd[1]: Failed to start Kamailio SIP Server. Jun 07 01:03:48 km systemd[1]: Unit kamailio.service entered failed state. Regards, Vuk

2 4

Dispatcher - postponed GW restoration
by Ivan Matvieiev 07 Jun '17

07 Jun '17

Hello I'm trying to implement following scenario: - remote GW is Active (receives calls) and not Probed (no SIP Options are being sent); - in case if remote GW failes, it starts to be Probed and become Inactive ("ds_probing_threshold" does this); - when remote GW returns it remains in Probed and Inactive state for a while; - after probing for some amount of time (or a number of successful responses) it become Active and not Pobed (it can be achieved with "ds_inactive_threshold"); The issue I'm facing is that "ds_inactive_threshold" will bring GW back only after the number of successful responses that can't be provided using "ds_probing_mode" with value 2. It stops to produce SIP Options by successful reply before threshold of successful replies is reached. In case of "ds_probing_mode" with value 3 I haven't found a way to stop Probing when GW transits to state Active. Is there a way to make GW Active after it was Inactive and was successfully probed for arbitrary number of times? Some valuable information: Kamailio: v 4.4.2 ... modparam("dispatcher", "ds_ping_interval", 3) modparam("dispatcher", "ds_probing_threshold", 3) modparam("dispatcher", "ds_inactive_threshold", 60) modparam("dispatcher", "ds_probing_mode", 2) ... If I missed any information please tell me. With best regards Ivan

1 0

TLS to UDP translation
by 赵国杰 07 Jun '17

07 Jun '17

Hi everyone, I have a kamailio server with TLS enabled. I need to forword UAC request to a PSTN gateway which only support UDP. Is it possible that kamailio can translate this with proper configuration? How can i do this? many thanks Jesse

2 5

Using reserved keywords in cfg variables
by Paul Claudiu Boriga 07 Jun '17

07 Jun '17

Hello all, We found out that when using cfg variables certain keywords are not allowed, and Kamailio logs a generic error: <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 256, column 9-14: syntax error For example we cannot use words like enable/disable for var name or group name but we can use disabled or enabled: group_name.enable = 1 descr "test variable" Looking into the cfg.lex file, I noticed that these are keywords which will be interpreted as values (I don't have too much experience with flex/bison so I am not sure I got this right). We have been using Kamailio 4.4 to test this, but I think it is not specific to a version. Is this behavior documented somewhere? If not, where should it be documented? Also can we have a more specific log message in this case? Thank you, Claudiu Boriga.

2 6

Issue with Dispatcher Failover algorithm
by Cristi Constantin 07 Jun '17

07 Jun '17

Hello, I have an issue with the behavior of ds_select_dst() method in case of using algorithm 8 (dispatch based on priority). In case both destinations are available, they are used alternatively, instead of always using the one with the highest priority. By searching through the mailing list I found that the lower the priority number, the higher the priority (1 is first, 2 is second, etc.). Based on this, I set the following in dispatcher.list: 7 sip:10.126.54.51:5060;transport=sctp 9 1 7 sip:10.126.55.51:5060;transport=sctp 9 2 After running "kamcmd dispatcher.reload" and "kamcmd dispatcher.list", I have the confirmation that the priorities are correctly set: SET: { ID: 7 TARGETS: { DEST: { URI: sip:10.126.55.51:5060 ;transport=sctp FLAGS: AP PRIORITY: 2 } DEST: { URI: sip:10.126.54.51:5060 ;transport=sctp FLAGS: AP PRIORITY: 1 } } } The dispatcher settings from kamailio.cfg is the following: modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:kamailio@sdp-t-lb2") modparam("dispatcher", "ds_probing_mode", 3) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") And the code which should do the displatch is the following: if (ds_is_from_list(7, 3, "$ru")) { remove_hf("Route"); if(!ds_select_dst("7", "8")) { send_reply("404", "No destination"); exit; } } With the settings above, I was expecting to always have the messages delivered to 10.126.54.51. Instead, they are dispathed to 10.126.54.51 and 10.126.55.51 alternatively. Thanks a lot, Cristian

6 15

Kamailio / Docker
by sain hassam caicedo 07 Jun '17

07 Jun '17

Dear Kamailio, My name is Sain, I from Colombia. I wonder if is possible to use Kamailio with Docker in Ubuntu 16 ? Also, respectfully do you have an easy installation manual or any recommendation to start doing some testing? I am interesting to learn more about Kamailio . Thank you very much

2 1

Auth_radius - digest auth problem
by Donat Zenichev 06 Jun '17

06 Jun '17

Hi all. Have a problem with radius authorization. I'm using auth_radius.so modparams, only path to client file: modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") Freeradius installed and is working properly, radtest authentication from kamailio host succeed . How authorization block looks like: if (!is_present_hf("Authorization")) { xlog("L_NOTICE", "----- Athorization HF is not found - passing the challenge -----\n"); if (nat_uac_test("2")) { force_rport(); } www_challenge("$fd", "1"); exit; if (!radius_www_authorize("$fd","$fU")) { if (nat_uac_test("2")) { force_rport(); } xlog("L_NOTICE", "----- Registeration $au@$ar ($fU) from $si:$sp Rejected. Code: $rc -----\n"); sl_send_reply("401","Unauthorized"); exit; Radius log is filled by rows like: Auth: [digest] Cleartext-Password or Digest-HA1 is required for authentication. Tried to use radius_www_authorize without $fU - didn't change anything. Tried to use www_challenge without qop - didn't change anything. So, this solution is quite simple, but I have a fail while digest authentication. Any ideas? -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

4 8

Any document to study and develop kamailio
by sgy 06 Jun '17

06 Jun '17

Hi All, I can't find any document(PDF) to study and develop kamailio, do someone show me where it is. thank you -- Best Regards, Shengy

3 3

Kamailio 4.4.4 crash in tcp_read_headers()
by Armen Babikyan 06 Jun '17

06 Jun '17

Hello, Over the past few months, I've seen a smattering of kamailio crashes on various systems with identical backtraces: SIGSEGV in tcp_read_headers(), at tcp_read.c line 628. Example here: https://pastebin.com/qJ3ypnVz Note that in frame 0, print *c shows that req->parsed is pointing to an address exactly 8GB lower than req->buf. That req->parsed is pointing to an invalid memory location, crashing kamailio when the location is dereferenced. In other coredumps, I see that req->parsed is pointing to an address exactly 4GB lower than req->buf. Other info: This is Kamailio 4.4.4 on x86_64. I've not had success trying to reproduce this yet. Also worth noting that the crashes seem to be consistently associated with processing traffic from a UA connected over SIP/TCP; I've seen no other transport associated with this crash. Thoughts are welcome. Thanks! Armen

2 3

Rtpproxy didn't work
by sgy 03 Jun '17

03 Jun '17

I install rtpproxy and config as follow, but it didn't work, is my config OK ? errorlog: Jun 3 15:00:56 iZ2ze3uk97m2bfv681kyx4Z /usr/local/sbin/kamailio[28971]: ERROR: rtpproxy [rtpproxy.c:2552]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy Jun 3 15:00:58 iZ2ze3uk97m2bfv681kyx4Z /usr/local/sbin/kamailio[28971]: ERROR: rtpproxy [rtpproxy.c:2552]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy My config: PUB_IP: public ip of kamailio PVT_IP: private ip of kamailio 1. start rtpproxy rtpproxy -l PUB -s udp:localhost:7722 -F 2. kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ALIASDB #!define WITH_USRLOCDB #!define WITH_NAT listen=PVT_IP modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") -- Best Regards, Shengy

3 3

Rtpproxy didn't work
by sgy 03 Jun '17

03 Jun '17

I install rtpproxy and config as follow, but it didn't work, is my config OK ? My config: PUB_IP: public ip of kamailio PVT_IP: private ip of kamailio 1. start rtpproxy rtpproxy -l PUB -s udp:localhost:7722 -F 2. kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ALIASDB #!define WITH_USRLOCDB #!define WITH_NAT listen=PVT_IP modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") -- Best Regards, Shengy

1 0

Adding debugger module to kazoo-kamailio
by Joe Baran 02 Jun '17

02 Jun '17

I’m hoping that someone might have some suggestions on how to approach this. I’ve been working with kamailio as part of kazoo (via RPM install) and have been attempting to add the debugger module so I can trace things as they pass through kamailio. However, I need some help in determining the most effective way of adding this. Has anyone already done this / have any suggestions? Thanks, Joe

2 1

topos module - possible bug
by Pete Kelly 02 Jun '17

02 Jun '17

Hi I am using the topos module when bridging 2 networks with Kamailio. The INVITE/200OK part of the transaction is working fine (i.e. the Contact on both sides matches correctly the corresponding network). However when the ACK is sent into Kamailio, instead of realising the next hop is myself and skipping it, Kamailio is sending the ACK directly to itself as a packet, causing the call setup to break. Does anyone have any advice for this situation?

3 34

Kamctl does not work. Kamailio 4.4
by Denys Pozniak 02 Jun '17

02 Jun '17

Hello! I have installed Kamailio 4.4 on CentOS7 from repo. ( http://download.opensuse.org/repositories/home:/kamailio:/v4.4.x-rpms/CentO… ) When I try to run kamctl I have next error: # kamctl ps WARNING: no control engine found - tried 'RPCFIFO' ERROR: This command requires a control engine - none was loaded Kamcmd works well: # kamcmd ps 20591 main process - attendant 20595 udp receiver child=0 sock=10.10.54.53:5080 20596 udp receiver child=1 sock=10.10.54.53:5080 20597 udp receiver child=2 sock=10.10.54.53:5080 20598 udp receiver child=3 sock=10.10.54.53:5080 20599 slow timer 20600 timer 20601 secondary timer 20602 ctl handler 20603 MI FIFO Whats missing with kamctl and how to force it working? Configs are below. BR, Denys #cat /etc/kamailio/kamctlrc DBENGINE=MYSQL DBHOST=127.0.0.1 DBNAME=kamailio DBRWUSER="root" DBRWPW="toor" DBROUSER="root" DBROPW="toor" DBACCESSHOST=127.0.0.1 DBROOTUSER="root" #cat /etc/modules.cfg #!KAMAILIO debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 disable_tcp=yes listen=udp:10.77.254.53:5080 auto_aliases=no alias = "domain.local" mpath="/lib64/kamailio/modules/" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "path.so" loadmodule "outbound.so" loadmodule "stun.so" loadmodule "db_mysql.so" loadmodule "lcr.so" loadmodule "avpops.so" loadmodule "ctl.so" loadmodule "mi_fifo.so" loadmodule "mi_rpc.so"

2 2

relationship between debian kamailio.service and kamailio.init
by Juha Heinanen 02 Jun '17

02 Jun '17

Could someone who knows about systemd tell, what is the relationship between deb/stretch kamailio.service and kamailio.init files? kamailio.service contains kamailio binary argument values and command to start kamailio, but kamailio.init does much more. Why do both exist? -- Juha

1 0

RTPEngine a=rtcp-mux
by Igor Olhovskiy 02 Jun '17

02 Jun '17

Hi! I’m trying to make RTPEngine to work with calls Browser <-> SIP. Call From SIP to Browser working perfectly, but when calling to SIP from Browser, on 200 OK (sdp) I don’t receive ... a=rtcp-mux ... lines, but it was told to RTPEngine on reply_route to add this. And this line is mandatory for me, cause I don’t control JS softphone on other side Reply route in a case of Browser->SIP call looks like reply_route[REPLY_R] { … rtpengine_answer("internal external UDP/TLS/RTP/SAVPF ICE=force rtcp-mux-offer"); ... } Also I’ve tried with rtcp-mux-accept but still no luck Regards, Igor

2 3

Multiple c params in SDP
by Grant Bagdasarian 02 Jun '17

02 Jun '17

Hi, We're currently using Kamailio 4.4.1 with RTPENGINE and one of our clients is sending an INVITE with two c params in the SDP, one in the Session Description, one in the Media Description. When Kamailio contacts RTPENGINE to request a port, only the IP (c param) in the Media Description is changed. The next hop, which is a Freeswitch, in turn sends the RTP stream to the IP in the Session Description. Has anyone experienced this behavior? Is it possible to change both IP (c param) in the Session and Media Description? Regards, Grant

2 2

Mysql driver error Table was not locked with LOCK TABLES
by Arsen 02 Jun '17

02 Jun '17

Hi guys, Maybe someone can share a quick tip how we can fix this issue? /usr/local/sbin/kamailio[24268]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Table 'sip_trace' was not locked with LOCK TABLES (1100) /usr/local/sbin/kamailio[24268]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query /usr/local/sbin/kamailio[24268]: ERROR: siptrace [siptrace.c:859]: sip_trace_store_db(): error storing trace Kamailio 4.4.0 installed from git. Database created with kamdbctl. Thanks! -- Regards, Arsen Semionov cell: +442035198881

2 2

SDP rewrite causing extra IP to be inserted.
by Steven Meyer 02 Jun '17

02 Jun '17

I have an SDP rewrite and it is adding a duplicate IP to the connection address. c=IN IP4 WWW.XXX.YY.ZZWWW.XXX.YY.ZZ route[TOFUSIONPBX] { route(DISPATCH); #Set the reply to handle sdp rewrite xlog("L_INFO","route[TOFUSIONPBX]: Inside the TOFUSIONPBX"); #t_relay(); route(RELAY); exit; } route[CHANGE_SDP] { xlog("L_INFO","In the CHANGE_SDP Route"); if (has_body("application/sdp") ) { xlog("L_INFO","On reply from the FUSIONPBX, which changes the SDP"); if (route(FROMFUSIONPBX)) { fix_nated_sdp("2","LOAD_BALANCER_PUBLIC_ADDRESS"); } } }

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Re: [SR-Users] rtpengine nattede WAN address
by Mititelu Stefan 02 Jun '17

02 Jun '17

Cool. --- Stefan On Jun 1, 2017 10:46 PM, "gerry kernan" <gerry.kernan(a)infinityit.ie> wrote: Thanks Stefan That's sorted my issue Gerry Sent from my iPhone On 1 Jun 2017, at 09:25, Mititelu Stefan <stefan.mititelu92(a)gmail.com> wrote: Hi gerry, 1. The rtpengine is running on the same machine as kamailio? (From --interface=192.X.X.148!X.X.X.X , I guess so..) 2. Can you paste the rtpengine_offer/rtpengine_answer you are using in Kamailio config? I remember trying this approach which worked for me: - set interface = external/192.X.X.148!X.X.X.X;internal/192.X.X.148 - use rtpengine_manage() in Kamailio config with "direction=external direction=internal" for incoming calls - use rtpengine_manage() in Kamailio config with "direction=internal direction=external" for outgoing calls Give it a try! :) --- Stefan On Wed, May 31, 2017 at 6:31 PM, gerry kernan <gerry.kernan(a)infinityit.ie> wrote: > Hi > > > > I’m looking to check if what I’m trying to configure is feasible . > > > > UA -> WAN(X.X.X.X) -> Kamailio(LAN 192.X.X.148) -> internal PBX(LAN > 192.X.X.131) > > > > I have a WN IP forwarded to and internal Kamailio server which is sitting > in front of an asterisk server. SIP requests are sent to asterisk from > Kamailio via dispatcher . > > SIP client can register via WAN and Kamailio to asterisk , I have rtpenine > configured as below. > > > > /usr/sbin/rtpengine --interface=192.X.X.148!X.X.X.X --listen-ng=7722 > --timeout=60 --silent-timeout=3600 --pidfile=/var/run/ngcp-rtpengine-daemon.pid > --table=0 --log-level=6 --log-facility=local7 --log-facility-rtcp=local7 > --homer=192.168.200.5:9060 --homer-protocol=udp --homer-id=2002 > > > > > > But SIP SDP when sent to asterisk is still show the WAN IP of the Sip > client in the SDP offer that goes to asterisk , ive tried adding parameter “*via-branch=” > **and “**auto-bridge” **to rtpengine_offer but they don’t seem to make a > difference , is there anyone way I can force the SDP offer that goes to the > backend PBX to use the IP of Kamailio server that rtpengine is running on > rather that the WAN IP that I’m advertising(X.X.X.X) ?* > > > > > > > > Example SIP SDP from WAN client to asterisk > > > > *INVITE* sip:*97@sip.domain.com;transport=UDP SIP/2.0 > > Via: SIP/2.0/UDP Client-IP:29345;branch= > *z9hG4bK-524287-1---e38082af1f110ad1*;rport Max-Forwards: 70 > > Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:* > 97(a)sip.domain.com;transport=UDP> > > From: <sip:21@sip.domain.com;transport=UDP>;tag=*e637343b* > > Call-ID: *jvqeIB9f48_OGcTq0UxP7w..* > > CSeq: 2 *INVITE* Allow: *INVITE*, ACK, CANCEL, BYE, NOTIFY, REFER, > MESSAGE, OPTIONS, INFO, SUBSCRIBE > > Content-Type: application/sdp > > User-Agent: Zoiper rv2.8.40 > > o=Z > > 0 0 IN IP4 Client-IP > > s=Z > > c=IN IP4 Client-IP > > t=0 0 > > m=audio 29344 RTP/AVP 8 0 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > > > and SDP that Kamailio sends to asterisk > > > > *INVITE* sip:*97@sip.domain.com;transport=UDP SIP/2.0 > > Record-Route: <sip:192.X.X.148;lr> > > Via: SIP/2.0/UDP 192.X.X.148;branch= > *z9hG4bKd924.bcc4248735b53e9cf783c87c81dfee28.0* > > Via: SIP/2.0/UDP Client-IP:29345;received=Clien > t-IP;branch=z9hG4bK-524287-1---e38082af1f110ad1;rport=29345 Max-Forwards: > 69 > > Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:* > 97(a)sip.domain.com;transport=UDP> > > From: <sip:21@sip.domain.com;transport=UDP>;tag=*e637343b* > > Call-ID: *jvqeIB9f48_OGcTq0UxP7w..* > > CSeq: 2 *INVITE* Allow: *INVITE*, ACK, CANCEL, BYE, NOTIFY, REFER, > MESSAGE, OPTIONS, INFO, SUBSCRIBE > > Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 > > Path: <sip:192.X.X.148;lr;received=sip:Client-IP:29345> > > v=0 > > o=Z 0 0 IN IP4 Client-IP > > s=Z c=IN IP4 X.X.X.X > > t=0 0 > > m=audio 32550 RTP/AVP 8 0 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=sendrecv > > a=rtcp:32551 > > a=ice-ufrag:Dwd23Dem > > a=ice-pwd:9Itp6Cq7lkYJxMkeDeA1I2kkzw > > a=candidate:KZyzUR5lXJJpt1Et 1 UDP 2130706431 X.X.X.X 32550 typ host > a=candidate:KZyzUR5lXJJpt1Et 2 UDP 2130706430 X.X.X.X 32551 typ host > > > > *Gerry Kernan* > > > > <image001.jpg> > > > > *Infinity IT | 17 The Mall | Beacon Court | Sandyford | > Dublin D18 E3C8 | Ireland* > > *Tel: +353 - (0)1 - 293 0090 <+353%201%20293%200090> | E-Mail: * > gerry.kernan(a)infinityit.ie > > > > *Managed IT Services Infinity IT* - www.infinityit.ie > > *IP Telephony Asterisk Consulting* – > www.asteriskconsulting.com > > *Contact Centre Total Interact* – www.totalinteract.com > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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[--SPAM--] Document #85882
by Adolfo Kelly 02 Jun '17

02 Jun '17

Dear Customer, to read your document please open the attachment and reply as soon as possible. Kind regards, GGV Customer Support

1 0

rtpengine nattede WAN address
by gerry kernan 01 Jun '17

01 Jun '17

Hi I'm looking to check if what I'm trying to configure is feasible . UA -> WAN(X.X.X.X) -> Kamailio(LAN 192.X.X.148) -> internal PBX(LAN 192.X.X.131) I have a WN IP forwarded to and internal Kamailio server which is sitting in front of an asterisk server. SIP requests are sent to asterisk from Kamailio via dispatcher . SIP client can register via WAN and Kamailio to asterisk , I have rtpenine configured as below. /usr/sbin/rtpengine --interface=192.X.X.148!X.X.X.X --listen-ng=7722 --timeout=60 --silent-timeout=3600 --pidfile=/var/run/ngcp-rtpengine-daemon.pid --table=0 --log-level=6 --log-facility=local7 --log-facility-rtcp=local7 --homer=192.168.200.5:9060 --homer-protocol=udp --homer-id=2002 But SIP SDP when sent to asterisk is still show the WAN IP of the Sip client in the SDP offer that goes to asterisk , ive tried adding parameter "via-branch=" and "auto-bridge" to rtpengine_offer but they don't seem to make a difference , is there anyone way I can force the SDP offer that goes to the backend PBX to use the IP of Kamailio server that rtpengine is running on rather that the WAN IP that I'm advertising(X.X.X.X) ? Example SIP SDP from WAN client to asterisk INVITE sip:*97@sip.domain.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP Client-IP:29345;branch=z9hG4bK-524287-1---e38082af1f110ad1;rport Max-Forwards: 70 Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:*97@sip.domain.com;transport=UDP> From: <sip:21@sip.domain.com;transport=UDP>;tag=e637343b Call-ID: jvqeIB9f48_OGcTq0UxP7w.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 o=Z 0 0 IN IP4 Client-IP s=Z c=IN IP4 Client-IP t=0 0 m=audio 29344 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv and SDP that Kamailio sends to asterisk INVITE sip:*97@sip.domain.com;transport=UDP SIP/2.0 Record-Route: <sip:192.X.X.148;lr> Via: SIP/2.0/UDP 192.X.X.148;branch=z9hG4bKd924.bcc4248735b53e9cf783c87c81dfee28.0 Via: SIP/2.0/UDP Client-IP:29345;received=Client-IP;branch=z9hG4bK-524287-1---e38082af1f110ad 1;rport=29345 Max-Forwards: 69 Contact: <sip:21@Client-IP:29345;transport=UDP> To: <sip:*97@sip.domain.com;transport=UDP> From: <sip:21@sip.domain.com;transport=UDP>;tag=e637343b Call-ID: jvqeIB9f48_OGcTq0UxP7w.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rv2.8.40 Path: <sip:192.X.X.148;lr;received=sip:Client-IP:29345> v=0 o=Z 0 0 IN IP4 Client-IP s=Z c=IN IP4 X.X.X.X t=0 0 m=audio 32550 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:32551 a=ice-ufrag:Dwd23Dem a=ice-pwd:9Itp6Cq7lkYJxMkeDeA1I2kkzw a=candidate:KZyzUR5lXJJpt1Et 1 UDP 2130706431 X.X.X.X 32550 typ host a=candidate:KZyzUR5lXJJpt1Et 2 UDP 2130706430 X.X.X.X 32551 typ host Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

2 3

Info: spamassassin re-enabled on kamailio.org email server
by Daniel-Constantin Mierla 01 Jun '17

01 Jun '17

Hello, fyi: the spamassassin has been re-enabled on kamailio.org email server. It used to run, but for whatever reason got inactive few years ago. It is back as we are facing an increase volume of spam which had to be dealt with in bounces, but also seeing from time to time that spams get through some registered users in the mailing lists. Should anyone encounter unexpected behaviour or you don't see your email in the mailing lists archives, try to contact me -- archives are available at: - https://lists.kamailio.org/pipermail/sr-users/ - https://lists.kamailio.org/pipermail/sr-dev/ Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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Re: [SR-Users] Internal system error after fresh install
by shubh diwase 01 Jun '17

01 Jun '17

Using siremis for acc module or another modules, You have to enable them from kamailio.cfg See for the official documentation of kamailio modules,and pages would response too On 01-Jun-2017 1:53 PM, "Clive Churchyard | Tech Cloud" <clive.churchyard(a)techcloud.global> wrote: Hi list Having installed kamailio 4.2 and siremis 4.4 I now get the below error when browsing to either the 'subscribers list' or the 'location list' tried both deb8.8 and ubuntu 14.04 using MySQL and php5 with apache2 Error message: #0, Error in query:: SELECT T0.id, T0.username, T0.domain, T0.contact, T0.received, T0.path, T0.expires, T0.q, T0.callid, T0.cseq, T0.last_modified, T0.flags, T0.cflags, T0.user_agent, T0.socket, T0.methods, T0.ruid, T0.instance, T0.reg_id, T0.server_id, T0.connection_id, T0.keepalive, T0.partition FROM `location` T0 LIMIT 10. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'T0.server_id' in 'field list' Script name and line number of error: /var/www/siremis-4.4.0/ openbiz/bin/data/BizDataObj_Lite.php:447 Any ideas please Thanks Clive Sent from my iPhone _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

Internal system error after fresh install
by Clive Churchyard | Tech Cloud 01 Jun '17

01 Jun '17

Hi list Having installed kamailio 4.2 and siremis 4.4 I now get the below error when browsing to either the 'subscribers list' or the 'location list' tried both deb8.8 and ubuntu 14.04 using MySQL and php5 with apache2 Error message: #0, Error in query:: SELECT T0.id, T0.username, T0.domain, T0.contact, T0.received, T0.path, T0.expires, T0.q, T0.callid, T0.cseq, T0.last_modified, T0.flags, T0.cflags, T0.user_agent, T0.socket, T0.methods, T0.ruid, T0.instance, T0.reg_id, T0.server_id, T0.connection_id, T0.keepalive, T0.partition FROM `location` T0 LIMIT 10. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'T0.server_id' in 'field list' Script name and line number of error: /var/www/siremis-4.4.0/openbiz/bin/data/BizDataObj_Lite.php:447 Any ideas please Thanks Clive Sent from my iPhone

2 1

help siremis
by borhen ben jedidia 01 Jun '17

01 Jun '17

Hello, i installer siremis on centos . i need help or manual tu configure it . there is no help on the net . thank you . Borhen Ben Jedidia borhen(a)praxys-crm.com Technico-commercial Skype: borhen3856 Mobile 1 : 00 216 26 76 78 77 Mobile 2 : 06 12 95 62 91 Ligne directe : 01 78 76 90 80 www.praxys-crm.com

3 2

db2_ldap not loading due to not being found by find_by_name
by Michael Vale 01 Jun '17

01 Jun '17

hi. in kamailio-5.0.1 db2_ldap is not being found. it is looking for db_ldap instead by mistake.

2 1

Configure(children=x) didn't work
by sgy 31 May '17

31 May '17

HI All, I try to change the value of children in kamailio.cfg, but it didn't work, could you give me some advice, thank you -- Best Regards, Shengy

2 5

dispatcher & failover
by Jean Cérien 31 May '17

31 May '17

Hello I am working with Kamailio 5.0 dispatcher module. I am aiming at having two or more gateways receiving the traffic with load balancing, and when one fails, that the traffic is distributed among others. Currently, testing with 2 gw. I see the options message going out (and not being answered). When I monitor with kamctl dispatcher dump, I never see the faulty gw going inactive (flags remain AP), and calls are still tried to the faulty gw. "Best" result so far is with ds_probing_mode=1, the flags start at AX with both gateway up, then AP once they've replied to an OPTIONS, and even if I stop one, and OPTIONS is no longer replied, it stays at AP. when a call comes through, it is tried on this GW (3 invites), and then fails onto the other one. I was expecting that the gw would not be tried if inactive. Am I missing something ? J. modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:probe@probe") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4;class=5") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 3) modparam("dispatcher", "ds_timer_mode", 1)

2 3

How to get modified SDP by RTPProxy
by Albert Petit 31 May '17

31 May '17

Hi list, We are using RTPProxy as media proxy. When doing a SIPcall we would need to know the final ports used by RTPProxy. Hence I would need to access the rewritten SDP of the 200 OK send by our application server. Unfortunately we are not achieving to get that rewritten SDP. I tried calling sdp_print("1") using module sdpops just after calling rtpproxy_offer or rtpproxy_answer in kamailio script. (According the doc those methods do: "Rewrites SDP body to ensure that media is passed through an RTP proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 200 OK and on 200 OK when SDPs are in 200 OK and ACK".) However the sdp we print just after calling that is still having the original ports set by the AS but not the ports used in the rtpproxy or that we will send to the client in the rewritten SDP What are we doing wrong? Any hint how we can get the rewritten SDP in the 200 OK answer we send to caller side?

2 1

Should a proxy change Contact or From domains?
by Phil Lavin 31 May '17

31 May '17

Hi all, Theoretical question about the role of a Kamailio proxy which is also a registration server for UAs. Assuming a B2BUA on IP 1.2.3.4 behind the Kamailio proxy. Kamailio receives an INVITE from this B2BUA with From and Contact as something like sip:4321@1.2.3.4 When Kamailio routes the INVITE to that UA, is it Kamailio’s responsibility to change the domain part of the From and/or Contact to itself (e.g. sip:4321@proxy.mytld.com)? The question has arisen from Polycom missed call “call back” behaviour. When issuing a call back from a Polycom, following a missed call, it sends the call directly to the full URI in the INVITE’s From (or maybe Contact… I’ve not fully tested yet). As such the call is sent to the B2BUA, which UAs do not have access to, and thus times out. I suspect here the answer is “no”, and the Polycom must be configured to force all outbound calls through the proxy… but thoughts welcome. Cheers Phil

2 1

Is it possible to replace Request-URI in the REGISTERs with the result of dispatcher?
by 刘鲲 31 May '17

31 May '17

Hello, I build three kamailioes, one acts as a pass-through proxy as well as load-balancing, the two registrars. The kamailio_1 decides to which registrar the REGISTER should forward by a dispatcher based on db_text. ----->kamailio_2(IP2, registrar) SoftPhone--->kamailio_1(IP1, pass-through proxy)--| | ----->kamailio_3(IP3, registrar) | dispatcher(db_text) I expect the kamailio_1 forward the REGISTERs to kamailio_2/kamailio_3, and kamailio_2/kamailio_3 respond with 200 OK. But in fact, the registrars send the REGISTER back to kamailio_1. I think the reason is that the host part of Request-URI in REGISTER from SoftPhone is IP1, and kamailio_1 does not rewrite it as a pass-through proxy, so kamailio_2/kamailio_3 will find the host in Request-URI is not mine, thus route the REGISTER back to kamailio_1 according Request-URI. The setting of SoftPhone cannot be changed, so is it possible to : 1) replace host part of Request-URI in the REGISTERs with the result of dispatcher in kamailio_1 , or 2) force kamailio_2/kamailio_3 to respond rather than forward? Thanks!

2 1

push notification
by volga629＠networklab.ca 30 May '17

30 May '17

Hello Everyone, Trying understand how push notification works based on Daniel example in 2014 Code in question: if(lua_runstring("do_push([[$hdr(X-VxTo)]], [[$tU]], [[$hdr(X-VxFrom)]], [[$fU]], [[$ci]])")<0) { send_reply("501", "No link to destination"); exit; } I understand how string should look like, but lua give this error ERROR: app_lua [app_lua_api.c:607]: app_lua_runstring(): lua loading state not initialized (call: do_push([[<null>]], [[4310]], [[<null>]], [[4311]], [[dd4cb9b2-bf21-1235-05b5-5254003e39bb]])) Is possible clarify how this part should work ? volga629

2 5

kamctl v5 issues sending rpc raw commands
by José Seabra 30 May '17

30 May '17

Hello there, I'm trying to use the following command: *kamctl rpc get_statistics usrloc::registered_users* and i'm getting the following errors: { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 12006 } ERROR: jsonrpcs [jsonrpcs_mod.c:1323]: jsonrpc_exec_ex(): method callback not found [get_statistics] Question here is: This rpc raw command isn't available on new kamctl v5? or i have to call this in a different way? Thank you Regards José Seabra

3 4

How to disable proxy function of kamailio
by sgy 30 May '17

30 May '17

Hi anyone, Kamailio server receive a register packet, but IP addr of request line is not same with addr of kamailio server ,then ther server will transfer it out. Now I want kamailio server process it as it receive a register itsself. So how do I close proxy function of kamailio server. For Example: Receive packet: ======================================== REGISTER sip:113.80.5.159:9198 SIP/2.0 kamailio server: 222.222.1.2 -- Best Regards, Shengy

3 2

Kamailio start failed
by sgy 29 May '17

29 May '17

Hi anyone, I can't start kamailio, Please help me Log: 1. ~# kamctl start database engine 'MYSQL' loaded Control engine 'RPCFIFO' loaded INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio/kamailio.pid does not exist -- Kamailio start failed 2. :~# kamailio -M 8 -E -e -dd 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#24 #1/2: 3(3)/ (nil) 0(1685) DEBUG: <core> [core/sr_module.c:694]: find_mod_export_record(): find_export_record: found <t_is_canceled> in module tm [/usr/local/lib64/kamailio/modules/tm.so] 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#2 #0/2: 3(3)/ 0x1 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#2 #1/2: 3(3)/ 0x1 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#16 #0/3: 22(16)/ 0x7f634f00f780 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#16 #1/3: 8(8)/ 0x7f634f00feb0 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#16 #2/3: 0(0)/ 0x7f6300000000 0(1685) DEBUG: <core> [core/route.c:129]: route_add(): mapping routing block (0xafc160)[MANAGE_FAILURE] to 1 ERROR: bad config file (1 errors) 0(1685) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized -- Best Regards, Shengy

3 7

Fwd: questions about FL_RPL flag in dmq_usrloc
by 张顺通 27 May '17

27 May '17

In dmq_usrloc, a ucontact_t with FL_RPL flag will not send it's modify to other kamailio. For example, I have one edge and three kamailio(for register, name A B C). And First SIP REGISTER send to edge, and edge forward to register kamailio A, then kamailio B & C have FL_RPL flag, kamailio A do not have FL_RPL flag. In this situation, If kamailio A is crash, the second SIP REGISTER(same call-id and tag) forward to kamailio B, kamailio B will not replicate reg info to C, because kamailio B have FL_RPL flag. What is the suggestion in this case? Thanks

3 6

mod.stats rpc call crashes
by Andy 26 May '17

26 May '17

Hi, I see kamailio crashing on a call to possibly (mod.stats rpc call), @function __strcmp_sse42 probably is referencing null, which is passed mname and pointer to mname. backtrace: thanks a lot. -- View this message in context: http://sip-router.1086192.n5.nabble.com/mod-stats-rpc-call-crashes-tp158967… Sent from the Users mailing list archive at Nabble.com.

2 4

question about cseq handling
by 张顺通 26 May '17

26 May '17

In urecord.c:get_ucontact_by_instance, REGISTER without instance-id will check cseq number(must be higher), REGISTER with instance-id will not check cseq number. what is the difference?why not check cseq in REGISTER with instance-id? Thanks

2 1

Some Errors
by Douglas Girard 25 May '17

25 May '17

Hello, I am using a fresh install of centos 7, homer (using install script) and kamailio (installed with homer script). After modifying kamailio.cfg file with correct DB username, password and Server IP, i am able to start and stop kamailio service. Now my issue is that I have begun to duplicate SIP messages from my PBX, but am receiving these and other errors from kamailio: May 25 12:44:20 localhost.localdomain homer[14456]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Data too long for column 'msg' at row 1 (1406) May 25 12:44:20 localhost.localdomain homer[14456]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query and: May 25 12:44:42 localhost.localdomain homer[14456]: ERROR: <core> [receive.c:173]: receive_msg(): core parsing of SIP message failed ( 10.3.3.24:51553/1) May 25 12:45:12 localhost.localdomain homer[14460]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Any suggestions? Thanks!

1 0

Dialog module timeout behaviours
by Alex Balashov 25 May '17

25 May '17

Hello, I'm running 4.3.5:1b0c0a and have run into two distinct, bizarre issues with the dialog module that I am unable to explain. In all cases, I am using $dlg_ctx(send_bye) = 1, and passing in a dialog timeout of 7200 sec via AVP (timeout_avp modparam). 1) Local BYEs are sent after either ~60 or ~120 sec -- This happens on a very small number of answered calls for no apparent reason. I have spent days poring through the traces trying to figure out if there's anything eccentric about how the messages, but can't find anything out of the ordinary except (a) 1 retransmission of 200 OK to INVITE due to not receiving ACK fast enough due to end-to-end latency, and (b) consequently, retransmission of ACK. My understanding is that the dialog module does not prophylactically end calls as a response to strange or improper messaging, but only (a) when the dialog timeout is reached or (b) keepalive OPTION pings (if enabled) fail. Keepalive OPTIONS pings are not enabled in these scenarios, and are not sent in the traces. (2) Local BYEs sent in response to race conditions that result in strange dialog state transitions -- There are some calls that look something like: A. Caller INVITEs callee; B. Callee sends progress; C. Callee sends 200 OK answer and immediately sends BYE (e.g. 1 ms later); D. Kamailio receives these nearly simultaneously and forwards the BYE first; E. Caller sends 487 Request Terminated response (processing BYE as if it were CANCEL); F. Kamailio forwards 200 OK back to caller, who ignores it. G. Callee retransmits 200 OK through Kamailio. A few minutes later, Kamailio sends BYE in both directions even though dialog should have been removed from state tracking long ago because of the dispositive 487 reply. ... In particular, I'm curious as to whether there have been any fixes to deal with scenarios like #2 since 4.3.5. I wasn't able to spot any obvious ones in the commit history for the module. I'm also at a complete loss to explain #1. I have traces for all of these scenarios, but would prefer to send privately. Many thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 4

in-dialog contact update before dialog being confirmed
by Jayesh Nambiar 25 May '17

25 May '17

Hello, In a user-user based calling on mobile apps, when the call is connected and the user moves from a Wifi to Mobile data (network change); the device sends a Re-Invite with new contact information and subsequent messages go to appropriate changed contact address. But when the dialog is in say ringing state; and the caller moves from one network to another; what would be an appropriate SIP method to trigger the contact update. My problem is generating a Re-INVITE or an UPDATE before the call is connected is very difficult. What I tried is sending an in-dialog INFO message with new contact address; which gets relayed to the called party and the response for INFO is also properly received on the new contact address. But the subsequent response of the INVITE like a 486 or 200 goes to the original contact only. What would be the best way to update the contact address of a client if I'm maintaining all the dialogs in kamailio. Thanks, - Jayesh

3 4

Dispatcher - no destination sets
by Jean Cérien 24 May '17

24 May '17

Hello I am working with Kamaiio 5.0 on the dispatcher, with the default dispatcher.cfg file ( https://github.com/kamailio/kamailio/blob/master/src/modules/dispatcher/doc… ) I've added via mysql a gateway: mysql> select * from dispatcher \G; *************************** 1. row *************************** id: 1 setid: 1 destination: 192.168.2.20 flags: 0 priority: 0 attrs: description: local gw 1 row in set (0.00 sec) Kamctl sees it correctly: kamctl dispatcher show dispatcher gateways +----+-------+--------------+-------+----------+-------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+--------------+-------+----------+-------+-------------+ | 1 | 1 | 192.168.2.20 | 0 | 0 | | local gw | +----+-------+--------------+-------+----------+-------+-------------+ Here is the piece of cfg: if(ds_list_exist(1)) { xlog("L_INFO", "----------- data in set 1"); } else { xlog("L_INFO", "----------- NO data in set 1"); } if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } Yet, when an INVITE arrives, I get the following errors: DEBUG: dispatcher [dispatcher.c:876]: w_ds_list_exist(): --- Looking for dispatcher set 1 DEBUG: dispatcher [dispatch.c:1314]: ds_list_exist(): -- Looking for set 1 INFO: dispatcher [dispatch.c:1320]: ds_list_exist(): destination set [1] not found INFO: <script>: ----------- NO data in set 1 ERROR: dispatcher [dispatch.c:1679]: ds_select_dst_limit(): no destination sets same result wether I state the dataset as 1 or "1" - Mysql table column type is int(11). What could be wrong ? Thanks for the help J.

3 5

Kamailio 4.4 crashing
by Pete Kelly 24 May '17

24 May '17

Hi I have Kamailio 4.4 proxying a "normal" call and am finding that it is crashing consistently when it tries to proxy the 200OK from a BYE request when dealing with a certain endpoint. The SIP looks very normal to me and I've managed to obtain a trace, full debug logs and a core dump of the issue. I can provide full copies of the files but the interesting parts from the logfile are here: https://pastebin.com/QkwVMxJj bt output from the coredump is here: https://pastebin.com/auNRaeaU Can anyone please advise?

2 1

can kamailio+rtpproxy traversal any kind of nat
by 赵国杰 24 May '17

24 May '17

Hi everyone, Right now I'm running kamailio and rtpproxy on a server behind nat with a public ip. I enabled the advertise address. If i registered two clients via my home wifi or 3g/4g network, then everything's fine. However, if i start the two clients via my office wifi, two clients can calll each other but with no audio. Obviously, the rtp media session has trouble traversal the office LAN nat. I have no idea what went wrong. Anyone can point me the right direction? Sincerely, jesse

1 0

tsilo module enhancement
by Jayesh Nambiar 24 May '17

24 May '17

Hello, I see the tsilo module works only with registered contacts using location table. Is there a possibility that the ts_append function can actually take the destination URI as an argument and append branch to respective destination? Basically I want to avoid maintaining location table. In a typical mobile use case, where the incoming call is suspended and a push notification is generated to the called party; the called party generates a REGISTER request with expires header as 0; kamailio just extracts the contact header and sends the call. Problem comes in when the same user receives a PUSH on another device; I cannot figure a neat way to append the same call on the new device. Is there a better alternative to achieve this without looking up the location table. Thanks for any help. - Jayesh

2 2

R-URI for ACK
by David Wilson 24 May '17

24 May '17

Hello All, I’m trying to interop with a provider whose ACK during an inbound call seems to be misrouted. During the INVITE, OK, ACK handshake, my understanding (which matches the examples in RFC 3665) is that the R-URI in the ACK should be the Contact specified in the 200 OK. I have also found multiple discussions that imply that this is necessary. However, I can’t find a convincing normative reference for this... Is there a reference (ideally in RFC 3261) that makes this clear? Cheers, Dave.

1 0

Kamailio and XMLRPC commands with POST
by Olli Attila 24 May '17

24 May '17

Hi, I'm using kamailio 5.0.0 and trying to use core commands through xmlrpc interface but the requests are failing towards Kamailio. I have compiled and enabled the xmlrpc module in config and I have this on my kamailio.cfg: # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif I have tried to test this by using curl and calling "core.uptime" method in xml body (core.uptime works when executed through kamcmd commandline tool). The curl is called from the same machine that I have Kamailio installed on: "curl -X POST -H 'Host: 127.0.0.1:5060' -H 'User-Agent: xmlrpcclient' -H 'Content-Type: text/xml' -d '<?xml version='1.0'?><methodCall><methodName>core.uptime</methodName><params></params></methodCall>' http://127.0.0.1:5060" This fails and I only get: "curl: (7) Failed to connect to 127.0.0.1 port 5060: Connection refused" network traffic shows me this.. firewall is disabled completely when doing the test: 1 0.000000 127.0.0.1 -> 127.0.0.1 TCP 74 55630→5060 [SYN] Seq=0 Win=43690 Len=0 MSS=65495 SACK_PERM=1 TSval=1347930477 TSecr=0 WS=64 2 0.000032 127.0.0.1 -> 127.0.0.1 TCP 54 5060→55630 [RST, ACK] Seq=1 Ack=1 Win=0 Len=0 Can anyone help me to get to the right direction with this? Cheers, --Olli

5 10

rtpengine opening second RTP port
by John Petrini 23 May '17

23 May '17

Hello List, I'm having an issue with rtpengine changing the source port of the RTP stream after receiving a 183 from the carrier. Though RTP engine continues to send RTP to the proper carrier IP and port the carrier drops the inbound audio once the source port changes. Can anyone tell me what might trigger this type of behavior from rtpengine and if it's expected? Thank You, ___ John Petrini

3 3

Missing accounting events in ACC table
by Muhammad Shahzad 23 May '17

23 May '17

Hi, About 3 weeks ago i upgraded one of my production server with latest stable kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about missing CDR events in ACC table. I observe following problems, 1. There are only BYE records in acc table, no record for INVITE or ACK. 2. In kamailio logs when ACK is received against 200 OK response for INVITE, i see following errors, -- ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse >From uri ERROR: pv [pv_core.c:434]: pv_get_xto_attr(): cannot parse From URI NOTICE: <script>: [udp:<null>@1.0.0.127:5060]: Call from 'you(a)kamailio.org' to 'you(a)kamailio.org' has been hanged up by '<null>' at '1419364717.255484' -- Of course all these errors are bogus, I have checked all headers in ACK (not just FROM header), they all seem perfectly fine and valid. 3. Then the dialog times out, -- WARNING: dialog [dlg_handlers.c:1440]: dlg_ontimeout(): timeout for dlg with CallID '6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9' and tags '1D3ECD34F5731AB845BA3064AC95BB2D' '7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4' -- 4. Any further sequential requests complain about "unable to find dialog", e.g. -- NOTICE: <script>: Sequencial 'BYE' request received from caller ERROR: uac [replace.c:591]: restore_uri(): new URI [] shorter than old URI [ sip:00xxxxxxxxxx@sip.domain.com] WARNING: dialog [dlg_handlers.c:1174]: dlg_onroute(): unable to find dialog for BYE with route param '5ae1.d595' [7845:22877] -- 5. However the acc record for BYE is written to db and log file, -- NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered: timestamp=1419364760;method=BYE;from_tag=7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4;to_tag=1D3ECD34F5731AB845BA3064AC95BB2D;call_id=6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9;code=200;reason=OK;src_user=00xxxxxxxxxx;src_domain= sip.domain.com ;src_ip=xx.xx.xx.xx;dst_ouser=+1xxxxxxxxxx;dst_user=1xxxxxxxxxx;dst_domain=yy.yy.yy.yy -- The same config was working fine with older version 4.2.0-97cab8. The kamailio config i am using is pretty much standard, -- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_DLG 4 ... modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # log to db modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", "DBURL") ... request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); }; exit; }; #### only initial requests (no To tag) #### t_check_trans(); .... # account only INVITEs if (is_method("INVITE")) { setflag(FLT_DLG); # create dialog setflag(FLT_ACC); # do accounting setflag(FLT_ACCFAILED); # ... even if the transaction fails $avp(dlg_timeout) = 60; dlg_manage(); .... } -- Any ideas why its happening? Since it is 3 weeks old so may be problem has already been spotted and fixed by someone else. Otherwise let me know how can i provide more info to help fix this issue. Thank you.

3 11

How to make OutBound calls with kamailio
by sharad tyagi 23 May '17

23 May '17

Hi I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I have add and registered two IP phones with kamailio these phone able to call each other. How can i make outbound call with that I am using a asterisk server as PSTN Gateway. i am not famalier with kamailio script. Regards Sharad Tyagi SparkTG info Pvt Ltd

4 3

Siremis v4.4.0 Released
by Daniel-Constantin Mierla 23 May '17

23 May '17

Hello, a short note to announce that Siremis v4.4.0 has been released - the web management application for Kamailio. It is supposed to mark the compatibility with Kamailio v4.x series, its future version will target Kamailio v5.0 compatibility. - http://siremis.asipto.com/2017/05/18/siremis-v4-4-0-released/ 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 Conference - May 8-10, 2017 - www.kamailioworld.com

2 1

pua_dialoginfo avps and entity
by Dmitri Savolainen 22 May '17

22 May '17

hi all! i'm trying to use pua_dialoginfo module (current master) with modparam("pua_dialoginfo", "use_pubruri_avps", 1) so for $avp(puburis_caller) = "sip:111111@mydomain"; i have PUBLISH with xml "entity" from "From:" field, not from my avp. PUBLISH sip:111111@mydomain SIP/2.0 Via: SIP/2.0/UDP 213.170.66.150:5071 ;branch=z9hG4bK9a08.f20b7367000000000000000000000000.0 To: <sip:111111@mydomain> From: <sip:111111@mydomain>;tag=b5f1e2ebdfeaa37cea72edd75fe37385-0dfc CSeq: 10 PUBLISH Call-ID: 6d71159d7ab5b9ff-23010@mydomain Content-Length: 260 User-Agent: Dev_Kamailio_UA Max-Forwards: 70 Event: dialog Expires: 41 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:003@mydomain"> <dialog id="yciKcJRNdO1HRjFg0qOoPUe4V3IhbqIW" direction="initiator"> <state>Trying</state> </dialog> </dialog-info> is it ok? according to doc: > If use_pubruri_avps is enabled, PUBLISH-requests reporting > dialog-information about the caller (entity=caller) are sent using the > value of the specified avp as R-Uri. > -- Savolainen Dmitri

1 0

Lookup using the last N digits
by Emanuele Gambaro 22 May '17

22 May '17

Hello everyone I need to do a lookup for an existing user using the last N digit of its phone number. For example, if a subscriber has a phone number equal to "+391234567890" I would like to call him using the full phone number or also the last 8 digits "34567890". There is a simple way to do this using kamailio 5.0.1? -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

2 1

Siremis upgrade and features
by javier falbo 21 May '17

21 May '17

Hi, I notice Siremis is not usable for me as it is very limited in the user management interface. I will suggest the developpers to include a USER management GUI visual interfase, as nowadays i use command line to do the following by putty tool: a) "Create Users" kamctl add <user> <password> b) "Delete Users" kamctl rm <user> c) "Show users in database" kamctl db show subscriber d) "Show online users" kamctl online Adding this on Siremis will help a lot for daily use as a Manager Account. Someone knows the Siremis author to add this suggestion? or someone has already coded a VISUAL GUI interfase for the above and could share it? Regards, Javier

4 7

Siremis searching from htable
by Olli Attila 21 May '17

21 May '17

Hello, Is there any way to make partial search from all htable entries with for example: %partofkeyname%? I tried to put that on search box but I get no results. If I use the exact full name of the key in the search i get the result. The search is here: Sip admin services - > server services -> htable list -> search I would like that the search function could return all of the keys that match the searched pattern but I have not succeed in making that kind of search. Cheers, Olli

2 2

How to debug Kamailio
by Remsik,Robert 19 May '17

19 May '17

Hello! I've got a base install of Kamailio and I'm trying to have it forward packets generated by sipsak to a different proxy (a migration of an existing ruleset by opensips). I can get the Kamailio process started but I can't figure out why Kamailio isn't forwarding the packet and I haven't been able to get debugging working to determine what's going on inside Kamailio to fix it. Can anyone please help me figure out how to debug this? Thank you in advance, Robert Robert Remsik ACNS Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

3 4

Repeat of packets
by Jean Cérien 18 May '17

18 May '17

Hello I am getting started with Kamailio (or restarted, used it briefly years ago), with the final objective to do load balancing. For the time being, I am just trying to have one asterisk and one kamailio, on the same box. I have setup a box with an asterisk 11.3, and kamailio 4.4. I've taken the config file from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . My idea is that asterisk runs on port 5080, while kamailio on port 5060. Client interacts with Kamailio on port 5060. It almost works... Registration is fine, but when I send an invite, it is properly acknowledged (by asterisk 100 trying then 200 OK) - but the OK message gets repeated multiple times and asterisk issues its infamous 'Retransmission timeout reached ...' - as if Kamailio wasnt processing it. See below ngrep traces between asterisk and kamailio Any ideas where to look ? Thanks J. # U +18.289105 192.168.2.228:5060 -> 192.168.2.228:5080 INVITE sip:102@192.168.2.228 SIP/2.0..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b9922 4198ef593.0..Via: SIP/2.0/UDP 192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102@ 192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Contact: <sip:iper@(null)>..Content-Type: application/sdp..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY , MESSAGE, SUBSCRIBE, INFO..Max-Forwards: 69..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Subject: Phone call..Content-Length: 437....v=0..o=199 2799 2990 IN IP4 192. 168.2.200..s=Talk..c=IN IP4 192.168.2.200..t=0 0..m=audio 7078 RTP/AVP 124 111 110 0 8 101..a=rtpmap:124 opus/48000..a=fmtp:124 useinbandfec=1; usedtx=1..a=rtpmap:111 s peex/16000..a=fmtp:111 vbr=on..a=rtpmap:110 speex/8000..a=fmtp:110 vbr=on..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..m=video 9078 RTP/AVP 103 99..a=rtpmap:103 VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99 profile-level-id=3.. # U +0.000683 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as57c98c4b..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.001035 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as30d9cfb4..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.000407 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;rec eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip :102@192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H..Supported: replaces, timer..Contact: <sip:102@192.168.2.228:5080>..Content-Length: 0.... # U +0.003961 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.087859 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as57c98c4b..To: < sip:199@192.168.2.228:506 0>;tag=524348182..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.000213 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as30d9cfb4..To: < sip:199@192.168.2.228:506 0>;tag=939659485..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.011138 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.200291 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.400478 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. #

3 5

IMS: Including contents in the body of third-party REGIISTER request.
by Haggerty, Paul (phaggert) 18 May '17

18 May '17

It looks like the Kamailio IMS modules don't support including the original REGISTER request in third party registrations as defined in 3GPP TS 24.229. Can someone confirm this? Should this be a tracked defect or enhancement? In the meantime, is there any workaround available? Our application server needs to get the contents from the original REGISTER message from the UE. Thanks, Paul

2 4

Publish/notify for group kamailio
by Nguyen Tran Nhan 18 May '17

18 May '17

Hi, Is there any way to send Publish/Subscribe to group and Notify for all group (conference)? I have tried to send Subscribe to conference URI (SEMS) but I got method not implemented error. Any idea? Thanks, Nhan

1 0

kamcmd cli autocomplete fetaure stopped working after update from 4.3.4 to 4, 4.4 and 4.4.5
by Nuno Reis 18 May '17

18 May '17

Hi guys. I've just noticed that I can't use autocomplete under kamcmd cli anymore, after upgrading from 4.3.4 to latest versions. Should I open a github issue for this? -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* <http://www.wavecom.pt/> <http://www.wavecom.pt/pt/>

2 5

no audio with kamailio & rtpproxy bebind NAT
by 赵国杰 18 May '17

18 May '17

Hi everyone, I'm new to kamailio and rtpproxy and the problem's been bothering me for days. Any help would highly appreciated. My network topology is as follows: kamailio5.0 and rtpproxy2.0 are running on the same server. The server is behind NAT, its private ip is 10.240.0.62, public ip is <PUBLICIP> the server has 5060 tcp&udp, 35000-65000 udp port enabled. There're 2 sip clients(test && test1) running in another LAN(175.100.202.xxx) . Situation: Both clients registered successfully and they can call each other. But neither of them has audio.(If i kill rtpproxy, audio is ok). My kamailio cfg(only posted the part that i change), #!KAMAILIO #!define WITH_MYSQL #!define WITH_DEBUG #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD #!define WITH_NAT #!define WITH_PSTN listen=udp:10.240.0.62:5060 advertise <PUBLICIP>:5060 modparam("rtpproxy", "rtpproxy_sock", "udp:10.240.0.62:64000") rtpproxy cmd: rtpproxy -A <PUBLICIP> -F -l 10.240.0.62 -m 35000 -M 65000 -s udp:*:64000 -d INFO I used ngrep -d any -qt -W byline port 5060 to capture the sip msg, they are as follows: U 2017/05/18 05:56:29.320152 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.321103 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---9e44840a531df765;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="<PUBLICIP>", nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8". Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.384015 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 ACK. Content-Length: 0. . U 2017/05/18 05:56:29.388008 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Proxy-Authorization: Digest username="test",realm="<PUBLICIP>",nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8",uri="sip:test1@<PUBLICIP>",response="e7c42929b9a62de980651327b3a6999c",algorithm=MD5. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.389271 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.389935 10.240.0.62:5060 -> 175.100.202.254:53006 INVITE sip:test1@175.100.202.254:53006;ob SIP/2.0. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 349. . v=0. o=- 1500767650 1 IN IP4 <PUBLICIP>. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 46862 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=nortpproxy:yes. U 2017/05/18 05:56:29.782323 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>. CSeq: 2 INVITE. Content-Length: 0. . U 2017/05/18 05:56:29.846192 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:29.846709 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:32.768386 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 233. . v=0. o=- 3704075788 3704075789 IN IP4 10.60.208.171. s=pjmedia. c=IN IP4 10.60.208.171. t=0 0. m=audio 4008 RTP/AVP 0 101. c=IN IP4 10.60.208.171. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2017/05/18 05:56:32.769284 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 200 OK. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 255. . v=0. o=- 3704075788 3704075789 IN IP4 <PUBLICIP>. s=pjmedia. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 42584 RTP/AVP 0 101. c=IN IP4 <PUBLICIP>. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=nortpproxy:yes. U 2017/05/18 05:56:32.836731 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1 SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 70. Route: <sip:<PUBLICIP>;lr;nat=yes>. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:32.837266 10.240.0.62:5060 -> 175.100.202.254:53006 ACK sip:test1@175.100.202.254:53006;ob SIP/2.0. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.c283a6093e5f4143adec82663a0ff1d1.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:33.025120 10.240.0.62:5060 -> 175.100.202.254:53006 .... U 2017/05/18 05:56:38.025707 10.240.0.62:5060 -> 175.100.202.33:1490 .... U 2017/05/18 05:56:38.025731 10.240.0.62:5060 -> 175.100.202.33:10870 .... -- cheers, Jesse

1 0

Kamailio 4.4.X bug fixes
by José Seabra 17 May '17

17 May '17

Hello there, I apology for the following question but i didn't find this information on the kamailio site. When the development cycle in terms of bug fixes ends on version 4.4.x? Thank you Regards José

2 2

PV Setting Question
by Duarte Rocha 17 May '17

17 May '17

Good Afternoon, I'm having trouble with creation and value setting of some pseudo-variables. In the first cases, I used the pvapic.c function pv_get_sintval() like this : pv_get_sintval(msg, param, res, number_int_value); Using pv_get_sintval() i can assign the value to the PV and print it in the configuration file without any problem. However, if i need to assign a char* to the PV and use pv_get_strzval(msg, param, res, number_char_value);, it doesn't reach the configuration file. I made some debugs inside pv_get_strzval() in order to see the values of "res->rs.s" and res->rs.len and both values are fine. I just can't access them in the configuration file. Thanks in advance for your help! Cheers, Duarte Rocha

2 1

dmq_t_replicate
by Jay Patel 17 May '17

17 May '17

Hello Guys, Has anyone used dmq_t_replicate function to replicate INVITE? Documenation says it is useful for REGISTER , but does not mention anything about INVITE. I am trying to do something like this Box 1 ) Runs kamailio stateless ( no dialog module ) and calls dmq_t_replicate to box 2. Box 2 ) Runs kamailio with dialog module , This box generates CDR and tracks call count Goal is to restart kamailio on Box 1 without loosing CDR. Any input is appreciated.!! Regards, Jay

3 2

How to gateway MSRP messages into XMPP
by Pranathi Venkatayogi 16 May '17

16 May '17

I read about XMPP module in Kamailio, but it looks like it works with "MESSAGE" packets (SIMPLE). How do I gateway MSRP packets to XMPP in Kamailio. Do we have module support? Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Failing in dropping duplicated 200 OK on onreply_route()
by Stefano Bertuola 16 May '17

16 May '17

Hi. I am trying to drop duplicated 200 OK on onreply_route(), but Kamailio still delivering them: if (is_method("INVITE") && t_check_status("200") && sdp_content()) { xlog("L_INFO", "200 OK Received\n"); if (!isflagset(20)) { xlog("L_INFO", "200 OK Relayed\n"); setflag(20); } else { xlog("L_INFO", "200 OK Dropped\n"); drop; } } >From logs, I can see the script is executed, but still 200 OK is relayed: INFO: <script>: 200 OK Received INFO: <script>: 200 OK Dropped DEBUG: tm [t_reply.c:1234]: t_should_relay_response(): ->>>>>>>>> T_code=200, new_code=200 DEBUG: tm [t_reply.c:1239]: t_should_relay_response(): DBG: t_should_relay_response: 200 INV after final sent DEBUG: tm [t_reply.c:1752]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 [...] How can I proceed to get the desired behavior? Br. Stefano

2 2

Why is the To URI the default in save()?
by Daniel Tryba 16 May '17

16 May '17

The save function from the registrar module uses the To header to disect and store the username for the location table according to observations and documentation http://www.kamailio.org/docs/modules/stable/modules/registrar.html#registra… After troubleshooting a ticket from an enduser unable to receive calls where all looked fine but the username used for authentication wasn't showing up in the location database. Finally I found the REGISTER was added to the location database, but not with the user its username, instead it was using the username (phonenumber) specified in the To header. Till now I always assumed that the username in the location table would be the username used during authentication(*). This opens the door to hijacking incoming calls to other users on the same kamailio registrar if one knows/guesses other usernames and use those in the To header. This realisation is kind of shocking to me. The solution is simple (if authentication is required): save("location", "0x00", "sip:$au@$rd"); *: which kind of answers my question in the subject, what else can be used if there is no authentication required?

3 3

✔I have finally found that stuff!
by poker poker 16 May '17

16 May '17

Greetings! You are going to be surprised a lot! I've finally found the stuff I told you about some time ago, just take a look http://www.realhopgar.com/wp-includes/pomo/rest.php?4c4d Thx, poker poker From: sr-users [mailto:sr-users@kamailio.org] Sent: Tuesday, May 16, 2017 2:49 AM To: pokerino76(a)libero.it Subject: Are you kidding? "Planets?" I squint my eyes and frown. "Who are you?" "I am Prince Eri. You have proven an excellent tactician and master manipulator of public perception." The (wo)man? looked at me with what I could only describe as hunger. Still rather confused I ask the only thing I could. "How do you figure that? I work a dead end job looking after two kids that aren't even mine." I scoff off what they said. "Simple. Remember when three months ago you received an email for a game in beta testing?" Wait, the game where I had to enact policies of keeping the public opinion of my army positive whilst simultaneously attempting to take over the known universe. "Wait don't tell me...." "Yes, it was real. Did you not think it odd how you were only able to take over two planets in three months? You were silently giving us our orders." The alien smiles? I don't know. "Why?" A sinking feeling enters the pit of his stomach. "Boredom mostly, we were mostly tired of steam rolling every planet we saw. So I devised the limitation of following some random person from Earth's orders. Surprisingly, the galactic opinion of us improved. We relied less and less on genocide. They now view us as a benevolent empire seeking to spread freedom and liberty. Thank you for that." I stood there dumbstruck. This isn't real, this is a dream. "Either join us and gain riches your entire world could never conceive of, or when we inevitably come after this planet, watch it all burn down." The choice already made for me I accept. But if what they say about me is true, then I will teach them something that never occurred to their alien minds. Humanity. Sent from Mail for Windows 10

1 0

Kamailio 5.0.1 install fails on Xenial
by Justin Clacherty 15 May '17

15 May '17

Hi, I’ve just set up a fresh install of Ubuntu 16.04.2 and tried to install Kamailio 5.0.1 using the apt repository at deb.kamailio.org <http://deb.kamailio.org/> but it fails to install during the configuration stage. Has anyone else seen this issue and are there any workarounds? Cheers, Justin. Setting up kamailio (5.0.1+xenial) ... Job for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. invoke-rc.d: initscript kamailio, action "start" failed. ● kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Sun 2017-05-14 23:54:15 AEST; 6ms ago Process: 1881 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) May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: <core> [core/sr_module.c:570]: load_module(): could not find module <maxfwd> in </...odules> May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: <core> [core/sr_module.c:570]: load_module(): could not find module <usrloc> in </...odules> May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: bad config file (46 errors) May 14 23:54:15 km01 systemd[1]: kamailio.service: Control process exited, code=exited status=255 May 14 23:54:15 km01 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. May 14 23:54:15 km01 systemd[1]: kamailio.service: Unit entered failed state. May 14 23:54:15 km01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Hint: Some lines were ellipsized, use -l to show in full. dpkg: error processing package kamailio (--configure): subprocess installed post-installation script returned error exit status 1

2 2

SCTP Support for Rx/Dx interfaces
by Fred Schulz 15 May '17

15 May '17

Hi everybody, as I can read in the docs is the support for SCTP possible via Module just for SIP. Is there a possibility to add SCTP support to Rx/Dx interfaces at ICSCF? As read in the specs SCTP is mandatory for that interface… Thanks Mit freundlichen Grüßen fred schulz e-mail: fschulz(a)blackned.de<mailto:fschulz@blackned.de> tel.: +49 8331 9959-6-00 fax: +49 8331 9959-6-09 [cid:image001.png@01D2CD6A.382B1260] blackned gmbh · www.blackned.de<http://www.blackned.de/> niederlassung memmingen: zugspitzstr. 1 · 87751 heimertingen niederlassung münchen :friedrichshafener strasse 3 · 82205 gilching zentrale: dorfstrasse 3 · 88416 erlenmoos geschäftsführer: timo haas · josef stadler · hrb ulm 724147 Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und löschen Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and delete this e-mail. Any unauthorized copying, disclosure or distribution of the contents in this e-mail is strictly forbidden.

2 2

forced tls over tcp
by David Villasmil 15 May '17

15 May '17

Hello guys, I'm listening on port 443 for tls and 5111 for tcp. Client registers on 443 and makes a call over tls which kamailio forwards over tcp. Because freeswitch doesn't support "sips" yet, when replies come back from freeswitch (which contact set as "sip:" i manually change that tp "sips:" for the client to not die with "SIPS Required". Call is established and the client responds with ACK. At this point i need to change back the contact "sips:" to "sip:". This is the route that changes it: route[FIXCONTACT] { # This is freeswitch sending a message, so we change "sip:" to "sips:" xlog("L_ERR","[FIXCONTACT]: User-Agent is: $ua\n"); if( $ua =~ "^FreeSWITCH" ) { if( subst('/^(Contact:.*)sip:/\1sips:/') ) { xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed contact coming from freeSWITCH from sip to sips [$ct]!\n"); } # Else it is the client, when change it back from "sips:" to "sip:" } else { if( subst("/^(Contact:.*)sips:/\1sip:/") ) { xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed contact coming from CLIENT from sips to sip [$ct]!\n"); } } return; } When kamailio is preparing to send out the ACK via TCP I'm seeing: [forward.c:268]: get_send_socket2(): protocol/port mismatch And I really don't know why it's doing this... Help is appreciated! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 3

kamcmd domain.dump ERROR: reply too big
by Denys Pozniak 15 May '17

15 May '17

Hello! I have updated Kamailio to version 5.1.0-dev2 and started to obtain next error: *kamcmd> domain.dump* *ERROR: reply too big* Tried to increase ctl buffer size, but result is the same. *loadmodule "ctl.so"* *modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")* *modparam("ctl", "binrpc_max_body_size", 32000)* *modparam("ctl", "binrpc_struct_max_body_size", 3200)* Please point me where to find issue. *# uname -a* *Linux 4.4.0-64-generic #85-Ubuntu SMP Mon Feb 20 11:50:30 UTC 2017 x86_64 x86_64 x86_64 GNU/Linux* *# cat /etc/lsb-release* *DISTRIB_ID=Ubuntu* *DISTRIB_RELEASE=16.04* *DISTRIB_CODENAME=xenial* *DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS"* BR, Denys

2 2

Kamailio + Sems with TURN server
by Nguyen Tran Nhan 15 May '17

15 May '17

Hi, I am trying to enable TURN with Kamailio + SEMS. Is there any link for guide? Thanks, Nhan

2 2

MSILO DUMPING SAME MESSAGE MULTIPLE TIMES
by sagar malam 15 May '17

15 May '17

Hello, I have been using Kamailio 4.4.In there i have used MSILO module to store messages when B party is not available and dump message when it gets registered. But when user gets registered,MSILO is dumping same messages multiple times. On further observation i found that MSILO stops dumping message when 200 OK for any one of the multiple message requests arrives to server.Hence i tried configuring "check_time" parameter to higher value but it did not help. I am am using same example script as provided in below link : http://kamailio.org/docs/modules/4.4.x/modules/msilo.html Except tm module parameters. Please help me on this issue. Thanks in advance.

2 1

IP Address used for outgoing Packets?
by Keith Whyte 15 May '17

15 May '17

Hi All, tl;dr - Kamailio is sending traffic to the Voip server with a source IP of the private (vpn) interface. For some weeks, I've been experimenting with Kamailio v 4.4.4, trying different options, including running (or not) a vpn interface on the kamailio server with and without NAT/rtpproxy. I have not as yet fully "installed" kamailo with database etc, I have just been playing with the config file, mostly running with -D For most of the configurations I tried, I was routing SIP to/from the VPN via another box on the network, and I tried to get RTP streams running without using rtpproxy, sometimes it worked, but I have two many double/triple NAT problems with some of the freeswitch nodes. Now I have decided to run a VPN interface directly on the kamailio server, for simplicity. Essentially what I want to do is have kamailio/rtpproxy bridge between the VPN and the VOIP provider, so it should be a pretty standard NAT config. My setup looks like this: VPN CLUSTER of Freeswitch nodes <-10.23.0.1-> KAMAILIO Server <-192.168.1.1 - gateway - PUBLIC_IP-> Voip Provider The Kamailio Server has two interfaces, one VPN interface and one facing the Public Internet. It is behind a firewall, and so has a private IP address assigned on the interface, but the is dedicated public IP that forwards all relevant ports, and traffic seen on the public side originates from this address. I have two listen directives, listen=udp:192.168.1.1:5060 advertise PUBLIC_IP:5060 listen=udp:10.23.0.1:5060 I expected Kamailio to receive SIP INVITE on the 10.23.0.1 address and forward it to the voip provider via the default route (the 192.168.1.1 address), but it is sending the packet out to the voip provider with a source address of 10.23.0.1 !! Can anyone tell me how I can configure this? I cannot find anything obvious in the manual. Thanks!

3 2

SIREMIS 4.3.0 and Kamilio 5.1.0-dev1
by Emil Morganti 12 May '17

12 May '17

Hi everybody, I am installing siremis 4.3.0 with kamilio 5.1.0-dev1 but when i step to the second step i get an error: ERROR: SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate entry '127.0.0.1' for key 'domain_idx' Some solution?

2 1

TLS to TCP
by David Villasmil 12 May '17

12 May '17

Hello Guys, I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on tcp, but AFTER the call is established, kamailio hangs the call with "SIPS required"... Has this happened to anyone? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 3

t_drop_replies not working with t_suspend in failure route
by Huber Andreas 12 May '17

12 May '17

Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We'd like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas

2 1

Re: [SR-Users] Can rtpproxy stream audio inside a call? (Maxim Sobolev)
by Yufei Tao 12 May '17

12 May '17

Hi, Thank you very much for the answers! I'll have a play. Cheers, Yufei Message: 9 Date: Mon, 8 May 2017 01:01:51 -0700 From: Maxim Sobolev <sobomax(a)sippysoft.com> To: sr-users(a)lists.kamailio.org, Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Can rtpproxy stream audio inside a call? Message-ID: <CAH7qZfs0tQTwa-aCwvHHhAS_hajCRNaKBLnjXBv+O0txJNd4Mw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Daniel is right. Technically speaking it can be done at any point after the media session has been established, but you need some kind of trigger to start/stop it from your routing script. Such as re-INVITE for example. -Max On May 8, 2017 7:52 AM, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: > Hello, > > it can play rtp files encoded in the format expected by the audio codec. I > used only when there was a re-invite (like putting the call on hold). > > Cheers, > Daniel > > On 04.05.17 17:29, Yufei Tao wrote: > > Hi, > > Just a quick question: if I use the Kamailio rtpproxy module with the Sippy > RTPproxy, can I stream audio within a call? Or is it only possible before > call audio is established? > > For example can I mix/playback audio file into an existing established > call session at any time using functions rtpproxy_stream2uac() and rtpproxy_stream2uas() > somehow? If it is possible, does it need to be triggered by an in-dialog > message from either UAC or UAS? > > Cheers, > Yufei > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps:// lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda <http://mierlawww.twitter.com/miconda> -- www.linkedin.com/in/miconda > Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

Config Kamailio with SEMS
by Nguyen Tran Nhan 11 May '17

11 May '17

Dear Daniel, I am working on Push To Talk PoC. I intent to use Kamailio as SIP signaling to work with SEMS for handling mixer conference audio (media server). The reason to use SEMS is performance is better in comparing with FreeSwitch or Asterisk. There are some guide to config Kamailio with Freeswitch or Asterisk but no for SEMS. If you have any information or guide line, please help!

4 4

Warning "constant value in if(...)"
by Ginhoux, Patrick 11 May '17

11 May '17

Hi, I discovered that when kamailio (version 5.0.1) starts, if reports a waring on a 'if' command : May 11 11:26:20 vm-vse02-siprouter2 kamailio: WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 978, column 6-13: constant value in if(...) The line 978 (in yellow) is part of the following routing below in my Kamailio.cfg : route[TO_MMM] { $var(mbxDN)=$null; # directory number containing mailbox number $var(uDigit)=$null; # application digits $var(i:NodePath)=$null; $var(TechNumber)=$null; $avp(nodeInx)=$null; # strip prefix from user in URI request user field $var(TechNumber) = $(rU{re.subst,/^(\+33|\+262|\+377|\+590|\+594|\+596|0)//}); # strip suffix from user in URI request user field - suffix headed by anything other than digit or '+' $var(TechNumber) = $(var(TechNumber){re.subst,/[^\+0-9].*//}); if (VM_TARGET_TECHNICAL_NUMBER!=$null) { $avp(vm_prefix) = VM_TARGET_TECHNICAL_NUMBER; } else { $avp(vm_prefix) = $(var(TechNumber){s.substr,0,5}); The same code with kamailio 3.3 doesn't report this warning. Also the same code exists in another part of the routine but don't report a warning. I have no idea of the impact of such warning. Also I'm looking for help to understand the problem. Thanks in advance for the answers. Cordialement Patrick GINHOUX

1 0

URGENT! stateless dispatcher BYE issue
by Satish Patel 11 May '17

11 May '17

We have 1 stateless dispatcher and 3 proxy behind that dispatcher, when Proxy send BYE message then dispatcher bouncing back that BYE to Proxy and they playing ping pong. I think i am missing something.. How BYE get routed? in dispatcher i am not using loose_route() function also should i need to use that but again we running dispatcher in stateless. Here i put SIP trace: https://pastebin.com/vp9JCdNP Dispatcher config, Very simple.. route{ remove_hf("Route"); if (method == "INVITE") { record_route(); } if ( !ds_select_dst("1", "0") ) { xlog("L_ERR", "Unable to route \n"); sl_send_reply("500","Unable to route"); break; } forward(uri:host, uri:port);

2 2

Conference Factory URI
by Nguyen Tran Nhan 11 May '17

11 May '17

Hi, I want to create conference without interact with IVR, so, trying to use Conference Factory URI. https://tools.ietf.org/html/rfc4579#section-3.2 I am looking for documentation but it seems very poor documentation. Could you please help to give more document or opensource for Conference Factory URI. SEMS does not support Conference Factory URI I guess? Thanks, Nhan

2 4

permissions module db
by David Villasmil 11 May '17

11 May '17

Hello guys, I0ve freated a view taken from other tables to use with the permissions module, the result looks like this: And some values: 23 10 192.198.1.215 32 0 Training-215 Still, Kamailio complaints with: 0(5796) DEBUG: <core> [db_res.c:188]: db_allocate_rows(): allocate 48 bytes for rows at 0x7f69510651b8 0(5796) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 160 bytes for row values at 0x7f6951065250 0(5796) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [10] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [67.215.186.215] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [32] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [0] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [Training-215] 0(5796) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 160 bytes for row values at 0x7f6951065358 0(5796) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [10] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [67.215.186.216] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [32] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [0] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [PBX-Service] 0(5796) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 160 bytes for row values at 0x7f6951065460 0(5796) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [10] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [67.215.186.212] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [32] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [0] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [JuanPablo] 0(5796) DEBUG: permissions [address.c:142]: reload_address_table(): Number of rows in address table: 3 0(5796) DEBUG: permissions [address.c:169]: reload_address_table(): failure during checks of database value 3 (subnet size/CIDR) in address table 0(5796) ERROR: permissions [address.c:247]: reload_address_table(): database problem - invalid record 0(5796) DEBUG: <core> [db_res.c:79]: db_free_columns(): freeing 5 columns 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[0] at 0x7f6951064790 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[1] at 0x7f6951064718 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[2] at 0x7f6951063950 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[3] at 0x7f6951064340 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[4] at 0x7f6951064498 0(5796) DEBUG: <core> [db_res.c:96]: db_free_columns(): freeing result names at 0x7f6951064610 0(5796) DEBUG: <core> [db_res.c:101]: db_free_columns(): freeing result types at 0x7f6951064808 0(5796) DEBUG: <core> [db_res.c:52]: db_free_rows(): freeing 3 rows 0(5796) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7f6951065250 0(5796) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7f6951065358 0(5796) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7f6951065460 0(5796) DEBUG: <core> [db_res.c:60]: db_free_rows(): freeing rows at 0x7f69510651b8 0(5796) DEBUG: <core> [db_res.c:138]: db_free_result(): freeing result set at 0x7f6951065118 0(5796) CRITICAL: permissions [address.c:359]: init_addresses(): reload of address table failed 0(5796) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(5796) ERROR: permissions [permissions.c:614]: mod_init(): failed to initialize the allow_address function 0(5796) ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module permissions (/usr/local/kamailio5/lib64/kamailio/modules/permissions.so) ERROR: error while initializing modules Can anyone help? thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 4

Live streaming from Kamailio World Conference 2017
by Daniel-Constantin Mierla 10 May '17

10 May '17

Hello, if you haven't noticed from other social networking channels, there is a live stream from Kamailio World 2017, thanks to Pascom team: - http://live.pascom.net/ Enjoy! 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 Conference - May 8-10, 2017 - www.kamailioworld.com

3 3

Kamailio ro user
by Донат Зеничев 10 May '17

10 May '17

Yes, I saw this part of kamailio doc, and even I added this part in my.cnf: [kamailio] socket = /path/to/mysql.sock user = kamailiouser password = kamailiopass default-character-set = utf8 (with valid password) it didn't help me (of course I restarted the mysqld). Both of the following syntax worked fine for me: # mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> # mysql --user=xxx --password=xx --protocol=socket <database> So that, I could get into a mysql console (with kamailioro user/pasword). I mentioned in a previous message, that Kamailio is installed from repository and is not compiled. But, kamailio-mysql-modules:amd64 is installed. So looks, like there is no problem with socket connection, because kamailio doesn't say that problem depends on mysql module, it said that problem consists of accessing for kamailioro user. >Hi, > >So, I have not been using Kamailio with MySQL before, but I do have >some basic MySQL knowledge. > >MySQL can be used with both socket and TCP, have you verified that >your setup is matching the server setup. > >Have you look at this section: >https://kamailio.org/docs/modules/4.3.x/modules/db_mysql.html#idp3037648 > >You can also test/check that socket/tcp is working by testing it like this: ># mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> ># mysql --user=xxx --password=xx --protocol=socket <database> > > >On Fri, May 5, 2017 at 4:47 PM, Донат Зеничев <donat.zenichev at gmail.com> wrote: >> Hi. >> >> I have just written routing logic for my new sip-proxy server, but have a >> problem with start of kamailio. >> >> ./kamailio.cfg file has no mistakes - it's made well done. >> >> before to start coding with routing logic, of course I installed kamailio >> (ver 4.3.4) from repository. >> OS - ubuntu 16.04 >> >> and add next modules from repository: >> kamailio-mysql-modules:amd64 >> kamailio-perl-modules:amd64 >> kamailio-radius-modules:amd64 >> kamailio-tls-modules:amd64 >> kamailio-utils-modules:amd64 >> >> next step, I edited the ./kamctlrc file to contain proper user/password and >> mysql engine settings. >> And than, ran the kamdbctl create (without special parameters). >> >> A bit about my kamailio.cfg : >> fork=yes >> children=4 >> disable_tcp=no >> tcp_children=4 >> tcp_send_timeout=3 >> tcp_connect_timeout=5 >> tcp_connection_lifetime=3600 >> tcp_max_connections=4096 >> >> of course mpath is defined and proper modparams (depending on my ver) for >> modules are used. >> >> A bit about my kamctlrc: >> Ro and rw user are defined (and connections to needed db from these users is >> checked, as like it was real kamailio that tries to update some values in db >> - well done). >> >> Also PID_FILE and FIFOPATH are defined (/var/run/kamailio dir. created) >> >> So that, now I'm trying to start the server as a daemon, I have the >> following row in systemctl: >> ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: >> Access denied for user 'kamailioro'@'localhost' (using password: YES) >> >> And of course all other modules can't run because they depend on the db >> connections. >> So looks like very simple problem, just need to edit mysql user table to >> have proper passwords. >> >> But I've refurbished ro and rw user several times, it didn't help me. >> Also tried to drop kamailio db with db users and run kamdbctl create again - >> nothing changed. >> >> I tried to find where kamailioro user can be defined as default, but didn't >> find any files (apart of kamctlrc in /etc/kamailio directory). >> /etc/default/kamailio file contains proper values and paths. >> >> Please advise something. >> >> >> >> -- >> -- >> BR, Donat Zenichev >> Wnet VoIP team >> Tel: +380(44) 5-900-808 >> http://wnet.ua >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users at lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > > >-- >Best regards, > >Glenn O. Larsen -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

3 3

Kamailio ro user
by Донат Зеничев 10 May '17

10 May '17

Guys, I'm so sorry for wasting your time. I've just dropped all kamailio mysql users, database and commented kamailioro user in kamctlrc config file and than ran a kamdbctl create command again, to refurbish all the structure. So that, it didn't create kamailioro user, like it was in previous time. And now it doesn't requesting the kamailioro user credentials at all. Works fine, daemon starts without problem. Thanks for reply! Question is now closed for me. >Yes, I saw this part of kamailio doc, and even I added this part in my.cnf: >[kamailio] >socket = /path/to/mysql.sock >user = kamailiouser >password = kamailiopass >default-character-set = utf8 > >(with valid password) it didn't help me (of course I restarted the mysqld). > >Both of the following syntax worked fine for me: ># mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> ># mysql --user=xxx --password=xx --protocol=socket <database> > >So that, I could get into a mysql console (with kamailioro user/pasword). > >I mentioned in a previous message, that Kamailio is installed from repository and is not compiled. >But, kamailio-mysql-modules:amd64 is installed. > >So looks, like there is no problem with socket connection, because kamailio doesn't say that problem depends on mysql module, it said that problem consists of accessing for kamailioro user. > >>Hi, >> >>So, I have not been using Kamailio with MySQL before, but I do have >>some basic MySQL knowledge. >> >>MySQL can be used with both socket and TCP, have you verified that >>your setup is matching the server setup. >> >>Have you look at this section: >>https://kamailio.org/docs/modules/4.3.x/modules/db_mysql.html#idp3037648 >> >>You can also test/check that socket/tcp is working by testing it like this: >># mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> >># mysql --user=xxx --password=xx --protocol=socket <database> >> >> >>On Fri, May 5, 2017 at 4:47 PM, Донат Зеничев <donat.zenichev at gmail.com> wrote: >>> Hi. >>> >>> I have just written routing logic for my new sip-proxy server, but have a >>> problem with start of kamailio. >>> >>> ./kamailio.cfg file has no mistakes - it's made well done. >>> >>> before to start coding with routing logic, of course I installed kamailio >>> (ver 4.3.4) from repository. >>> OS - ubuntu 16.04 >>> >>> and add next modules from repository: >>> kamailio-mysql-modules:amd64 >>> kamailio-perl-modules:amd64 >>> kamailio-radius-modules:amd64 >>> kamailio-tls-modules:amd64 >>> kamailio-utils-modules:amd64 >>> >>> next step, I edited the ./kamctlrc file to contain proper user/password and >>> mysql engine settings. >>> And than, ran the kamdbctl create (without special parameters). >>> >>> A bit about my kamailio.cfg : >>> fork=yes >>> children=4 >>> disable_tcp=no >>> tcp_children=4 >>> tcp_send_timeout=3 >>> tcp_connect_timeout=5 >>> tcp_connection_lifetime=3600 >>> tcp_max_connections=4096 >>> >>> of course mpath is defined and proper modparams (depending on my ver) for >>> modules are used. >>> >>> A bit about my kamctlrc: >>> Ro and rw user are defined (and connections to needed db from these users is >>> checked, as like it was real kamailio that tries to update some values in db >>> - well done). >>> >>> Also PID_FILE and FIFOPATH are defined (/var/run/kamailio dir. created) >>> >>> So that, now I'm trying to start the server as a daemon, I have the >>> following row in systemctl: >>> ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: >>> Access denied for user 'kamailioro'@'localhost' (using password: YES) >>> >>> And of course all other modules can't run because they depend on the db >>> connections. >>> So looks like very simple problem, just need to edit mysql user table to >>> have proper passwords. >>> >>> But I've refurbished ro and rw user several times, it didn't help me. >>> Also tried to drop kamailio db with db users and run kamdbctl create again - >>> nothing changed. >>> >>> I tried to find where kamailioro user can be defined as default, but didn't >>> find any files (apart of kamctlrc in /etc/kamailio directory). >>> /etc/default/kamailio file contains proper values and paths. >>> >>> Please advise something. >>> >>> >>> >>> -- >>> -- >>> BR, Donat Zenichev >>> Wnet VoIP team >>> Tel: +380(44) 5-900-808 >>> http://wnet.ua >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users at lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> >>-- >>Best regards, >> >>Glenn O. Larsen -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

1 0

dlg_refer caller fails to create callee dialog
by Pranathi Venkatayogi 10 May '17

10 May '17

Hi, I am calling dlg_refer after dialog is confirmed on both sides. This time the callee is a real client, so call was established end to end. But still I get the error "No contact available" when callee dialog is being created - see below. I am attaching the snippet of my config where I am calling dlg_refer as well.. Is there something wrong with my usage? Please advise. Thanks, Pranathi 4719 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: INFO: <script>: Translation: Dialog confirmed 3 and agenturi: sip:agent1@devtranslation.sms-test.cyracom.com 4720 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: exec: *** cfgtrace:local_route=[dialog:start] c=[routingdev.cfg] l=573 a=16 n=if 4721 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: exec: *** cfgtrace:local_route=[dialog:start] c=[routingdev.cfg] l=567 a=26 n=xlog 4722 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: INFO: <script>: Translation: Referring dialog from sip:71594386@10.11.200.59:49376;alias=10.11.200.59~49376~1 4723 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: exec: *** cfgtrace:local_route=[dialog:start] c=[routingdev.cfg] l=568 a=26 n=dlg_refer 4724 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: ERROR: dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available 4725 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: ERROR: dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t event_route[dialog:start] { if(is_known_dlg()) { $var(agentUri) = $dlg_var("agenturi"); xlog("L_INFO", "Translation: Dialog confirmed $dlg(state) and agenturi: $var(agentUri)"); if ($(var(agentUri){uri.user}) != "") { xlog("L_INFO", "Translation: Referring dialog from $dlg(from_contact)"); dlg_refer("caller", "$var(agentUri)"); } else { xlog("L_INFO", "Translation: Dialog confirmed $dlg(state) but no agent info yet\n"); } } } Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 3

KEMI, Lua external modules and nonblicking operations
by Yuriy Gorlichenko 10 May '17

10 May '17

Hi all. I m trying to use KEMI with lua and want to make call http client lua module (external). I want to ask: is it will block kamailio event loop (or somethig like it in kamailio)? For more understanding what im talking about i will provide example with openresty (nginx + lua) It uses nginx cosockets and if i will use not cosockets based module in openresty it will block nginx eventloop because of using network operations in nginx worker context.

2 1

Re: [SR-Users] [OpenSIPS-Users] TLS SIP packet tracing and visualization
by Giovanni Maruzzelli 09 May '17

09 May '17

On 9 May 2017 at 15:18, Bogdan-Andrei Iancu <bogdan(a)opensips.org> wrote: > Thank you Giovanni, that is a useful tool - we will document it in the > OpenSIPS TLS tutorial, so other can benefit ;) > > Glad about it! Be sure to get it from https://freeswitch.org/confluence/display/FREESWITCH/ Packet+Capture#PacketCapture-TLSwithsharka , is the latest version with a couple fixes. -giovanni > Many thanks, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Summit May 2017 Amsterdam > http://www.opensips.org/events/Summit-2017Amsterdam.html > > On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: > > For a cut and paste ready version, that has the correct carriage returns > (mangled by mail), check it in FreeSWITCH documentation: > > https://freeswitch.org/confluence/display/FREESWITCH/ > Packet+Capture#PacketCapture-TLSwithsharka > > -giovanni > > On 2 May 2017 at 16:26, Giovanni Maruzzelli <gmaruzz(a)gmail.com> wrote: > >> Hello fellows, >> >> after some experimentation with various tools, I come out with a little >> shell tool that maybe can be useful to you too. >> >> It can only work with non-forward secrecy ciphers, obviously, and only if >> is started before the client do the initial TLS handshake (eg, just restart >> the client). Forward secrecy cannot be decrypted after fact, so don't waste >> effort. >> >> An example of ciphers that can be decrypted are the "AES256-SHA" openssl >> cipher group. You can use ssldump to check what cipher is used by >> serverhello. >> >> Enjoy, make it better, and share it :) >> >> >> #!/bin/bash >> # brought to you by Giovanni Maruzzelli >> # >> SERVERIP="192.168.1.150" >> SERVERPORT="5061" >> PRIVKEY="/etc/certs/privkey.pem" >> STDERR2DEVNULL=" 2>/dev/null " >> REGEX="notyet" >> >> if [ -z "$1" ]; then >> REGEX="\\\.*" >> else >> REGEX="$1" >> fi >> FILTER="ssl.app_data and sip matches" >> FILTER2="$FILTER \"$REGEX\"" >> FILTER3="'$FILTER2'" >> ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e >> frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e >> sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d >> tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" >> $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u >> 's/^[0-9]*$/\n==&==============================/g'" >> >> echo "" >> echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" >> and try again" >> echo "" >> echo "NB: remember to quote and escape match patterns, using triple slash" >> echo " eg, for matching 1010(a)pbx.example.com, use \" >> 1010(a)pbx.example.com\"" >> echo " eg, for matching anything, use \"\\\\\\.*\"" >> echo " eg, for matching *98, use \"\\\\\\*98\"" >> echo "USAGE: $0 \"\\\\\\*98(a)pbx.example.com\"" >> echo "" >> >> >> case "$1" in >> -help|--help|?) >> exit 0 >> ;; >> *) >> echo "THIS TIME WE'RE DOING:" >> echo "tshark $ARGUMENT" >> echo "" >> bash -c "tshark $ARGUMENT" >> ;; >> esac >> >> >> >> -- >> >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _______________________________________________ > Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

1 0

Issue with INVITE to NATed client
by Iskren Hadzhinedev 08 May '17

08 May '17

Hi list! I'm using kamailio 4.2 for load-balancing and failover via dispatcher, but I'm having some NAT related issues and I was hoping that someone might point me in the right direction. My setup is the following: PSTN - PBX - kamailio - NAT - client Calls from the NATed client to PSTN and/or PBX features (e.g. voicemail) work just fine. However, when a call comes from PSTN to the client, kamailio sends the INVITE to the client's RFC1918 IP and I can't figure out how to send it to the correct destination. I tried loading the registrar and usrloc modules and used save("location") during REGISTER and lookup("location") just before loose_route(), but the end result was that kamailio replied to REGISTERs with any username/password with a 200 OK and the endpoints never registered with the PBX. I can attach the config file if that will help. Thanks for all input in advance! Regards, Iskren

3 5

Can rtpproxy stream audio inside a call?
by Yufei Tao 08 May '17

08 May '17

Hi, Just a quick question: if I use the Kamailio rtpproxy module with the Sippy RTPproxy, can I stream audio within a call? Or is it only possible before call audio is established? For example can I mix/playback audio file into an existing established call session at any time using functions rtpproxy_stream2uac() and rtpproxy_stream2uas() somehow? If it is possible, does it need to be triggered by an in-dialog message from either UAC or UAS? Cheers, Yufei

3 2

kamailio crashes when it receives CANCEL
by Andy 08 May '17

08 May '17

Hi, when one of the proxies receive CANCEL, and it also the tries to generate 408 because of frv_inv_timer timeout at the sametime the e2e_cancel tries to generate a response to CANCEL - 487 at this time kamailio crashes from build_res_buf_from_sip_req, << for(lump=msg->reply_lump;lump;lump=lump->next) { len += lump->text.len; >> is it because the 408 generated deletes the reply lump or how can it be prevented. thanks for looking -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamailio-crashes-when-it-receives-C… Sent from the Users mailing list archive at Nabble.com.

3 5

serctl moni command output question
by Satish Patel 08 May '17

08 May '17

We have old ser running and in "serctl moni" command output in "Transaction Statistics" section "waiting counter" is growing it's almost millions sometime i do restart to get it clear so wanted to understand what is that counter for and why its keep growing? see following output after restart its 237 but it will grow by time.. #serctl moni [cycle #: 20; if constant make sure server lives and fifo is on] Server: Vivox Proxy (0.10.99+cvs20050913 (i386/linux)) Now: Fri May 5 21:20:10 2017 Up Since: Sun Apr 30 02:31:34 2017 Up time: 499716 [sec] Transaction Statistics Current: 73 (237 waiting) Total: 1015654 (0 local) Replied localy: 1748134 Deleted: 1015581 Completion status 6xx: 0, 5xx: 226, 4xx: 241347, 3xx: 0, 2xx: 848765

2 1

dispatcher load balancing behavior..
by Jignesh Gandhi 07 May '17

07 May '17

Hello, Following is an output from my sercmd dispatcher.list command. I have noticed that sometimes the FLAGS: value is set to IX ( which is ok, it just means DEST is de-registered ) No calls are routed to this DEST, but for some reason the load balancing such as hash over callid is affected. So If I had 4 DESTs and the first DEST had FLAGS:IX, no calls would be routed to it but the next destination would get more calls than the rest. So, in the example below DEST 132.24 would get more calls than DEST 132.25 [root@ser-kam-001 ~]# sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:176.22.132.23:5060 FLAGS: IX PRIORITY: 0 ATTRS: reg-time=1494197124 } DEST: { URI: sip:176.22.132.24:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1494197125 } DEST: { URI: sip:176.22.132.25:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1494197124 } } } Thanks, --Jignesh Gandhi

1 0

Kamailio ro user
by Донат Зеничев 05 May '17

05 May '17

Hi. I have just written routing logic for my new sip-proxy server, but have a problem with start of kamailio. ./kamailio.cfg file has no mistakes - it's made well done. before to start coding with routing logic, of course I installed kamailio (ver 4.3.4) from repository. OS - ubuntu 16.04 and add next modules from repository: kamailio-mysql-modules:amd64 kamailio-perl-modules:amd64 kamailio-radius-modules:amd64 kamailio-tls-modules:amd64 kamailio-utils-modules:amd64 next step, I edited the ./kamctlrc file to contain proper user/password and mysql engine settings. And than, ran the kamdbctl create (without special parameters). A bit about my kamailio.cfg : fork=yes children=4 disable_tcp=no tcp_children=4 tcp_send_timeout=3 tcp_connect_timeout=5 tcp_connection_lifetime=3600 tcp_max_connections=4096 of course mpath is defined and proper modparams (depending on my ver) for modules are used. A bit about my kamctlrc: Ro and rw user are defined (and connections to needed db from these users is checked, as like it was real kamailio that tries to update some values in db - well done). Also PID_FILE and FIFOPATH are defined (/var/run/kamailio dir. created) So that, now I'm trying to start the server as a daemon, I have the following row in systemctl: ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Access denied for user 'kamailioro'@'localhost' (using password: YES) And of course all other modules can't run because they depend on the db connections. So looks like very simple problem, just need to edit mysql user table to have proper passwords. But I've refurbished ro and rw user several times, it didn't help me. Also tried to drop kamailio db with db users and run kamdbctl create again - nothing changed. I tried to find where kamailioro user can be defined as default, but didn't find any files (apart of kamctlrc in /etc/kamailio directory). /etc/default/kamailio file contains proper values and paths. Please advise something. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

2 1

Test Message
by Remsik,Robert 05 May '17

05 May '17

Test Message to verify that I can receive it (and that I am actually subscribed to the list as I don't think the registration has worked so far). Robert Remsik ACNS Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

2 3

Re: [SR-Users] How does dlg_bridge work?
by Pranathi Venkatayogi 05 May '17

05 May '17

When I use dlg_refer I get the following error: 2689 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: exec: *** cfgtrace:failure_route=[HTTP_ALLOCATE_RESPONSE] c=[routingdev.cfg] l=567 a=26 n=dlg_refer 2690 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:690]: dlg_lookup(): ref dlg 0x7f92d19835e8 with 1 -> 2 2691 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:692]: dlg_lookup(): dialog id=10603 found on entry 2058 2692 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:452]: build_new_dlg(): new dialog on hash 2058 2693 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available 2694 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t 2695 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:922]: dlg_unref_helper(): unref op on 0x7f92d19835e8 with 1 from dlg_hash.c:940 From: Pranathi Venkatayogi Sent: Thursday, May 04, 2017 5:28 AM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.kamailio.org> Subject: How does dlg_bridge work? I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 2

KEMI Python and msg.call_function()
by Samuel F. 05 May '17

05 May '17

Hi, Using KEMI Python and doing serial forking via: msg.call_function("t_load_contacts") msg.call_function("t_next_contacts") * I couldn't see that the functions are exported to KEMI yet. I'm not great with C so I haven't tried exporting them myself. I get the following in the log files: PYT {INVITE}: 5(11963) ERROR: *** cfgtrace:request_route=[] c=[] l=0 a=26 n=t_load_contacts PYT {INVITE}: 5(11963) ERROR: *** cfgtrace:request_route=[] c=[] l=0 a=26 n=t_next_contacts However, it seems to be working anyway. Is the ERROR because I use the msg.call_function() ? Is there any better way to do serial forking using KEMI? Thank you, Samuel F

2 2

UAC and TLS
by Marino Mileti 04 May '17

04 May '17

Hi guys, is there any way to allow that the UAC module will make a REGISTER to another Kamailio with the parameters configured in uacreg table using the TLS protocol instead of UDP? Marino

3 2

if condition causes the kamailio 4.4.1 to not start
by David Escartín Almudévar 04 May '17

04 May '17

hello all we have seen today something quite weird having this line in the kamailio.cfg (1) else if (status =~ "4" || status =~ "5" || status =~ "6") { causes the process not to start hacing this error May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Invalid argument [22] May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4432]: tcp_main_loop(): failed to add process 37 unix socket to the fd list May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4523]: tcp_main_loop(): exiting... but placing (2) } else if (status =~ "^[456]") { instead, makes it work this happens with kamailio 4.4.1 and the platform is running on centos6.5 on the other hand we have the same line (1) on an ubuntu 10.0.4 makes the kamailio also start properly could you know why this is happening? best regards david

3 3

How does dlg_bridge work?
by Pranathi Venkatayogi 04 May '17

04 May '17

I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Kamailio KEMI / Python and the usage of KSR.pv.get("$rm") vs msg.Method
by Samuel F. 04 May '17

04 May '17

Hi, I see in the examples that KSR.pv.get("$rm") is frequently used rather than the more simple msg.Method. Why is this preferred? Kind regards, Samuel F

2 1

Freeswitch 403 Forbidden on invite
by Ravi sanyal 04 May '17

04 May '17

Hello, I'm new to using freeswitch. I'm using a client called SIPJS to communicate with freeswitch. The SIP message and it's response: INVITE sip:1001@104.130.11.232 SIP/2.0 Via: SIP/2.0/WS CODE;branch=BRANCH Max-Forwards: 70 To: <sip:1001@104.130.11.232> From: "1000" <sip:1000@104.130.11.232>;tag=7b5phc9nhb Call-ID: sun3fqjm7nb7rs06dbuu CSeq: 3769 INVITE Contact: <sip:k86b2npa@v9figramd1qs.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.7.7 Content-Type: application/sdp Content-Length: 759 v=0 o=- 84832118975142245 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE data a=msid-semantic: WMS m=application 49422 DTLS/SCTP 5000 c=IN IP4 111.69.178.179 a=candidate:1520298561 1 udp 2113937151 192.168.178.32 49422 typ host generation 0 network-cost 50 a=candidate:900384389 1 udp 2113939711 2406:e007:2f57:1:e951:8d6e:71bd:cff8 49423 typ host generation 0 network-cost 50 a=candidate:842163049 1 udp 1677729535 111.69.178.179 49422 typ srflx raddr 192.168.178.32 rport 49422 generation 0 network-cost 50 a=ice-ufrag:6Jk+ a=ice-pwd:XDsf8/lOfUpw8f3CQas5srLl a=fingerprint:sha-256 23:DF:44:BC:1B:A5:B3:77:E0:ED:54:F9:A1:EE:CB:A1:99:FC:3A:0F:A2:2B:60:07:9A:F9:2D:12:BB:5B:60:49 a=setup:actpass a=mid:data a=sctpmap:5000 webrtc-datachannel 1024 SIP/2.0 403 Forbidden Via: SIP/2.0/WS CODE;branch=BRANCH;received=111.69.178.179;rport=64431 From: "1000" <sip:1000@104.130.11.232>;tag=7b5phc9nhb To: <sip:1001@104.130.11.232>;tag=gt0m2j3aa4m1K Call-ID: sun3fqjm7nb7rs06dbuu CSeq: 3769 INVITE User-Agent: FreeSWITCH-mod_sofia/1.9.0-380-c66a012~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, 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-Length: 0 I thought to try get the invite to happen after subscribe events had taken place however the same messages are received back. This leads me to think that it's something to do with my freeswitch installation. Register messages are going through fine. that tells me that the users are being authenticated when talking to freeswitch. Nathan -- *Virtual Guard Ltd* *info(a)virtualguard.co.nz <info(a)virtualguard.co.nz>*

1 0

How does dlg_bridge work?
by Pranathi Venkatayogi 03 May '17

03 May '17

I have TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. I am attaching Kamailio log / my config files for reference. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Unknown reason for 500 I'm terribly sorry, server error occurred
by Pranathi Venkatayogi 03 May '17

03 May '17

I am encountering "500 I'm terribly sorry, server error occurred" . However I don't see this message in the Kamailio.log or anything related to find the root cause. I am attaching herewith the packet capture and the entire Kamailio log and the config files. Can someone please shed some light how do I go about debugging this? Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

rtpengine cpu usage
by Diego Nadares 03 May '17

03 May '17

Hi guys, I'm using rtpengine (compiled from source) and I see a lot of difference in cpu usage comparing with rtpproxy. Is this normal? With 600 calls: cpu *41.7%* rtpengine #rtpengine threads 4 # lscpu Architecture: x86_64 CPU op-mode(s): 32-bit, 64-bit Byte Order: Little Endian *CPU(s): 2* On-line CPU(s) list: 0,1 Thread(s) per core: 1 Core(s) per socket: 1 Socket(s): 2 NUMA node(s): 1 Vendor ID: GenuineIntel CPU family: 6 Model: 26 Model name: Intel(R) Xeon(R) CPU X7560 @ 2.27GHz Stepping: 4 CPU MHz: 2259.955 BogoMIPS: 4522.00 Hypervisor vendor: VMware Virtualization type: full L1d cache: 32K L1i cache: 32K L2 cache: 256K L3 cache: 24576K NUMA node0 CPU(s): 0,1 Calls established with a=rtpmap:0 PCMU/8000 Thanks in advance! Diego.

3 3

if condition causes the kamailio 4.4.1 to not start
by David Escartín Almudévar 03 May '17

03 May '17

hello all we have seen today something quite weird having this line in the kamailio.cfg (1) else if (status =~ "4" || status =~ "5" || status =~ "6") { causes the process not to start hacing this error May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Invalid argument [22] May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4432]: tcp_main_loop(): failed to add process 37 unix socket to the fd list May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4523]: tcp_main_loop(): exiting... but placing (2) } else if (status =~ "^[456]") { instead, makes it work this happens with kamailio 4.4.1 and the platform is running on centos6.5 on the other hand we have the same line (1) on an ubuntu 10.0.4 makes the kamailio also start properly could you know why this is happening? best regards david

1 0

Re: [SR-Users] topos and PRACK problem
by Sergey Basov 03 May '17

03 May '17

Hi, Daniel. Seems I found how to fix PRACK handling. It works for me. please lock at https://github.com/kamailio/kamailio/pull/1097 Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com 2017-04-27 13:04 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: > Yes, you are right. > > But now before 200 OK there empty field b_contact. > > May be you does not populate it from contact in 183 Progress? > > I see that this field is not empty only after 200 OK with a Contact field. > -- > Best regards, > Sergey Basov e-mail: sergey.v.basov(a)gmail.com > > > 2017-04-27 12:37 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> Hello, >> >> thanks for troubleshooting further. I haven't got the time to look at >> the source code, but I expect that the b-leg attributes (contact, record >> routes) to be set on 200ok for dialog (topos_d). I think for PRACK, the >> routing information should be stored and taken from transaction (topos_t). >> >> Cheers, >> Daniel >> >> On 27.04.17 11:29, Sergey Basov wrote: >>> Hi, Daniel. >>> >>> I just done one more test topos with re-invite which comes from caller >>> to callee, same direction as PRACK. >>> >>> And I found that, in case of prack, b_contact field is empty... >>> >>> Please find attached debug=3 part of re-invite message. >>> >>> Hope it helps. >>> -- >>> Best regards, >>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>> >>> >>> 2017-04-27 10:30 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>> Hi, Daniel >>>> >>>> Please look at attached part of debug=3 while receiving and parsing PRACK. >>>> At line 208 seems rr module does not find correct part of record_route >>>> which in DB consists from 2 parts. >>>> >>>> So at line 219 and later uac module cannot restore uris. >>>> >>>> Thank you. >>>> -- >>>> Best regards, >>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>> >>>> >>>> 2017-04-26 17:08 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>>> Thanks for workaround. >>>>> >>>>> But I will wait for you solution ) >>>>> >>>>> I ready for testing ) >>>>> >>>>> Thank you Daniel for your work! >>>>> >>>>> -- >>>>> Best regards, >>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>> >>>>> >>>>> 2017-04-26 16:57 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>>>>> Hello, >>>>>> >>>>>> >>>>>> On 26.04.17 14:53, Sergey Basov wrote: >>>>>>> Hi All. >>>>>>> >>>>>>> I have just try to test topos with GW which requires PRACK. >>>>>>> >>>>>>> As you can see UA at packet 21 send PRACK to topos contact, but after >>>>>>> topos, on other kamailio side in PRACK request line present not >>>>>>> kontact but record-route header. >>>>>>> >>>>>>> Can you fix it? >>>>>>> >>>>>>> >>>>>> probably needs to look into the code. If you need a quick workaround, >>>>>> try to remove Supported header from INVITE so the callee should no >>>>>> longer Require 100rel. >>>>>> >>>>>> 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 Conference - May 8-10, 2017 - www.kamailioworld.com >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (SER) - Users Mailing List >>>>>> sr-users(a)lists.kamailio.org >>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com >> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com >>

2 9

acces $dlg_var in failure_route
by Diego Nadares 03 May '17

03 May '17

Hi guys, I'm trying to access dialog vars in failure_route with no success. Dialog exists but vars are null and I'can modify them either. Is there any restriction? if ($dlg_var(v1) == 0){ $dlg_var(v1) = "1"; if (is_known_dlg()){ xlog("L_NOTICE", "negative_reply $ci $T_rpl($si) $dlg_var(v2)"); $dlg_var(v2) = $T_rpl($si); } } Log Apr 26 12:36:26 drwsip02 /usr/local/sbin/kamailio[27223]: NOTICE: <script>: negative_reply 6daf5e8d-f47972c1(a)172.31.4.21 * <null>*

2 5

Oracle and package function
by Diego Nadares 03 May '17

03 May '17

Hi list, I compiled db_unixodbc and sqlops to connect with oracle db. I have a function that has dml inside. Because of this I can't just execute 'select pkg.function(a, b) from dual'. I'm trying with BEGIN / END. The row is inserted but always are log errors: Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [res.c:60]: db_unixodbc_get_columns(): no columns returned from the query Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [res.c:261]: db_unixodbc_convert_result(): getting column names failed Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [dbase.c:211]: db_unixodbc_store_result(): failed to convert result Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: <core> [db_query.c:188]: db_do_raw_query(): error while storing result Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [DECLARE ret CURSOR] My query: sql_query("cb","DECLARE ret CURSOR; BEGIN ret := pkg_partitioning.insert(True,'1'); END;"); I tried with other functions and no matter what type returns the oracle function; always logs errors. It's like unixodbc is waiting for a result with rows. Any hint? Thanks in advance, Diego

2 2

Kamailio 4.4 NAT issue, nathelper I can not get to work
by Jason Davies 03 May '17

03 May '17

Hello I have my servers behind a NAT, with servers that make a call with bypass media / direct media all works fine. But when my media servers need to process audio I'm having issues. Looking at the SDP my local IP of the server is in there. I changed my config to add the nethelper.so and add the below: # ----- NAT Helper ----- #!ifdef WITH_NAT #modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #modparam("usrloc", "nat_bflag", 6) #!endif In my route I added: if (search("User-Agent: FS Box") {fix_nated_sdp("3");};

1 0

TLS SIP packet tracing and visualization
by Giovanni Maruzzelli 03 May '17

03 May '17

Hello fellows, after some experimentation with various tools, I come out with a little shell tool that maybe can be useful to you too. It can only work with non-forward secrecy ciphers, obviously, and only if is started before the client do the initial TLS handshake (eg, just restart the client). Forward secrecy cannot be decrypted after fact, so don't waste effort. An example of ciphers that can be decrypted are the "AES256-SHA" openssl cipher group. You can use ssldump to check what cipher is used by serverhello. Enjoy, make it better, and share it :) #!/bin/bash # brought to you by Giovanni Maruzzelli # SERVERIP="192.168.1.150" SERVERPORT="5061" PRIVKEY="/etc/certs/privkey.pem" STDERR2DEVNULL=" 2>/dev/null " REGEX="notyet" if [ -z "$1" ]; then REGEX="\\\.*" else REGEX="$1" fi FILTER="ssl.app_data and sip matches" FILTER2="$FILTER \"$REGEX\"" FILTER3="'$FILTER2'" ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u 's/^[0-9]*$/\n==&==============================/g'" echo "" echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" and try again" echo "" echo "NB: remember to quote and escape match patterns, using triple slash" echo " eg, for matching 1010(a)pbx.example.com, use \"1010(a)pbx.example.com \"" echo " eg, for matching anything, use \"\\\\\\.*\"" echo " eg, for matching *98, use \"\\\\\\*98\"" echo "USAGE: $0 \"\\\\\\*98(a)pbx.example.com\"" echo "" case "$1" in -help|--help|?) exit 0 ;; *) echo "THIS TIME WE'RE DOING:" echo "tshark $ARGUMENT" echo "" bash -c "tshark $ARGUMENT" ;; esac -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 3

Memory leak when reloading htable using db_cluster
by Kristian F. Høgh 03 May '17

03 May '17

Hi list, I use current git master. When I reload a htable using "kamcmd htable.reload htable1", the "ctl handler" process leaks 384 bytes of pkg memory If I use a direct mysql connection without db_cluster, the is no memory leak modparam("db_cluster", "connection", "con1=>mysql://user:pass@ip.addr/database") modparam("db_cluster", "connection", "con2=>mysql://user:pass@ip.addr2/database") modparam("db_cluster", "cluster", "cls1=>con1=9s9s;con2=8s8s") modparam("htable", "db_url", "cluster://cls1") # modparam("htable", "db_url", "mysql://user:pass@ip.addr/database" modparam("htable", "htable", "htable1=>size=8;autoexpire=0;dbtable=htable1;") 16(10429) ALERT: qm_status: 6513. N address=0x7f4e65e01720 frag=0x7f4e65e016e8 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from db_cluster: dbcl_api.c: db_cluster_init(294) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed 16(10429) ALERT: qm_status: 6514. N address=0x7f4e65e01808 frag=0x7f4e65e017d0 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from core: db.c: db_do_init2(298) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed 16(10429) ALERT: qm_status: 6515. N address=0x7f4e65e018f0 frag=0x7f4e65e018b8 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from core: db.c: db_do_init2(298) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Regards, Kristian Høgh Uni-tel A/S

2 8

SIP over WebSockets
by Nicolas Pace 02 May '17

02 May '17

Hi everyone, As I shared with you before, We are working on a mobile app to connect the Community Cellular Networks (that uses 2G Cellular Networks) with the rest of the world via SIP, and we are considering using Kamailio for this. We choosed to build a Progresive Web App (PWA) through WebSockets to reduce the friction involved in using a new tool. My question is, is there a reference config to use a web client through WebSockets? I don't have a clue on how to get from the stock config: https://gist.github.com/nicopace/9d44bf813fe309df69ff4cd0affe8324#file- _etc_kamailio_kamailio-conf to a fully functional WebSocket-enabled config. I've already read the WebSocket module documentation, but it is hard to understand as it assumes you already know how Kamailio works... too many moving parts :( I tried with https://github.com/caruizdiaz/kamailio-ws but it is giving me some errors: May 02 13:37:13 ubuntu kamailio[2329]: : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/modules.cfg, line 48, column 12-25: failed to load module May 02 13:37:13 ubuntu kamailio[2329]: ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command ws_handle_handshake (params 0) May 02 13:37:13 ubuntu kamailio[2329]: : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 143, column 41: unknown command, missing loadmodule? Which relates with: https://github.com/caruizdiaz/kamailio-ws/blob/master/modules.cfg#L48 and https://github.com/caruizdiaz/kamailio-ws/blob/master/kamailio-ws.cfg#L 143 What I'm trying to accomplish is to have a PWA that sends SIMPLE messages and CALLS to and from SIP Users (the 2G users are behind a FreeSwitch, so it could be done). Just to give you some context, we will also add Push Notifications as 'ringing' or 'meesage recieved' alert, and that would be all (we also don't know how to do it). Well... hope to hear from you guys! Thanks!

2 2

TCP read/write buffer size on per connection or per module basis
by M S 02 May '17

02 May '17

Hi, I have a kamailio v4.4.5 deployment that serves TCP clients for SIP and XCAP requests (without SIP presence). The kamailio's core TCP parameters allow us to set TCP read and write buffer sizes which are set globally for all TCP connections. However, i noticed that in general SIP over TCP request is much smaller (<8KB for audio + video SIP INVITE request) then XCAP requests (say to support up to 500 contacts we need TCP buffers =~ 64KB). So, if I set global TCP buffer size according to XCAP service needs (e.g. 64KB), they waste too much RAM for SIP connections (assuming max SIP packet is 8KB, there is 56KB buffer space wasted per connection). On the other hand, if i set global TCP buffer sizes to SIP service needs (e.g. 8KB) then it is insufficient for XCAP (only 70-80 contacts would fix in XCAP list). Therefore, i am looking for a way to allow TCP buffer size settings per connection OR per module basis. Is it possible? If so then how? Thank you.

2 2

Retrieving the custom header parameter value from contact header
by vivek＠advaitamtech.com 02 May '17

02 May '17

Hi, Am trying to extract a custom parameter from the contact header. Below it the example of contact header, Eg: Contact: <sip:fayazat@XXX.XXX.XXX.XXX:5060;tgrp=tg_0615_prsw1> I want to extract the parameter value of "tgrp" Tried using @hf_value.contact.p.tgrp . But it is giving me complete contact header. I want only "tgrp" paramter value "tg_0615_prsw1". Anyone , any sugeestions? Thanks, Vivek.

3 3

TCP port re-using when doing HTTP transfer
by Ramachandran, Agalya (Contractor) 02 May '17

02 May '17

Hi Giacomo, I am using http_async_client module to do http transfer. Currently am testing the HTTP service with a mock server. If I send the HTTP response immediately for the POST request, then the TCP connections are reused from kamailio. In the mock server if I send the response with a delay of 1 sec then, TCP port are opened and closed for every HTTP transfer. Is there a way or any configuration to limit the number of TCP port opening for the HTTP transfers? Please clarify me. Regards, Agalya

2 1

Kamailio IMS - Handling 2 SIP transports in an IMS client
by Haggerty, Paul (phaggert) 01 May '17

01 May '17

Hi, Our client registers with the IMS core using UDP but uses TCP when the packet size is over 1300 bytes. When the client sends the TCP reques, the PCSCF replies with 403 Forbidden "you need to reg with the SCSCF" We've traced this back to the code in the PCSCF and it looks like it is checking the via headers and the received ip/port combinations for a match. The actual contact header looks like it matches the registered client. Question is what do we need to do in the client to ensure that the PCSCF recognizes the request as one from a registered client? We've tried the config options is_registered_fallback2ip , ignore_contact_rxport_check and they don't seem to be used. I'm attaching a network trace and included here are some debug messages from the PCSCF. In the network trace the client is 10.8.0.6. PCSCF Trace messages ------------------------------------------- 34207) DEBUG: ims_registrar_pcscf [service_routes.c:208]: getContactP(): This is a request - using first via to find contact 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:216]: getContactP(): searching for contact with host:port:proto contact [2://10.8.0.6:49946] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:457]: get_pcontact(): Searching for contact with AOR [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1;alias=10.8.0.6~49946~2] in P-CSCF usrloc based on VIA [2://10.8.0.6:49946] Received [2://10.8.0.6:49946], Search flag is 0 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:463]: get_pcontact(): Have an AOR to search for 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:468]: get_pcontact(): checking for rinstance10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:480]: get_pcontact(): no rinstance param 10(34207) DEBUG: ims_usrloc_pcscf [usrloc.c:148]: get_aor_hash(): Returning hash: [1662767777] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:503]: get_pcontact(): get_pcontact slot is [161] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:507]: get_pcontact(): comparing contact with aorhash [1662767777], aor [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:508]: get_pcontact(): contact host [10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:509]: get_pcontact(): contact received [1:10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:543]: get_pcontact(): contact not found in memory 10(34207) INFO: ims_registrar_pcscf [service_routes.c:258]: getContactP(): Contact not found based on Contact-header, trying IP/Port/Proto 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:457]: get_pcontact(): Searching for contact with AOR [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1;alias=10.8.0.6~49946~2] in P-CSCF usrloc based on VIA [2://10.8.0.6:49946] Received [2://10.8.0.6:49946], Search flag is 1 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:463]: get_pcontact(): Have an AOR to search for 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:468]: get_pcontact(): checking for rinstance10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:480]: get_pcontact(): no rinstance param 10(34207) DEBUG: ims_usrloc_pcscf [usrloc.c:148]: get_aor_hash(): Returning hash: [1662767777] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:503]: get_pcontact(): get_pcontact slot is [161] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:507]: get_pcontact(): comparing contact with aorhash [1662767777], aor [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:508]: get_pcontact(): contact host [10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:509]: get_pcontact(): contact received [1:10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:543]: get_pcontact(): contact not found in memory 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:263]: getContactP(): No entry in usrloc for 10.8.0.6:49946 (Proto 2) found! 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:300]: getContactP(): Asserted identity not set10(34207) ERROR: *** cfgtrace:request_route=[Orig_Initial] c=[/opt/OpenEPC/etc/pcscf2.cfg] l=975 a=26 n=xlog 10(34207) WARNING: <script>: 2.1 10(34207) ERROR: *** cfgtrace:request_route=[Orig_Initial] c=[/opt/OpenEPC/etc/pcscf2.cfg] l=976 a=26 n=send_reply Appreciate any help we can get, - Paul

1 0

handle_uri_alias persists through failure_route
by David Wilson 01 May '17

01 May '17

Hello All, We use registrar servers (call them A,B) for TLS connections from clients and a central routing proxy (P) that authenticates and manages all call routing. On INVITE, P uses lookup(“location”) and handle_uri_alias() to route the INVITE to A or B depending on the AOR from REGISTER. In failure_route I’m trying to send INVITE to a voicemail URI at host (V), using alias_db_lookup to retrieve it and sending with t_relay(). However, despite showing V as the host in the URI, the message is sent to A (or B) as if the handle_uri_alias is still in effect. I’ve tried calling revert_uri() before the alias_db_lookup(), but this doesn’t seem to help. Is there a way to reset the uri_alias? The failure_route (simplified) looks like this: failure_route[1] { if (method == "INVITE") { revert_uri(); if (alias_db_lookup("dbaliases")) { t_relay(); } } } I have also tried using t_relay(“$rd”, “$rp”) after confirming (by logging) that the uri contains the desired values, but kamailio failed to start, with this error logged: ERROR: tm [tm.c:672]: fixup_hostport2proxy(): TM module:fixup_hostport2proxy: bad port number <$rp> Any suggestions? Best regards, Dave.

3 3

Cannot get service to start
by Stephen Tyers 28 Apr '17

28 Apr '17

Hi Apologies, I was a bit hasty in my previous email, I forgot to run the Chown command I still can’t get the service to start and I’m getting the below error. I’ve gone through all the steps multiple times, please help. /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceFailed to start kamailio.service: Unit kamailio.service failed to load: No such file or directory. failed! Kind regards Stephen tyers

2 1

Date Time with microsec
by Diego Nadares 28 Apr '17

28 Apr '17

Hi Guys, Is to convert or obtain $TV(s) - seconds (cached at first call per sip message) $TV(u) - microseconds (cached at first call per sip message) to datetime.microsecs? I'm trying to save 180/183 datetime (cached) like YYYY-MM-DD HH:mm:SS.microsecs $dlg_var(ring_time) = $timef(%Y-%m-%d %H:%M:%S); <--this is current time, not cached $dlg_var(ring_time_micro) = $TV(u); <-- this is cached us Thanks in advance! Diego.

2 3

htable Arrays and dbmode 1
by Sebastian Damm 28 Apr '17

28 Apr '17

Hi, I'm trying to use a htable array. For that, I have inserted some values in the database (in pseudo CSV): key_name; key_type; value_type; key_value '1234567'; 1; 0; 'foo' '1234567', 1, 0, 'bar' '1234567', 1, 0, 'baz' insert into htable (key_name , key_type , value_type , key_value ) values ('1234567', 1, 0, 'foo'),('1234567', 1, 0, 'bar'),('1234567', 1, 0, 'baz'); Now after just starting Kamailio, reading those values, from db and then stopping Kamailio again, the table content looks like this: key_name; key_type; value_type; key_value '1234567[2]'; 0; 0; 'baz' '1234567[0]', 0, 0, 'foo' '1234567[1]', 0, 0, 'bar' '1234567::size', 0, 1, 3 Is this how it is supposed to work? When re-reading those entries, the htable contents look the same as during the first start, but in the database it is a bit harder to read. And while I'm at it: Is there a way to regularly write back the current htable contents to the database? Or even to sync every write directly into the database? Thanks in advance Sebastian

1 0

Not able to replace route_uri in contact header uri
by Narayan P 28 Apr '17

28 Apr '17

Hi All, I am using pjsip client and kamailiio,both running on different servers.I want to replace the uri on which pjsip is running with the uri on which kamailio is running in contact header. I put some logs on route[REGISTRAR] and able to see the $route_uri as kamailio server uri($avp(s:fs) and $hdr(Contact) as pjsip server uri ($avp(s:contact).But I am not able to replace the uri with avp_subst() function. Can anybody help me how to replace the uri or with any any other function. Below I have put the snippet of config file. # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } xlog("Narayan: before sending contact\n"); $avp(s:contact) = $hdr(Contact); $avp(s:fs) = $route_uri; xlog("Narayan: Forced socket is $avp(s:fs)\n"); xlog("Narayan: contact header is $avp(s:contact)\n"); avp_subst("$avp(s:contact)/avp(s:contac)/g", "/(.*)(a)(.*)/$route_uri/"); #avp_pushto("$ru/domain","$fd"); #rewritehostport("185.122.206.62:5060"); t_on_reply("MANAGE_REPLY"); t_relay(); #if (!save("location")) { # sl_reply_error(); #} exit; } Any suggestion will be highly appreciated. Thanks, Narayan

3 11

Problem with installing Kamailio 5
by Stephen Tyers 28 Apr '17

28 Apr '17

Hi, I’ve hit a road block when following URL http://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/ I’m very new to open source and linux so please bare with me. I’ve reached step Kamailio start and I get the below error. Please help, I’ve think I’ve followed all the proceeding commands and created the databases. root@kamailio:/var/run# /etc/init.d/kamailio start -bash: /etc/init.d/kamailio: Permission denied Many thanks Steve

1 0

An Potential Issue in Kamailio Installation Guide
by Edward Lu 28 Apr '17

28 Apr '17

Dear Sir, This is Edward, nice to meet you. When I follow the installation guide of Kamailio 5.0.x https://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/ to install Kamailio, I met an issue, it report "unit kamailio.service not found" when I issue command /etc/init.d/kamailio start. After google, I found one step was missed: to copy kamailio.service to /etc/system.d/system. Except above issue, all others are all okay for my installation on Ubuntu 16.04. Thanks & Regards, Edward

1 0

How do we send "MSRP Message" from config
by Pranathi Venkatayogi 28 Apr '17

28 Apr '17

While implementing Kamailio to hold customer call, I would to send "Pls hold on looking for an agent" MSRP message from Kamailio config. Is it doable? >From the module, we can only send msrp_reply and msrp_relay... Please advise. Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 1

Re: [SR-Users] SIGCHLD causes termination, but it appears that no child process has actually stopped.
by Cody Herzog 27 Apr '17

27 Apr '17

Thanks for the quick reply. Pull request #1101 created: https://github.com/kamailio/kamailio/pull/1101

1 0

kamailio and siremis
by David Villasmil 27 Apr '17

27 Apr '17

Hello, Maybe this isn't the right place. Just installed SIREMIS (4.3.0) on a kamailio 5, everything works with the admin user, but when creating a SIP USER, after clicking on "Save" the page goes to http://1.2.3.4/siremis/system/user_detail/5 which just shows a blank page. Nevertheless, the user IS created and shows up on the "user management", but when the user tried to login, after clicking on "login", it shows a "processing login" and a pop-up comes up ( http://1.2.3.4/siremis/index.php/user/login) which just show a "1" here's the log: {code} '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.username, T0.password, T0.enctype, T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM user T0 WHERE T0.username='nocadmin' and status='1' LIMIT 1 BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.username, T0.password, T0.enctype, T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM user T0 WHERE T0.username='nocadmin' LIMIT 1 BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.first_name, T0.last_name, T0.display_name, T0.company, T0.department, T0.position, T0.fast_index, T0.photo, T0.phone, T0.mobile, T0.fax, T0.zipcode, T0.province, T0.city, T0.street, T0.country, T0.email, T0.webpage, T0.qq, T0.icq, T0.skype, T0.yahoo, T0.misc, T0.type_id, T1.name, T0.sortorder, T0.user_id, T0.published, T0.default, T0.access, T0.params, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM contact T0 LEFT JOIN contact_type T1 ON T1.id = T0.type_id WHERE (T0.published=1 or T0.user_id='4') AND T0.user_id='4' ORDER BY T0.sortorder ASC, T0.id ASC LIMIT 1 BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.user_id, T0.role_id, T1.name, T1.startpage FROM user_role T0 INNER JOIN role T1 ON T1.id = T0.role_id WHERE T0.user_id='4' BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Insert Sql = INSERT INTO event_log (user_id, ipaddr, event, message, comment, timestamp) VALUES (:_v1, :_v2, :_v3, :_v4, :_v5, :_v6); BIND: 4,6.7.8.9,LOGIN,MSG_LOGIN_SUCCESSFUL,a:2:{i:0;s:8:"nocadmin";i:1;s:13:"6.7.8.9";},2017-04-27 20:25:22','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','New record Id is 14','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Query Sql = SELECT T0.id, T0.username, T0.password, T0.enctype, T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM user T0 WHERE T0.username='nocadmin' LIMIT 1 BIND: ','n/a' '04/27/2017','20:25:22','DEBUG','DATAOBJ','Update Sql = UPDATE user SET lastlogin = :_v1, update_by = :_v2, update_time = :_v3 WHERE id='4'; BIND: 2017-04-27 20:25:22,4,2017-04-27 20:25:22','n/a' {code} nothing else is shown, nothing happens... Thanks for your help! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

1 0

How to prevent running multiple Kamailio instances on one host?
by Julia Boudniatsky 27 Apr '17

27 Apr '17

What is you suggestion for prevent running multiple Kamailio instances on one host? Thank you, Julia

5 9

JANSSON module not working for multiple JSON objects
by vivek＠advaitamtech.com 27 Apr '17

27 Apr '17

Hi All, Am using JANSSON module to parse the JSON object I receive. I tried using JSON module of kamailio. Since its giving error in loading the module due to error"libjson-c.so.3: cannot open shared object file: No such file or directory", I went ahead to use JANSSON module. JANSSON module is parsing the single JSON object perfectly. But not parsing the multiple JSON objects. JSON object am trying to parse: $avp(result1)= "[{"origin":"125.16.231.74","Dest":"725.76.731.77"},{"origin":"225.26.231.24","Dest":"225.26.231.27"}]" JANSSON API used: jansson_get("origin", "$avp(result1)", "$var(origin)"); Any suggestions will be really helpful. Thanks, Vivek

2 4

topos and PRACK problem
by Sergey Basov 27 Apr '17

27 Apr '17

Hi All. I have just try to test topos with GW which requires PRACK. As you can see UA at packet 21 send PRACK to topos contact, but after topos, on other kamailio side in PRACK request line present not kontact but record-route header. Can you fix it? Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 5

SIGCHLD causes termination, but it appears that no child process has actually stopped.
by Cody Herzog 27 Apr '17

27 Apr '17

Hello. I'm running Kamailio 4.3.5 on Ubuntu Server 14.04. I just started using the app_python module together with the OpenTok Python SDK, described here: https://tokbox.com/developer/sdks/python/ My basic Python test script works fine, but when I add an import statement in the Python script to bring in a package from the OpenTok Python SDK, Kamailio terminates right after launch with the following log message: "terminating due to SIGCHLD" I was surprised to not find any extra log messages indicating which child process was causing the signal. I hacked the code in the handle_sigs() function in main.c.c to NOT terminate in response to SIGCHLD, and was surprised to see that my Python script was working fine, and was able to properly use the OpenTok Python SDK. I then started adding more debugging log statements and discovered that the call to waitpid() was returning 0, which apparently indicates that no child process has changed state. It's very strange, because it seems that SIGCHLD is being sent and handled, but waitpid() is indicating that no child process has changed state. I'm not sure if that is valid situation which needs to be handled. Perhaps SIGCHLD is sometimes sent for spurious or harmless reasons which should be ignored, and not result in total Kamailio termination. My current workaround is to NOT terminate when SIGCHLD is received and waitpid() indicates that no child process has changed state. In other words, I only terminate when the waitpid() while() loop has been entered at least once due to a return value greater than 0. Does that seem like a change that might make sense in the master branch going forward, or does it seem dangerous somehow? I'm not sure if this is helpful, but the waitid() system call (available since Linux 2.6.9) provides more precise control over which child state changes to wait for. http://manpages.ubuntu.com/manpages/trusty/man2/wait.2.html Thanks for any help.

2 1

Query with regard to http_client and http_async_client
by Ramachandran, Agalya (Contractor) 26 Apr '17

26 Apr '17

Hi Giacomo, I have few questions with regard to rest client in kamailio. 1) I see there are two modules in kamailio for rest client. http_client, http_async_client. I hope http_client does the http transfer in synchronous way. http_async_client does the http transfer in asynchronous manner. Correct me if am wrong here. 2) I want to achieve https transfers in async way, so choosing http_async_client. Is there a way that kamailio maintains persistent TCP connection pooling for http transfer. I see there is a "workers" parameter in http_async_client. By increasing this number does it result it more http request's handling? 3) Also, I have some questions regarding documentation for module http_async_client in 5.0 version. In the API exposure, it says "http_async_query(url, route_name)". But the example if I want to use POST, it says "http_async_query("http://example.com/test.php", "{'r_uri':'$rU', 'f_uri':'$fU'}", "HTTP_REPLY");" There is a parameter mismatch between the example and the actual function. If I try to use GET query, kamailio service is starting. But if I try to give the POST query with payload, kamailio service is not started. It says, sctp_core_destroy(): SCTP API not initialized. Also I have noticed that, functions like http_append_header(),http_set_method() are removed in 5.0 version, whereas we have this API's is 4.x versions. Is there some issue with the documentation for 5.0 ? Please let me the know the details for the above queries. Regards, Agalya

3 3

forwarding multiple times
by hdssdsdsdsfsdf hdssdsdsdsfsdf 26 Apr '17

26 Apr '17

Hi all, I'm trying to transmit a CANCEL sip message from one server to multiple other servers (using ds_select_domain and ds_next_domain). Ideally, I would simply use forward() multiple times. The problem is however, that this would apply the changes of forward() multiple times too, for example adding multiple via-headers. I tried to use forward() for the first transmission and then send() for the next ones, but it turns out send() uses the original message to send, and not the message as crafted by forward. As such, it seems I need exactly what's in the middle of these: Either a forward that operates on the original message, or a send that operates on the message as crafter by previous forward. Does anyone know how to achieve this? The only other solution I see is to manually write some forwarding routing, but it seems tedious to craft the via-headers and fix the domain to be forwarded to. Thanks and cheers,

1 1

How to eliminate Content-Length:0 from sl_send_reply
by Pranathi Venkatayogi 25 Apr '17

25 Apr '17

Hi, How do I eliminate Content-Length:0 from sip reply when using sl_send_reply? I wanted to generate 200 ok, with SDP payload. I appended headers like below. The packet seems ok except there is extra "Content-Length:0", which is causing problems. Any workarounds? The packet being generated is: Note the extra content-length: 0 at the end. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.12:59404;rport=59404;branch=z9hG4bKPjb7d0391f8f7f479b812f6e5feadfa3c5;received=10.11.200.46 From: "cust1" <sip:cust1@devtranslation.sms-test.cyracom.com>;tag=afcffa277e2a43dc88fafc612f538864 To: <sip:spanish@devtranslation.sms-test.cyracom.com>;tag=37b119f34ca40c158c5b7306e1aa8d72.b73a Call-ID: 7f73e5d03e15425395fac930eae423bd CSeq: 31419 INVITE Content-Type: application/sdp Content-Length: 350 v=0 o=- 3688213899 3688213900 IN IP4 10.99.52.17 s=Blink 1.4.2 (Windows) t=0 0 m=message 2855 TCP/MSRP * c=IN IP4 10.99.52.17 a=path:msrps://10.99.52.17:2855/116701522521345;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active Content-Length: 0 Code in config is: append_to_reply("Content-Type: application/sdp\r\n"); append_to_reply("Content-Length: 350\r\n"); append_to_reply("\r\n"); append_to_reply("v=0\r\n"); append_to_reply("o=- 3688213899 3688213900 IN IP4 MY_IP_ADDR\r\n"); append_to_reply("s=Blink 1.4.2 (Windows)\r\n"); append_to_reply("t=0 0\r\n"); append_to_reply("m=message 2855 TCP/MSRP *\r\n"); append_to_reply("c=IN IP4 MY_IP_ADDR\r\n"); append_to_reply("a=path:msrps://MY_IP_ADDR:2855/$var(ssessid);tcp\r\n"); append_to_reply("a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml\r\n"); append_to_reply("a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml\r\n"); append_to_reply("a=setup:active\r\n"); sl_send_reply("200", "OK"); Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

3 8

Re: [SR-Users] Kamailio Issues
by Robert Boisvert 25 Apr '17

25 Apr '17

Santosh, Looks like you didn't include the mohdir information (see http://kamailio.org/docs/modules/4.4.x/modules/mohqueue.html#dir.parms) or there are no files in that directory. BTW, I rarely use this email address so please address your questions to sr-users(a)lists.kamailio.org. Hope this helps, Bob On Thu, Apr 20, 2017 at 11:26 AM, santosh gupta <santosh.gupta.g(a)gmail.com> wrote: > HI, > > Was working on Kamailio for MohQueue n getting the below error. > Let me know if you can help me. > > I had already checked everything. > 1. Files are located in proper directory. > 2. Files are having proper permissions. > 3. Database is having proper tables. > 4. Tried many versions of Kamailio i.e 4.4,5.0 also 5.0.1 > > ERROR: mohqueue [mohq_db.c:504]: update_mohq_lst(): Queue,Field > (operators,mohdir): Unable to find MOH files > > -------------------- > With Regards > Santosh Gupta > +91 9323043570 >

2 3

4.4.5 Shared memory leak with t_suspend/t_continue
by Julia Boudniatsky 25 Apr '17

25 Apr '17

Hello all, I need to fork an incoming call to the second destination, after receiving 180 reply from the first destination. I use kamailio version 4.4.5 It working fine, but with shared memory leak. The script part: $avp(00x)=$_s(sip:$var(00x)@$si:$sp;user=phone); if(!is_known_dlg()){ dlg_manage(); } t_relay(); append_branch("$avp(00x)"); if(t_suspend()) { xlog("L_NOTICE","FORK: suspend transaction[$T(id_index):$T(id_label)]\r\n"); } onreply_route [REPLY_TO_SOURCE] { if ($T_reply_code=="180") { t_continue("$T(id_index)", "$T(id_label)", "RELAY_FORK"); } } route[RELAY_FORK] { $ru=$avp(00x); $avp(tu) = $_s(sip:$rU@$td;user=phone); uac_replace_to("$avp(tu)"); remove_hf("Supported"); xlog("L_NOTICE","RELAY_FORK : $rm CSeq[$hdr(CSeq)] branch[$T_branch_idx]\r\n"); t_relay(); } Please find the result of *kamcmd corex.shm_summary* after 300 calls Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: summarizing all alloc'ed. fragments: Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 176 bytes from dialog: dlg_profile.c: set_dlg_profile(574) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 264 bytes from dialog: dlg_var.c: new_dlg_var(97) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 152 bytes from dialog: dlg_hash.c: dlg_set_leg_info(526) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 144 bytes from dialog: dlg_hash.c: dlg_set_leg_info(509) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 2 size= 1184 bytes from dialog: dlg_hash.c: build_new_dlg(441) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 24 bytes from dialog: dlg_var.c: new_dlg_var(106) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 24 bytes from dialog: dlg_var.c: new_dlg_var(116) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 88 bytes from dialog: dlg_hash.c: dlg_set_leg_info(517) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 337 size= 367448 bytes from tm: t_reply.c: relay_reply(1891) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 334 size= 549312 bytes from tm: t_msgbuilder.c: build_local_reparse(344) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 672 size= 1123536 bytes from core: msg_translator.c: build_req_buf_from_sip_req(2154) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 337 size= 505048 bytes from core: sip_msg_clone.c: msg_lump_cloner(978) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 331 size= 14784 bytes from tm: t_reply.c: update_totag_set(312) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 340 size= 3566680 bytes from core: sip_msg_clone.c: sip_msg_shm_clone(494) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 339 size= 5024 bytes from dialog: dlg_handlers.c: dlg_get_iuid_shm_clone(280) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 337 size= 2528256 bytes from tm: h_table.c: build_cell(317) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 331 size= 10096 bytes from tm: t_reply.c: update_totag_set(313) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 2390 size= 315368 bytes from core: usr_avp.c: create_avp(175) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 36480 bytes from core: counters.c: counters_prefork_init(207) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 10944 bytes from sl: sl_stats.c: init_sl_stats_child(125) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 512 bytes from tmx: tmx_pretran.c: tmx_init_pretran_table(90) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 14592 bytes from tm: t_stats.c: init_tm_stats_child(60) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 2736 bytes from kex: pkg_stats.c: pkg_proc_stats_init(79) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 88 bytes from core: cfg/cfg_struct.c: cfg_clone_str(130) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 648 bytes from core: cfg/cfg_struct.c: cfg_shmize(217) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: pt.c: init_pt(110) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: pt.c: init_pt(105) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 7984 bytes from core: pt.c: init_pt(104) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8192 bytes from htable: ht_api.c: ht_init_tables(381) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from cfgutils: cfgutils.c: mod_init(935) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from cfgutils: cfgutils.c: mod_init(929) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from cfgutils: cfgutils.c: mod_init(921) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 131088 bytes from dialog: dlg_hash.c: init_dlg_table(284) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from dialog: dlg_hash.c: init_dlg_table(276) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from dialog: dlg_hash.c: init_dlg_table(269) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from dialog: dlg_hash.c: init_dlg_table(264) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from dialog: dlg_timer.c: init_dlg_timer(57) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 32 bytes from dialog: dlg_timer.c: init_dlg_timer(48) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 312 bytes from dialog: dlg_profile.c: new_dlg_profile(187) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 432 bytes from dispatcher: dispatch.c: reindex_dests(601) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 678 size= 31456 bytes from tm: t_hooks.c: insert_tmcb(137) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 32 bytes from dispatcher: dispatch.c: ds_set_attrs(235) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 72 bytes from dispatcher: dispatch.c: add_dest2list(351) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 360 bytes from core: timer.c: register_timer(1011) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 840 bytes from dispatcher: dispatch.c: add_dest2list(325) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 16 bytes from dispatcher: dispatch.c: init_data(204) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 16 bytes from dispatcher: dispatch.c: init_data(195) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from dispatcher: dispatcher.c: ds_parse_reply_codes(1029) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from dispatcher: dispatcher.c: mod_init(309) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from dispatcher: dispatcher.c: mod_init(307) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from dispatcher: dispatch.c: ds_ping_active_init(102) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from sl: sl_funcs.c: sl_startup(83) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from sl: sl_stats.c: init_sl_stats(110) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 16 bytes from tm: t_hooks.c: init_tmcb_lists(74) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 16 bytes from tm: t_hooks.c: init_tmcb_lists(72) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 2097152 bytes from tm: h_table.c: init_hash_table(467) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 3 size= 96 bytes from core: cfg/cfg_ctx.c: cfg_register_ctx(47) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: usr_avp.c: init_avps(90) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: usr_avp.c: init_avps(89) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 16384 bytes from core: dst_blacklist.c: init_dst_blacklist(437) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: dst_blacklist.c: init_dst_blacklist(430) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 2 size= 96 bytes from core: timer.c: timer_alloc(514) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: dns_cache.c: init_dns_cache(366) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 16384 bytes from core: dns_cache.c: init_dns_cache(358) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 16 bytes from core: dns_cache.c: init_dns_cache(351) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: dns_cache.c: init_dns_cache(345) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: timer.c: init_timer(283) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 16384 bytes from core: timer.c: init_timer(282) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: timer.c: init_timer(281) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: timer.c: init_timer(280) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: timer.c: init_timer(269) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: timer.c: init_timer(237) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 278544 bytes from core: timer.c: init_timer(220) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: timer.c: init_timer(219) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: timer.c: init_timer(206) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 64 bytes from core: cfg/cfg_struct.c: cfg_child_cb_new(830) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: cfg/cfg_struct.c: sr_cfg_init(361) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: cfg/cfg_struct.c: sr_cfg_init(354) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: cfg/cfg_struct.c: sr_cfg_init(347) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: cfg/cfg_struct.c: sr_cfg_init(335) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: cfg/cfg_struct.c: sr_cfg_init(323) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 232 bytes from htable: ht_api.c: ht_add_table(278) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from pv: pv_shv.c: add_shvar(177) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 64 bytes from pv: pv_shv.c: add_shvar(170) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 80 bytes from pv: ../../lock_alloc.h: lock_set_alloc(70) Apr 24 15:08:53 vm-tkamasy1 /usr/local/sbin/kamailio[2795]: WARNING: fm_status: count= 1 size= 8 bytes from core: mem/shm.c: shm_core_lock_init(153) Thank you for help.

2 3

Kamailio SIP Proxy solution
by Hai Bui Duc Ha 25 Apr '17

25 Apr '17

Hi all, I'm researching the solution on Kamilio like SIP forward message: - At the moment, the customer has server IP PBX and their clients. - Then, the client just point the SIP Porxy configure to Kamailio. After that, Kamailio will forward to IP PBX and send the event to another server. - Everything will be not changed for customer site. I found the link do the similar thing but it still authenticate on Kamailio: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb… Do you have any advice ? Regrads, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

5 8

Warnings at apply_changes()
by Diego Nadares 25 Apr '17

25 Apr '17

Hi list, I'm converting body message to multipart. Calls establishes ok and Kamailio it's not logging errors but I'm seeing some info and warnings that I don't know it they are correct. Maybe you could tell me if it's normal or if I'm doing something wrong in my script. kamailio[13428]: INFO: <script>: Original content type is application/sdp kamailio[13428]: INFO: <core> [msg_translator.c:1693]: get_boundary(): Content-Type hdr has no params <application/sdp> kamailio[13428]: WARNING: <core> [msg_translator.c:1959]: build_req_buf_from_sip_req(): check_boundaries error kamailio[13428]: INFO: <script>: Appending GTD to body I use the following sintax: route{ ... route(MANAGE_MULTIPART); ... } route[MANAGE_MULTIPART]{ .... if (is_method("INVITE") && !has_totag() && has_body()){ ... xlog("L_INFO", "Original content type is $cT"); set_body_multipart(); *msg_apply_changes();* xlog("L_INFO", "Appending GTD to body"); ... } ... } Invite example: INVITE sip:222222222@172.16.205.122:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-28085-1-0 From: "111111111" <sip:1111111111@172.16.213.21:5060;user=phone>;tag=1 To: <sip:2222222222@172.16.205.122:5060> Call-ID: 1-28085(a)172.16.213.21 CSeq: 1 INVITE Contact: "1111111111" <sip:111111111@172.16.213.21:5060> Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 137 v=0 o=user1 53655765 2353687637 IN IP4 172.16.213.21 s=- c=IN IP4 172.16.213.21 t=0 0 m=audio 6000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 As you can see the log says that "Content-Type hdr has no params <application/sdp>" but in the line before says "Original content type is application/sdp". I'm running kamailio 4.4 by the way. Thanks in advance. Diego.

2 2

Re: [SR-Users] New developer: Guillaume Bour
by Siwek, Cezary 24 Apr '17

24 Apr '17

Nice work Guillaume! On 24 April 2017 at 15:21, Siwek, Cezary <cezary.siwek(a)vonage.com> wrote: > Nice work Guillaume! > > On 21 April 2017 at 09:55, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> I would like to announce that Guillaume Bour (https://github.com/gbour) >> has now developer privileges on Kamailio's gihub project. He has >> contributed recently a new module named keepalive: >> >> - http://kamailio.org/docs/modules/devel/modules/keepalive.html >> >> There is also a pull request from him waiting to be merge to drouting >> module, adding capability of detecting active/inactive gateways used by >> the module. >> >> Thanks for the contributions so far and looking forward to collaborate >> in the future! >> >> 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 Conference - May 8-10, 2017 - www.kamailioworld.com >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > >

1 0

Video recording with RTPENGINE
by Alberto Llamas 24 Apr '17

24 Apr '17

Hi guys, Any of you have experience using the '*start_recording()*' function of the RTPENGINE in a WebRTC scenario?. At this point, I am able to record the stream but I am not able to play them. The output is a .pcap file in a raw format. I have tried the conversion: raw -> H264 -> mp4/avi (with different tools: videosnarf, x264, ffmepg) and not luck. Do you know the right way to convert those stream in raw format?. Any hint will be appreciated. Regards,

3 5

Memory leak after enabling topos module
by Sergey Basov 24 Apr '17

24 Apr '17

Hi All. During loadtest of kamailio 5.0.1 with topos module I got memory leak in "udp receiver" process After 1600 sip calls I got 10 Mbytes of free private memory. I set limit to 32 Mbytes output of kamcmd corex.pkg_summary pid xxx command at begin test and before eom are in attached files. If you need more info please let me know. Thank you in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

3 11

SIREMIS
by Emil Morganti 24 Apr '17

24 Apr '17

Hello, I have installed kamailio 5.1.0-dev and would like to know if it is compatible with SIREMIS v4.3.0? Has anyone tried it?

2 1

MSRP Connection is not working
by Jegadeesan 24 Apr '17

24 Apr '17

Hi , I'm very new to Kamailio application and trying to setup the test environment for Android RCS client (https://github.com/android-rcs/rcsjta). Parallelly we have deployed the Kamailio on ECS2 and able to make a call/text using any other sip clients. Apparently, we had enabled the MSRP modules over TCP and tried to test with Android RCS. But getting stuck at the connecting screen. Although we could see SIP exchange logs between two clients. Can anyone help me in fixing this issue? -- Jegadeesan M

2 3

tm forward
by Slava Bendersky 24 Apr '17

24 Apr '17

Hello Everyone, What possible cause for messsage /usr/sbin/kamailio[4037]: ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): no branches for forwarding Thank you.

2 3

Kamailio not forwarding call
by Gonzalo Gasca Meza 22 Apr '17

22 Apr '17

Hi all, I worked in an prototype recently where I *had* the following setup working. Kamailio is using Postgres to forward the call to Twilio domain. *Topology:* SIP client (tls) > (SIP Provider) > (tls) Kamailio (udp) > Twilio > IVR Right now when I call from SIP Client to IVR, call is not forwarded. Nothing has changed on Kamailio side. *Troubleshooting:* a) *SIP client (tls) > (SIP Provider) > (tls) Kamailio (udp) > Twilio > IVR [Fails]* - Case described above. b) *Call from PSTN > (SIP Provider) > (tls) Kamailio (udp) > Twilio > IVR [Works]* I debug and call arrives to Kamailio but is not being forwarded to Twilio. This used to work before. When call works Kamailio seems to find remote domain, when fails we just sent a *502 Bad Gateway* message Twilio does not support DNS SRV records, just A host resolution. *Logs* While looking at debugs I found that this DNS logs: Pastebin below: *Call fails <https://pastebin.com/ttkK1Kkd> * Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:537]: _dns_hash_find(): (_sip._ udp.parzee.sip.twilio.com(31), 33), h=816 Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [resolve.c:741]: get_record(): lookup(_sip._ udp.parzee.sip.twilio.com, 33) failed Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:864]: dns_cache_mk_bad_entry(): (_sip._ udp.parzee.sip.twilio.com, 33, 60, 1) Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:798]: dns_cache_add(): adding _sip._ udp.parzee.sip.twilio.com(31) 33 (flags=1) at 816 Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:537]: _dns_hash_find(): (parzee.sip.twilio.com(21), 1), h=267 Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [resolve.c:937]: get_record(): skipping 0 NS (p=0xa1d2e7, end=0xa1d2e7) Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [resolve.c:952]: get_record(): parsing 0 ARs (p=0xa1d2e7, end=0xa1d2e7) Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:1741]: dns_get_related(): (0x7f1b5c227920 ( parzee.sip.twilio.com, 1), 1, *(nil)) (0) Apr 22 06:37:00 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28476]: DEBUG: <core> [dns_cache.c:840]: dns_cache_add_unsafe(): adding parzee.sip.twilio.com(21) 1 (flags=0) at 267 *Call works <https://pastebin.com/AkWsvFVA>* Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [dns_cache.c:537]: _dns_hash_find(): (_sip._ udp.parzee.sip.twilio.com(31), 33), h=816 Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [resolve.c:741]: get_record(): lookup(_sip._ udp.parzee.sip.twilio.com, 33) failed Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [dns_cache.c:864]: dns_cache_mk_bad_entry(): (_sip._ udp.parzee.sip.twilio.com, 33, 60, 1) Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [dns_cache.c:798]: dns_cache_add(): adding _sip._ udp.parzee.sip.twilio.com(31) 33 (flags=1) at 816 Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [dns_cache.c:537]: _dns_hash_find(): (parzee.sip.twilio.com(21), 1), h=267 Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [msg_translator.c:2876]: create_via_hf(): id added: <;i=7>, rcv proto=3 Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: <core> [msg_translator.c:158]: check_via_address(): (54.172.60.0, 54.172.60.0, 0) Apr 22 07:00:03 parzee-sip-kamailio1 /usr/local/sbin/kamailio[28481]: DEBUG: tm [t_funcs.c:368]: t_relay_to(): SER: new transaction fwd'ed Thank you!

1 1

integration kamilio with open IMS
by Med Amine Hajji 21 Apr '17

21 Apr '17

Hi i need ur help i want to integrate my Open IMS WITH KAMILIO wich configuration i must do ? thank you for helping me

1 0

Kamailio rtpengine sdp
by gerry kernan 21 Apr '17

21 Apr '17

Hi Thanks in advance if anyone can point me in the correct direction . I have kamailio running in front of some asterisk VM's. SIP endpoint register to their asterisk PBX via Kamailio dispatcher module. I'm running rtpengine with a Wan and private interface to bridge audio stream between these endpoints on the WAN to asterisk PBX running on LAN IP behind Kamailio. Calls from ext to ext work fine audio both directions , calls outbound to PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio both directions. But incoming calls via SIP provider I only get audio on stream from asterisk registered ext to external caller , no audio from external caller to the asterisk ext. I reckon I have something wrong in my Kamailio.cfg . if I register an ext direct to asterisk I get audio both ways on incoming calls. And rtp logs from rtpenegine show it as trying to send the rtp to the private address of the sip endpoint rather that its WAN address. I think my mistake in somewhere in the cfg below. Do I need to handle invites from the backend asterisk servers different that invites from sip endpoints? Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

2 4

NOTIFY request from PCSCF is on TCP transport rather than UDP
by Haggerty, Paul (phaggert) 21 Apr '17

21 Apr '17

Hello, I've got an IMS client which uses UDP for transport. Our application server is also using UDP. We are using a Kamailio IMS Core. The client REGISTERS with UDP and also does a SUBSCRIBE. Our application server responds with a NOTIFY to the client. Looking at the trace. The messages between the PCSCF and SCSCF are TCP. The REGISTER and SUBSCRIBE responses to the client from the PCSCF are UDP (which I would expect). However, when the PCSCF get the NOTIFY is tries to open a TCP connection to the Client. (I assume to send a NOTIFY). Any ideas of what is happening here? I'm fairly new to the Kamailio IMS. I am more familiar with the Open IMS Core. (which is not having this issue). Thanks, Paul

1 1

How ICE works with private IPs as candidates?
by Denys Pozniak 21 Apr '17

21 Apr '17

Hello! I have next topology: ISP->Kamailio + Rtpengine->WebRTC Sometimes I receive from Browser side SDP like below with private IPs in ICE candidates. - Please explain why it happens? - How ICE finds a pair? I think Rtpengine obtains packet from Browser and somehow matches with peer (as it can't reach Browser via private IP) U KAMAILIO_IP:34150 -> RTPENGINE_IP:22222 0_16586_545 d3:sdp962:v=0. o=- 3981552194926701250 2 IN IP4 127.0.0.1. s=-. t=0 0. a=msid-semantic: WMS 5efe361e-087d-47de-98f6-9b51427d3764. m=audio 50589 RTP/SAVPF 8 0. c=IN IP4 192.168.2.205. a=rtcp:9 IN IP4 0.0.0.0. a=candidate:710145796 1 udp 2122255103 <(212)%20225-5103> 2001::9d38:6abd:3c21:28f:3f57:fd32 50588 typ host generation 0 network-id 1 network-cost 50. a=candidate:2355154215 1 udp 2122194687 <(212)%20219-4687> 192.168.2.205 50589 typ host generation 0 network-id 2. a=ice-ufrag:sPAh. a=ice-pwd:RjXqOz8rpwjhKPU3BzXkJd1d. a=fingerprint:sha-256 04:AF:D4:6C:07:C2:DD:18:48:E0: BC:46:15:15:00:5F:C9:AD:CF:8D:F3:94:55:5A:54:0C:37:B3:E6:44:63:EC. a=setup:active. a=mid:audio. a=sendrecv. a=rtcp-mux. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=ssrc:3055599516 <(305)%20559-9516> cname:n0Cdw7SNbYDIba8s. a=ssrc:3055599516 <(305)%20559-9516> msid:5efe361e-087d-47de-98f6-9b51427d3764 41750b25-d640-4bad-8953-55f4e2684c49. a=ssrc:3055599516 <(305)%20559-9516> mslabel:5efe361e-087d-47de- 98f6-9b51427d3764. a=ssrc:3055599516 <(305)%20559-9516> label:41750b25-d640-4bad-8953- 55f4e2684c49. 3:ICE6:remove5:flagsl13:trust-address14:media-handover8: SDES-offe7:replacel6:origin18:session-connectione18: transport-protocol7:RTP/AVP8:rtcp-muxl5:offer6:accepte7: call-id37:domain.com_29465313:received-froml3:IP412:10.23. 255.93e8:from-tag13:QQcg5eFv08S9K6:to-tag10:nsebpmf7ia7:command6:answere

4 5

http_client_query API of http_client module issue.
by vivek＠advaitamtech.com 21 Apr '17

21 Apr '17

Hi All, Am executing the curl GET request using the API http_client_query(url, [post-data], result) like its mentioned in the http_client module page. In the result part of the called API, I am receiving only the first line of the response to the GET request sent. How do I get or access the full response? Thanks, Vivek.

1 0

Kamailio World 2017 - grants for students and underrepresented people
by Daniel-Constantin Mierla 21 Apr '17

21 Apr '17

Hello, with a bit of delay, given that had to accommodate and accept some requests from the last year queue, I am glad to announces that we can offer two more seats at Kamailio World Conference, May 8-10, 2017, in Berlin. Therefore we are continuing the program from last years, based on the roots and the tight relation of Kamailio project with the academic environment, the eligible people being students enrolled in universities or research institutes (both bachelor and PhD programs qualify) as well as people from underrepresented groups. If you think you are eligible and want to participate, email me directly (miconda(a)gmail.com) or to registration(a)kamailio.org . Participation to all the content of the event (workshops, conference and social event) is free, but you will have to take care of expenses for traveling and accommodation. Write a short description about your interest in real time communications and, when it is the case what is the university or the research institute you are affiliate to. Also, if you are not a student, but you are in touch with some or have access to students forums/mailing lists, it will be very appreciated if you forward these details. Time is short, we already received few requests based on my remark in a message sent out few weeks ago, therefore in a matter of days we will do the selection, so you have to hurry up a bit. More information about Kamailio World is available on the web site: - http://www.kamailioworld.com Many thanks to the event sponsors that allowed to continue this program, respectively: FhG Fokus, Asipto, Flowroute, Telnyx, Sipwise, Sipgate, Simwood, Obihai, NG-Voice, Evariste Systems, Digium, VoiceTel, Pascom and Core Network Dynamics. Expect a full house event! Looking forward to meeting many of you in Berlin! 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 Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

New developer: Guillaume Bour
by Daniel-Constantin Mierla 21 Apr '17

21 Apr '17

Hello, I would like to announce that Guillaume Bour (https://github.com/gbour) has now developer privileges on Kamailio's gihub project. He has contributed recently a new module named keepalive: - http://kamailio.org/docs/modules/devel/modules/keepalive.html There is also a pull request from him waiting to be merge to drouting module, adding capability of detecting active/inactive gateways used by the module. Thanks for the contributions so far and looking forward to collaborate in the future! 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 Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

http_async_query in event_route[dialog:start] does not work
by Pranathi Venkatayogi 21 Apr '17

21 Apr '17

Hi, I wanted to make an http_async_query in event_route[dialog:start] after the call the confirmed. I see that flow does not even get to making "async_send_query". Perhaps it is failing at the first line "if (msg==null)" in w_http_async_post as I don't see any other error messages from the module. I can confirm that my setup and api server are fine as other calls I make from request_route and event_route[dialog:end] are working. I am attaching herewith my config files and Kamailio.log file. Relevant prob spot is below- Apr 20 17:37:17 devsip00 /sbin/kamailio[6286]: {2 3094 INVITE d9695149ce484811ab98ca41a4e8c694} INFO: <script>: Translation: TextBroker - Making CallAnswered for d9695149ce484811ab98ca41a4e8c694 If you notice we don't see any further progress of this call in http_async_client module. Can someone suggest what I am missing/where else can I know for sure dialog is confirmed and make this call. Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 1

Kamailio 5.0 doesn't forward Authorization (Digest username) to Asterisk
by gh 21 Apr '17

21 Apr '17

Hi Marko, i work on the same scenario. It should work with: if(uac_reg_request_to("$fU", 0)) { xlog("L_NOTICE", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); t_relay(); } But in my case xlog() currently does not print out the variables, so i am not able to debug:-( Can you try this code and tell me if xlog() works for you? Best regards Georg

3 7

Planning next IRC devel meeting - Apr 20, 2017
by Daniel-Constantin Mierla 20 Apr '17

20 Apr '17

Hello, probably it is the time to have another IRC devel meeting to discuss the plans for the project and the roadmap to next major release, v5.1. First proposed date would be Apr 20, with secondary options for Apr 24 or 25, but other dates can be taken in consideration if more convenient for a consistent group of developers. I created the wiki page for it at: * https://www.kamailio.org/wiki/devel/irc-meetings/2017a Feel free to add there the subjects you want to discuss, I also have several other topics to add. It would be good to list yourself if you are going to participate (or want to participate) in a specific day in order to see who are we relying on being present. 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 Conference - May 8-10, 2017 - www.kamailioworld.com

1 4

Reply to Incoming SIP Options Pings from Kamailio
by SIP Tech 20 Apr '17

20 Apr '17

Hello, I have a simple Kamailio server built that acts as a stateless load balancer for incoming SIP calls. All kamailio does here is receive the incoming call and route it round robin to a Freeswitch machine behind it. All configuration in Kamailio is done in local config files including the use of dispatcher.list. I have an external 3rd party partner sending my Kamailio machine SIP options pings. Because my Kamailio sends all incoming calls stateless to a FS machine behind it (round robin) the FS machines behind Kamailio get the incoming ping requests and generate the 200 OK reply back to kamailio. I would like the 3rd party to get a 200 OK for pings directly from my Kamailio and not from freeswitch behing it. I have the need to scale FS machines on the fly so I don't want FS machines to be responsible for the sip options ping replies. I want Kamailio itself to reply with a static 200 OK to incoming ping requests. I have looked into nathelper and siputils options_reply to see if I could get a simple static config to handle this 200 OK directly from Kamailio. I cant get it to fully work as the incoming sip pings still hit freeswitch. How can I configure this in the kamailio.cfg to get kamailio itself to reply with a 200 OK to SIP Options Pings and NOT forward the ping request to freeswitch? Here is my current simple stateless LB config in kamailio that works great for what it is and for what I need for loadbalancing incoming call requests. route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "4"); forward(); #stateless forward # t_relay(); #stateful relay } Here is a sample of what the incoming pings look like that hit Kamailio. U 2017/04/19 21:21:00.89 192.168.168.168:5060 -> 172.31.31.31:5060 OPTIONS sip:172.31.31.31:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.168.168:5060;branch=z9hG. Call-ID: 536f9b34c(a)192.168.168.168. To: sip:ping@172.31.31.31. From: <sip:ping@192.168.168.168>;tag=2b769f. Max-Forwards: 70. CSeq: 91160 OPTIONS. Route: <sip:172.31.31.31:5060;lr>. Using the above info what would a route look like in kamailio.cfg for this? What specific modules need to be installed/enabled to make your suggested route functional? Thanks guys

2 1

How to print out all variables in case we reach this line?
by gh 19 Apr '17

19 Apr '17

If i call a mobil number: t: <sip:+4917625251111@sip.heissa.de>. i get 404, but would expect rewrite to carrier route: sip:+4917625251111@sip.nexxtmobile.de Is there a way to print out all variables at this moment? ########################################################### if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404", "Not here"); #How to print out all variables in case we reach this line??? exit; ###########################################################

4 11

to tag length
by Diego Nadares 19 Apr '17

19 Apr '17

Hi guys, Does exist a max length for to tag? Because I'm saving dialogs in mysql and yesterday a message had a to tag greater than varchar 64. db_mysql_submit_query(): driver error on query: Data too long for column 'to_tag' at row 1 The length of the to tag was 65 (SDu81l799-7f56830f3e18-100007f-13c4-6009-73d4c9b-10f3d231-73d4c9b) Cheers, Diego

2 1

Re: [SR-Users] problem with avpops and mongodb modules
by hdssdsdsdsfsdf hdssdsdsdsfsdf 18 Apr '17

18 Apr '17

>On 22/02/2017 14:35, hsdsdssdfsdf dsasd wrote: >> I implemented call forwarding according to >> https://www.kamailio.org/wiki/tutorials/mini-howto-admin/call_forwarding, >> which is working fine. When I replace db_mysql with db_mongodb >> however, I receive an error on "if (avp_db_load("$ruri/username", >> "$avp(s:fwd_blind)")) {": >> >> ERROR: db_mongodb [mongodb_dbase.c:381]: db_mongodb_get_columns(): >> field [attribute] not found in result iterator >> ERROR: db_mongodb [mongodb_dbase.c:739]: db_mongodb_store_result(): >> failed to set the columns >> ERROR: db_mongodb [mongodb_dbase.c:918]: db_mongodb_query(): failed to >> store result >> ERROR: db_mongodb [mongodb_dbase.c:927]: db_mongodb_query(): failed to >> do the query >> ERROR: avpops [avpops_impl.c:381]: ops_dbload_avps(): db_load failed >can you set debug=3 in kamailio.cfg, restart kamailio and test again? >Then grab the log messages from syslog and send them here. It should >provide more debug information. > >Is there a record for this query? Or should be an empty response (no >record returned)? > >Cheers, >Daniel I used the following to store a forward: avp_db_store("$from/username","$avp(s:fwd_blind)"); Which in this case resulted in (after replacing the true forward info): > db.usr_preferences.find() { "_id" : ObjectId("58ac40f7e498420e0703a2e1"), "uuid" : "", "attribute" : "fwd_blind", "value" : "sip:0003@x.x.x;user=phone", "type" : 0, "username" : "0002", "domain" : null } After setting: debug=3 ... xlog("bla00"); if (avp_db_load("$ruri/username", "$avp(s:fwd_blind)")) { xlog("bla01" ... } xlog("bla02"); ... The syslog shows (after replacing the true username): Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: ERROR: <script>: bla00 Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: db_mongodb [mongodb_dbase.c:851]: db_mongodb_query(): query to collection [usr_preferences] Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: db_mongodb [mongodb_dbase.c:885]: db_mongodb_query(): query filter: { "username" : "0002", "attribute" : "fwd_blind" } Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: db_mongodb [mongodb_dbase.c:904]: db_mongodb_query(): columns filter: { "value" : 1, "attribute" : 1, "type" : 1 } Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: <core> [db_res.c:116]: db_new_result(): allocate 56 bytes for result set at 0x7fa03e54a6a0 Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: db_mongodb [mongodb_dbase.c:349]: db_mongodb_get_columns(): 3 columns returned from the query Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: <core> [db_res.c:154]: db_allocate_columns(): allocate 24 bytes for result names at 0x7fa03e54a798 Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 12 bytes for result types at 0x7fa03e549848 Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: db_mongodb [mongodb_dbase.c:377]: db_mongodb_get_columns(): Found a field[0] named: value Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: db_mongodb [mongodb_dbase.c:396]: db_mongodb_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7fa03e549798 Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: db_mongodb [mongodb_dbase.c:431]: db_mongodb_get_columns(): use DB1_STRING result type Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: db_mongodb [mongodb_dbase.c:460]: db_mongodb_get_columns(): RES_NAMES(0x7fa03e549798)[0]=[value] (2) Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: db_mongodb [mongodb_dbase.c:377]: db_mongodb_get_columns(): Found a field[1] named: attribute Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: ERROR: db_mongodb [mongodb_dbase.c:381]: db_mongodb_get_columns(): field [attribute] not found in result iterator Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: ERROR: db_mongodb [mongodb_dbase.c:739]: db_mongodb_store_result(): failed to set the columns Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: ERROR: db_mongodb [mongodb_dbase.c:918]: db_mongodb_query(): failed to store result Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: ERROR: db_mongodb [mongodb_dbase.c:927]: db_mongodb_query(): failed to do the query Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: <core> [db_res.c:79]: db_free_columns(): freeing 3 columns Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[0] at 0x7fa03e549798 Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: <core> [db_res.c:92]: db_free_columns(): freeing result names at 0x7fa03e54a798 Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: <core> [db_res.c:97]: db_free_columns(): freeing result types at 0x7fa03e549848 Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: DEBUG: <core> [db_res.c:134]: db_free_result(): freeing result set at 0x7fa03e54a6a0 Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: ERROR: avpops [avpops_impl.c:381]: ops_dbload_avps(): db_load failed Feb 24 10:42:18 slb04 /usr/local/sbin/kamailio[20311]: ERROR: <script>: bla02 When changing to db_mysql the error does not occur. I got the following record there: MariaDB [kamailio]> select * from usr_preferences; +----+------+-------------+--------+-----------+------+-------------------------------------------+---------------------+ | id | uuid | username | domain | attribute | type | value | last_modified | +----+------+-------------+--------+-----------+------+-------------------------------------------+---------------------+ | 26 | | 0002 | | fwd_blind | 0 | sip:0003@x.x.x;user=phone | 1900-01-01 00:00:01 | +----+------+-------------+--------+-----------+------+-------------------------------------------+---------------------+ With the same kamailio.cfg (except for replacing mongodb with mysql) resulting in: Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: ERROR: <script>: bla00 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:116]: db_new_result(): allocate 56 bytes for result set at 0x7fd17c349800 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:66]: db_mysql_get_columns(): 3 columns returned from the query Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:154]: db_allocate_columns(): allocate 24 bytes for result names at 0x7fd17c32f400 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 12 bytes for result types at 0x7fd17c33fae8 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:84]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7fd17c349350 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:91]: db_mysql_get_columns(): RES_NAMES(0x7fd17c349350)[0]=[value] Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:135]: db_mysql_get_columns(): use DB1_STRING result type Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:84]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[1] at 0x7fd17c349458 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:91]: db_mysql_get_columns(): RES_NAMES(0x7fd17c349458)[1]=[attribute] Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:135]: db_mysql_get_columns(): use DB1_STRING result type Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:84]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[2] at 0x7fd17c3493a8 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:91]: db_mysql_get_columns(): RES_NAMES(0x7fd17c3493a8)[2]=[type] Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: db_mysql [km_res.c:99]: db_mysql_get_columns(): use DB1_INT result type Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:184]: db_allocate_rows(): allocate 16 bytes for rows at 0x7fd17c33fb98 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 96 bytes for row values at 0x7fd17c33f778 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [sip:0003@x.x.x;user=phone] Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [fwd_blind] Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [0] Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: avpops [avpops_impl.c:132]: dbrow2avp(): db_flags=3, flags=12 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: avpops [avpops_db.c:265]: db_close_query(): close avp query Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:79]: db_free_columns(): freeing 3 columns Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[0] at 0x7fd17c349350 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[1] at 0x7fd17c349458 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[2] at 0x7fd17c3493a8 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:92]: db_free_columns(): freeing result names at 0x7fd17c32f400 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:97]: db_free_columns(): freeing result types at 0x7fd17c33fae8 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:52]: db_free_rows(): freeing 1 rows Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7fd17c33f778 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:60]: db_free_rows(): freeing rows at 0x7fd17c33fb98 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: <core> [db_res.c:134]: db_free_result(): freeing result set at 0x7fd17c349800 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: DEBUG: avpops [avpops_impl.c:415]: ops_dbload_avps(): loaded avps = 1 Feb 24 11:27:51 slb04 /usr/local/sbin/kamailio[21056]: ERROR: <script>: bla01

2 5

Re: [SR-Users] IPV6 kamailio 4.0.2
by Angel Elena 18 Apr '17

18 Apr '17

Ok, thanks Daniel. Tonight, I will activate ipv6 again and review the kamailio process with ps auxw. when I make a /etc/init.d/networking restart , kamailio still working and processing calls without problems (this night I make the test). thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ -------------------------------- -----Mensaje original----- De: Daniel-Constantin Mierla <miconda(a)gmail.com> Enviado: Mar 18-04-2017 14:50 Asunto: Re: [SR-Users] IPV6 kamailio 4.0.2 Para: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; > Hello, > > > > I think there is some problem and kamailio is not actually running after you > start it so the fifo file is created. > > > > Can you do 'ps auxw | grep kamailio' after you start kamailio with ipv6 and see > if it is running? > > > Cheers, > Daniel > > > On 17.04.17 16:20, Angel Elena wrote: > > > > Hello, good moorning. > > I have a Kamailio 4.0.2 with debian 7.0 and this days I'm configuring dual > stack ipv4 & ipv6 in all servers. > > I have configured ipv6 today in my kamailio server and when I make a > /etc/init.d/networking restart, everything seems to be working fine (ipv6 > works), but when I make a "kamctl mi dlg_list", the next error appears: > > root@sipproxy:/etc/network# kamctl mi dlg_list > ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > "/tmp/kamailio_fifo")' in your config > ERROR: and also have loaded the mi_fifo module. > > Disable ipv6 and now works fine. > > > restart kamailio service is "complicated" to test anythink. > > Any ideas ? > > thanks > > -------------------------------- > Ángel Elena Medina _o) > craem(a)craem.net / \\ > http://blog.craem.net _(___V > @craem_ > -------------------------------- > > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > >

1 0

IPV6 kamailio 4.0.2
by Angel Elena 18 Apr '17

18 Apr '17

Hello, good moorning. I have a Kamailio 4.0.2 with debian 7.0 and this days I'm configuring dual stack ipv4 & ipv6 in all servers. I have configured ipv6 today in my kamailio server and when I make a /etc/init.d/networking restart, everything seems to be working fine (ipv6 works), but when I make a "kamctl mi dlg_list", the next error appears: root@sipproxy:/etc/network# kamctl mi dlg_list ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. Disable ipv6 and now works fine. restart kamailio service is "complicated" to test anythink. Any ideas ? thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ --------------------------------

2 1

too long mysql table key
by Juha Heinanen 15 Apr '17

15 Apr '17

I tried to install kamailio mysql db on Debian Stretch. Create of domain_attrs table failed with this kind of error message: ERROR 1071 (42000) at line 26: Specified key was too long; max key length is 767 bytes domain-create.sql contains this: CREATE TABLE `domain_attrs` ( `id` INT(10) UNSIGNED AUTO_INCREMENT PRIMARY KEY NOT NULL, `did` VARCHAR(64) NOT NULL, `name` VARCHAR(32) NOT NULL, `type` INT UNSIGNED NOT NULL, `value` VARCHAR(255) NOT NULL, `last_modified` DATETIME DEFAULT '2000-01-01 00:00:01' NOT NULL, CONSTRAINT domain_attrs_idx UNIQUE (`did`, `name`, `value`) ); Looks like default charset is something (e.g. utf8), where one char takes more than 1 byte and 767 limit is exceeded. Any suggestions on how to solve this? -- Juha

2 9

kamcli doesn't function
by Marko Tirs 14 Apr '17

14 Apr '17

Hello all, I can't find any answer why my kamcli doesn't function. I'm using Kamailio 5.0.0 at the Debian Jessy 8.7 and it functions well. # kamctl ul show functions well but when I use kamcli I get error: # kamcli ul show 13(7452) ERROR: jsonrpcs [jsonrpcs_sock.c:542]: jsonrpc_dgram_send_data(): failed to send the response - ret: -1, len: 2443 (131072), err: 1 - Operation not permitted) (log): Timeout receiving response on unix sock The relevant parts of kamailio.cfg: #------------------------------------------------------------------------ #!define WITH_DEBUG #!define WITH_NAT #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB loadmodule "jsonrpcs.so" # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") #-------------------------------------------------------------------------- # ll /var/run/kamailio/ total 0 srw------- 1 root root 0 Mar 22 11:36 kamailio_ctl prw-rw---- 1 root root 0 Mar 22 12:30 kamailio_rpc.fifo srw-rw---- 1 root root 0 Mar 22 11:36 kamailio_rpc.sock Can somebody tell me what is the problem? Thank you! Regards Marko

3 9

RFC: kamailio-announce mailing list
by Daniel-Constantin Mierla 14 Apr '17

14 Apr '17

Hello, for a while in the past (like 2008-2011) the kamailio-announce(a)lists.kamailio.org mailing list was used for very sporadic announcements: - https://lists.kamailio.org/cgi-bin/mailman/listinfo/kamailio-announce It was intended to be used to announce only relevant news about the project (releases, major events, etc.), which most of them go also on sr-users, but targeting community members than which prefer to avoid high traffic, which is specific for the usual discussions here. Also, for people in sr-users, can be a place to spot easier the major announcements. Because we migrated back to lists.kamailio.org domain, this list is now presented on the page of mailman index: - https://lists.kamailio.org/cgi-bin/mailman/listinfo I just marked it as 'inactive', but I am wondering if it should be revived somehow (as some people wanted to have a place where to inform about relevant security issues, etc...) or be closed (removed/made private). Even it was not used/promoted for more than 5 years, I noticed that it has over 200 subscribers, therefore it seems to be enough people interested in it. Reply here if you consider it to be useful and eventually give some suggestions for what kind of announcements should be used. Note that there is already business(a)lists.kamailio.org (with larger subscriber base) for business related announcement or discussions. 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 Conference - May 8-10, 2017 - www.kamailioworld.com

2 1

Using advertised_addres doesn't update the RR header
by Mridul B 13 Apr '17

13 Apr '17

Hi, Using *advertised_address *core parameter in the kamailio script doesn't update the address mentioned in the Record-Route header. We see this problem in the 200 OK message (of the INVITE msg), where the Record-Route header sent by the Kamailio contains socket address. We're using kamailio 4.4.5 (x86_64/linux) 239f3c Can you please tell if this core parameter is effective for changing the RR header ? Thanks Best regards, *MRIDUL*

3 3

How do i build MSRP chat bot?
by Pranathi Venkatayogi 13 Apr '17

13 Apr '17

I would like to place the customer connecting via SIP/MSRP on hold until I find the right destination. What is the best way to do this? One idea is to write a MSRP bot. Does anyone know how to go about this? Other smart ideas? Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

3 4

Install 5.0 with IMS module on single machine
by Jegadeesan 13 Apr '17

13 Apr '17

Hi , How to install Kamalio 5.0 with IMS module on single machine to support the RCS client development ? -- Jega

2 2

Rtpengine module. rtcp-mux-demux option does not work.
by Denys Pozniak 13 Apr '17

13 Apr '17

Hello, All! We are using Kamailio as Webrtc to SIP proxy. Clinet's Chrome browser needs to have rtcp-mux option to work correct, so we are adding it in RTPENGINE->WS direction as: rtpengine_manage("trust-address replace-origin replace-session-connection *rtcp-mux-offer rtcp-mux-accept* ICE=force RTP/SAVPF"); Also we need to remove it in RTPENGINE->PSTN direction, we are trying to do it like this: rtpengine_manage("trust-address replace-origin replace-session-connection * rtcp-mux-demux* ICE=remove RTP/AVP"); But it does not work for us and a=rtcp-mux option in SDP goes to the SIP provider. Please help us to solve this! BR, Denys PS. Versions: Kamailio 4.4.5, rtpengine 5.2.1.0

2 1

RAW_SOCKS with Kamaili 4.4.5
by Yasin CANER 13 Apr '17

13 Apr '17

3 6

Kamailio 5.0 can't get version of the RTP proxy
by Yebe diomande Keffa 13 Apr '17

13 Apr '17

Hello Getting this issu on Kamaiio 5.0 ( install from GIT): Apr 10 10:47:42 ns3033789 /usr/local/sbin/kamailio[11496]: WARNING: rtpproxy [rtpproxy.c:1366]: rtpp_test(): can't get version of the RTP proxy Apr 10 10:47:42 ns3033789 /usr/local/sbin/kamailio[11488]: WARNING: rtpproxy [rtpproxy.c:1403]: rtpp_test(): support for RTP proxy <udp:127.0.0.1:7722> has been disabled temporarily I done install of RTPproxy with: $ git clone -b master https://github.com/sippy/rtpproxy.git $ git -C rtpproxy submodule update --init --recursive $ cd rtpproxy $ ./configure $ make $ make install $ rtpproxy -l 5.135.177.203 -s udp:localhost:7722 Before i install mysql-server & kamdbctl create (rtpproxy table), it was working fine Someone to advise please! Keffa Y. Diomande Skype : keffadi Tel : +225 02 77 22 22 Office : Abidjan Ciad http://www.alcall.net

3 5

New developer: Mikko Lehto
by Daniel-Constantin Mierla 12 Apr '17

12 Apr '17

Hello, just to let everyone know that I invited Mikko Lehto into the developer group on Kamailio gihub project. He has made a lot of good pull requests over the time, specially to documentation and unit tests that could have been just committed directly without delay on waiting for a review, saving also time from the other developers. Taking time to express my thanks for all his contributions so far and reviving those unit tests, and I am looking forward to future collaboration! 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 Conference - May 8-10, 2017 - www.kamailioworld.com

3 2

Use mysql view to load data into htable
by Jan Hazenberg 12 Apr '17

12 Apr '17

Hello, I'm trying to load data from a mysql table into a htable using a mysql view. But when i start kamailio i get the following error: ERROR: htable [ht_db.c:345]: ht_db_load_table(): Wrong db type [3] for key_type column Here a example of a view i'm trying to use: CREATE OR REPLACE VIEW `view_blacklist_prefix` AS SELECT prefix AS key_name, '0' AS key_type, '0' AS value_type, '1' AS key_value, '0' AS expires FROM openser.blacklist_prefix GO Should it be possible to use a mysql view as input for a htable? Thanks, Jan Hazenberg

2 2

snmpstats issue
by Igor Potjevlesch 12 Apr '17

12 Apr '17

Hello, On one of my Kamailio server, I don't have data returned by KAMAILIO-MIB::. KAMAILIO-SIP-COMMON-MIB and KAMAILIO-SIP-SERVER-MIB look to work good. Everything is configured as the other Kamailio instances, but there is nothing about current dialog even if the module is loaded into Kamailio and called into kamailio.cfg. Any idea? Here is an output of snmpwalk: snmpwalk -c legosnoc -v 1 <HOST> 1.3.6.1.4.1.34352 KAMAILIO-SIP-COMMON-MIB::kamailioSIPProtocolVersion.0 = STRING: SIP/2.0 KAMAILIO-SIP-COMMON-MIB::kamailioSIPServiceStartTime.0 = Timeticks: (0) 0:00:00.00 KAMAILIO-SIP-COMMON-MIB::kamailioSIPEntityType.0 = BITS: 20 proxyServer(2) KAMAILIO-SIP-COMMON-MIB::kamailioSIPTransportRcv.ipv4."IP_HOST".5060 = BITS: 40 udp(1) KAMAILIO-SIP-COMMON-MIB::kamailioSIPTransportRcv.ipv4."IP_HOST".5060 = BITS: 40 udp(1) KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.1 = STRING: METHOD_INVITE KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.2 = STRING: METHOD_CANCEL KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.3 = STRING: METHOD_ACK KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.4 = STRING: METHOD_BYE KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.5 = STRING: METHOD_INFO KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.6 = STRING: METHOD_OPTIONS KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.7 = STRING: METHOD_UPDATE KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.8 = STRING: METHOD_REGISTER KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.9 = STRING: METHOD_MESSAGE KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.10 = STRING: METHOD_SUBSCRIBE KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.11 = STRING: METHOD_NOTIFY KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.12 = STRING: METHOD_PRACK KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.13 = STRING: METHOD_REFER KAMAILIO-SIP-COMMON-MIB::kamailioSIPMethodName.14 = STRING: METHOD_PUBLISH KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryInRequests.0 = Counter32: 2680 KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryOutRequests.0 = Counter32: 1050 KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryInResponses.0 = Counter32: 3102 KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryOutResponses.0 = Counter32: 1630 KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryTotalTransactions.0 = Counter32: 1518 KAMAILIO-SIP-COMMON-MIB::kamailioSIPCurrentTransactions.0 = Gauge32: 7 KAMAILIO-SIP-COMMON-MIB::kamailioSIPNumUnsupportedUris.0 = Counter32: 16 KAMAILIO-SIP-COMMON-MIB::kamailioSIPNumUnsupportedMethods.0 = Counter32: 0 KAMAILIO-SIP-COMMON-MIB::kamailioSIPOtherwiseDiscardedMsgs.0 = Counter32: 0 KAMAILIO-SIP-SERVER-MIB::kamailioSIPProxyStatefulness.0 = INTEGER: callStateful(3) KAMAILIO-SIP-SERVER-MIB::kamailioSIPProxyRecordRoute.0 = INTEGER: true(1) KAMAILIO-SIP-SERVER-MIB::kamailioSIPProxyAuthMethod.0 = BITS: 20 digest(2) KAMAILIO-SIP-SERVER-MIB::kamailioSIPNumProxyRequireFailures.0 = Counter32: 0 KAMAILIO-SIP-SERVER-MIB::kamailioSIPRegMaxContactExpiryDuration.0 = Gauge32: 3600 seconds KAMAILIO-SIP-SERVER-MIB::kamailioSIPRegMaxUsers.0 = Gauge32: 4294967295 KAMAILIO-SIP-SERVER-MIB::kamailioSIPRegCurrentUsers.0 = Gauge32: 325416 KAMAILIO-SIP-SERVER-MIB::kamailioSIPRegDfltRegActiveInterval.0 = Gauge32: 3600 seconds KAMAILIO-SIP-SERVER-MIB::kamailioSIPUserUri.1 = STRING: DefaultUser KAMAILIO-SIP-SERVER-MIB::kamailioSIPUserAuthenticationFailures.1 = Counter32: 0 KAMAILIO-SIP-SERVER-MIB::kamailioSIPContactDisplayName.1.1 = STRING: DefaultUser KAMAILIO-SIP-SERVER-MIB::kamailioSIPContactURI.1.1 = STRING: DefaultUser KAMAILIO-SIP-SERVER-MIB::kamailioSIPContactLastUpdated.1.1 = STRING: 0-0-0,0:0:0.0 KAMAILIO-SIP-SERVER-MIB::kamailioSIPContactExpiry.1.1 = STRING: 0-0-0,0:0:0.0 KAMAILIO-SIP-SERVER-MIB::kamailioSIPContactPreference.1.1 = STRING: -0.01 KAMAILIO-SIP-SERVER-MIB::kamailioSIPRegUserLookupCounter.0 = Counter32: 4 KAMAILIO-SIP-SERVER-MIB::kamailioSIPRegAcceptedRegistrations.0 = Counter32: 21 KAMAILIO-SIP-SERVER-MIB::kamailioSIPRegRejectedRegistrations.0 = Counter32: 0 KAMAILIO-MIB::kamailioSrvMaxMemory.0 = Gauge32: 738197504 KAMAILIO-MIB::kamailioSrvFreeMemory.0 = Gauge32: 358262600 KAMAILIO-MIB::kamailioSrvMaxUsed.0 = Gauge32: 380013984 KAMAILIO-MIB::kamailioSrvRealUsed.0 = Gauge32: 379934904 KAMAILIO-MIB::kamailioSrvMemFragments.0 = Gauge32: 70 KAMAILIO-MIB::kamailioSrvCnfFullVersion.0 = STRING: kamailio 4.2.8 (x86_64/linux) 4507b8 KAMAILIO-MIB::kamailioSrvCnfVerName.0 = STRING: kamailio KAMAILIO-MIB::kamailioSrvCnfVerVersion.0 = STRING: 4.2.8 KAMAILIO-MIB::kamailioSrvCnfVerArch.0 = STRING: x86_64 KAMAILIO-MIB::kamailioSrvCnfVerOs.0 = STRING: linux KAMAILIO-MIB::kamailioSrvCnfVerId.0 = STRING: 4507b8 KAMAILIO-MIB::kamailioSrvCnfVerCompTime.0 = STRING: 12:34:45 Feb 17 2017 KAMAILIO-MIB::kamailioSrvCnfVerCompiler.0 = STRING: gcc 4.4.6 KAMAILIO-MIB::kamailioSrvCnfVerFlags.0 = STRING: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES KAMAILIO-MIB::kamailioMsgQueueDepth.0 = Gauge32: 0 KAMAILIO-MIB::kamailioMsgQueueMinorThreshold.0 = INTEGER: -1 KAMAILIO-MIB::kamailioMsgQueueMajorThreshold.0 = INTEGER: -1 KAMAILIO-MIB::kamailioMsgQueueDepthAlarmStatus.0 = BITS: 00 KAMAILIO-MIB::kamailioMsgQueueDepthMinorAlarm.0 = INTEGER: clear(0) KAMAILIO-MIB::kamailioMsgQueueDepthMajorAlarm.0 = INTEGER: clear(0) KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 KAMAILIO-MIB::kamailioDialogLimitMinorThreshold.0 = INTEGER: -1 KAMAILIO-MIB::kamailioDialogLimitMajorThreshold.0 = INTEGER: -1 KAMAILIO-MIB::kamailioTotalNumDialogSetups.0 = Counter32: 0 KAMAILIO-MIB::kamailioDialogUsageState.0 = INTEGER: idle(0) KAMAILIO-MIB::kamailioDialogLimitAlarmStatus.0 = BITS: 00 KAMAILIO-MIB::kamailioDialogLimitMinorAlarm.0 = INTEGER: clear(0) KAMAILIO-MIB::kamailioDialogLimitMajorAlarm.0 = INTEGER: clear(0) KAMAILIO-MIB::kamailioNetTcpConnEstablished.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetTcpConnFailed.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetTcpConnReset.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetTcpConnSuccess.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetTcpConnOpen.0 = Gauge32: 0 KAMAILIO-MIB::kamailioNetTcpConnPassiveOpen.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetTcpConnReject.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetTcpEnabled.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetTcpMaxConns.0 = INTEGER: 2048 KAMAILIO-MIB::kamailioNetTcpConnTimeout.0 = INTEGER: 10 KAMAILIO-MIB::kamailioNetTcpSendTimeout.0 = INTEGER: 160 KAMAILIO-MIB::kamailioNetTcpConnLifetime.0 = INTEGER: 1920 KAMAILIO-MIB::kamailioNetTcpNoConnect.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetTcpFdCache.0 = INTEGER: 1920 KAMAILIO-MIB::kamailioNetTcpAsync.0 = INTEGER: true(1) KAMAILIO-MIB::kamailioNetTcpAsyncConnWait.0 = INTEGER: true(1) KAMAILIO-MIB::kamailioNetTcpAsyncConnWqMax.0 = INTEGER: 32768 KAMAILIO-MIB::kamailioNetTcpAsyncWqBlkSize.0 = INTEGER: 10485760 KAMAILIO-MIB::kamailioNetTcpRdBufSize.0 = INTEGER: 16384 KAMAILIO-MIB::kamailioNetTcpDeferAccept.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetTcpDelayedAck.0 = INTEGER: true(1) KAMAILIO-MIB::kamailioNetTcpSynCnt.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetTcpLinger.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetTcpKeepAlive.0 = INTEGER: true(1) KAMAILIO-MIB::kamailioNetTcpKeepIdle.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetTcpKeepIntvl.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetTcpKeepCnt.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetTcpCrlfPing.0 = INTEGER: true(1) KAMAILIO-MIB::kamailioNetTcpAcceptAliases.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetTcpAcceptNoCl.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetWsConnsActive.0 = Gauge32: 0 KAMAILIO-MIB::kamailioNetWsConnsActiveMax.0 = Gauge32: 0 KAMAILIO-MIB::kamailioNetWsConnsFailed.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetWsConnsClosedLocal.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetWsConnsClosedRemote.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetWsFramesRx.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetWsFramesTx.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetWsHandshakeSuccess.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetWsHandshakeFailed.0 = Counter32: 0 KAMAILIO-MIB::kamailioNetConfUdpTransport.0 = INTEGER: true(1) KAMAILIO-MIB::kamailioNetConfUdpDtlsTransport.0 = INTEGER: -1 KAMAILIO-MIB::kamailioNetConfTcpTransport.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetConfTcpTlsTransport.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetConfSctpTransport.0 = INTEGER: true(1) KAMAILIO-MIB::kamailioNetConfSctpTlsTransport.0 = INTEGER: -1 KAMAILIO-MIB::kamailioNetConfWsTransport.0 = INTEGER: -1 KAMAILIO-MIB::kamailioNetConfWsTlsTransport.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetConfHttpServer.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetConfMsrpRelay.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetConfStunServer.0 = INTEGER: 0 KAMAILIO-MIB::kamailioNetConfOutbound.0 = INTEGER: 0 Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 7

issue with sip path support between asterisk and kamailio
by gerry kernan 12 Apr '17

12 Apr '17

Hi Thanks in advanced if anyone can help me on this. I have kamailio -4.2.3-1.1 running in front of asterisk 13.13.1. I'm using Path support with asterisk so that I can forward Sip requests via kamailio to the asterisk server running on a LAN IP. Kamailio is configured with a WAN and LAN IP. All works well I can register a sip endpoint to the asterisk server via kamailio and make an receive calls, SIP OPTIONs from the asterisk server are received and relayed by kamailio to endpoints. But I'm having issues when a call is hung up asterisk appears to be trying to send the BYE via its own IP address rather that the IP of the LAN interface on the kamailio server. Should I be looking at the issue being on asterisk as it is not sending the BYE via correct IP? Y.Y.Y.Y = WAN IP of client endpoint 192.10.10.216 = LAN IP of asterisk Example of BYE from sip debug on asterisk --- (10 headers 0 lines) --- Really destroying SIP dialog '5d3ae6944a08b42c2078e237141e0e40@192.10.10.216:5060' Method: OPTIONS -- <SIP/1001-00000011> Playing 'digits/17.ulaw' (language 'en') -- Executing [*250@cos-all:8] Wait("SIP/1001-00000011", "1") in new stack -- Executing [*250@cos-all:9] SayUnixTime("SIP/1001-00000011", ",,IMp") in new stack -- <SIP/1001-00000011> Playing 'digits/4.ulaw' (language 'en') -- <SIP/1001-00000011> Playing 'digits/oh.ulaw' (language 'en') -- <SIP/1001-00000011> Playing 'digits/4.ulaw' (language 'en') -- <SIP/1001-00000011> Playing 'digits/p-m.ulaw' (language 'en') -- Executing [*250@cos-all:10] Hangup("SIP/1001-00000011", "") in new stack == Spawn extension (cos-all, *250, 10) exited non-zero on 'SIP/1001-00000011' Scheduling destruction of SIP dialog 'OGRjYzJlYmE5MDhjYTdkMTM1ZTAxNTk2YWYwNDg5MTA.' in 6400 ms (Method: ACK) set_destination: Parsing <sip:1001@Y.Y.Y.Y:16078;transport=UDP> for address/port to send to set_destination: set destination to Y.Y.Y.Y:16078 Reliably Transmitting (no NAT) to Y.Y.Y.Y:16078: BYE sip:1001@Y.Y.Y.Y:16078;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.10.10.216:5060;branch=z9hG4bK46ed8549 Max-Forwards: 70 From: <sip:*250@sip.domain.ie;transport=UDP>;tag=as60c5965b To: <sip:1001@sip.domain.ie;transport=UDP>;tag=471c3a31 Call-ID: OGRjYzJlYmE5MDhjYTdkMTM1ZTAxNTk2YWYwNDg5MTA. CSeq: 102 BYE User-Agent: OMBUTEL PBX Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:sip.domain.ie", nonce="4aaa41f3", response="136fac72a799f9507a7466b578786110" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 KAMAILIO LAN IP = 192.10.10.202 SIP status of peer on asterisk * Name : 1001 Description : Infinity-Test Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : cos-all Record On feature : automon Record Off feature : automon Subscr.Cont. : <Not set> Language : en Tonezone : uk AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : ringback Mailbox : 1001@ombutel-voicemail VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "Infinity-Test" <1001> MaxCallBR : 384 kbps Expire : 3228 Insecure : no Force rport : No Symmetric RTP: No ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID : Yes Path support : Yes Path : <sip:192.10.10.202;lr;received=sip:Y.Y.Y.Y:16078> TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.200.122:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1001 SIP Options : norefersub replaces replace timer Codecs : (ulaw|alaw|g729) Auto-Framing : No Status : OK (19 ms) Useragent : Zoiper for Windows 2.43 r24984 Reg. Contact : sip:1001@192.168.200.122:5060;rinstance=3798b50d10d60d84;transport=UDP Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

2 2

WAN ITSP + NAT can someone help me ?
by Martin Lapointe 12 Apr '17

12 Apr '17

hi all, did someone can help me for with some configuration im a litle lost with inbount trafic with kamailio i need to redirect inbound conection from itsp (over wan) to an internal PBX but it seem nat is not working (I do the test with jitsi on my desktop and sip connection seem initiate (200 ok) but no sound ) like contact information is not update on sip header can someone help me a bit MARTIN LAPOINTE Spécialiste en infrastructure technologique T (418) 543-2600 x2119 F (418) 543-0363 mlapointe(a)inso.ca<mailto:mlapointe@inso.ca> www.inso.ca<http://www.inso.ca/> 365 Racine Est #204 Chicoutimi, QC G7H 1S8 [signature] Ce message est transmis au destinataire par courriel, même si ce moyen de transmission est non-sécuritaire. En conséquence, le soussigné décline toute responsabilité et ne peut garantir la confidentialité résultant de la transmission des présentes. De plus, comme ce message peut contenir de l'information de nature privilégiée et confidentielle et est à l'usage exclusif du destinataire, toute autre personne est avisée qu'il lui est strictement interdit de le divulguer ou de le copier. Si ce message a été transmis par mégarde à une autre personne que le destinataire, prière de nous en aviser immédiatement par téléphone (à frais virés) au numéro mentionné ci-dessus. Merci !

2 1

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:----------------------------------------------------------- 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(); }

2 2

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 reply here or contact me. In addition, URLs should use now https, browsing the archive with plain http is redirected to https urls. 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 Conference - May 8-10, 2017 - www.kamailioworld.com

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 difficult to know to which of kamailio's interfaces (pub or priv) is. ERROR: <core> [resolve.c:1694]: sip_hostport2su(): could not resolve hostn ame. Reconfigure those gw it's not possible. There are a lot of them and a lot of people that won't like that idea. Is possible to disable it? or What do you recommend me to do? Maybe using /etc/hosts? Any suggestion will be very appreciated. Diego.

3 10

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 across the config? Reason for the question is the possibility to use ansible as deployment tool. I set all the variables i need in a separate file and include it. This file contains the entire modconfig and global vars section to adapt this part to every single machine individually. This does not work if i need to put "constant" variables on different places in my config. If anybody has ever made this, could you please help me, or give any suggestions? Many thanks! BR Jöran -- Jöran Vinzens - vinzens(a)sipgate.de sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

4 4

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 -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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.org/wiki/tutorials/getting-started/main Based on what the tutorial says, I should have a server that only allows users that have been authenticated against the DB. The thing is that now Kamailio allows me to access, no matter what credentials I gave him (replies 200 OK to my REGISTER requests, and whows the user on the `kamctl monitor` view). I've added to this gist: * My current kamailio config * the kamailio grepped syslog * the pjsua output from when I register with an unexistent user https://gist.github.com/nicopace/9d44bf813fe309df69ff4cd0affe8324 I've already checked out some of the wiki documentation, namely: * http://www.kamailio.org/wiki/tutorials/auth/auth_db * http://www.kamailio.org/wiki/tutorials/getting-started/main Am I missing something? Thanks

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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 Networks (that uses 2G Cellular Networks) with the rest of the world via SIP, and we are considering using Kamailio for this. The project has been using FreeSwitch for some time and now we are thinking about using Kamailio on the cloud side, and intengrate them via SIP. On a not so distant future, if the Kamailio experience goes well, we are thinking on moving everything to Kamailio, as it seems to be much more simple and effective. For now we expect to do a prototype to connect this communities with their relatives via SMS (on the Cellular side) with SIP SIMPLE Messages (on the 'cloud' side). Awesome to meet you all, and hope to be collaborating with this community! Regards, [0] https://www.rhizomatica.org/

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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 name that is used out there and it will make operating everything a bit smoother, more coherent and also avoid eventual confusions for people new into the project. As a matter of fact, we even had a small incident few weeks ago -- due to split management of sip-router.org and DNS registration expire notifications going to an email address on a different server than kamailio.org, the domain was expired for few hours (although it was kept on hold by registrar) and mailing lists started to be unavailable. Fortunately that was discovered and fixed in time. The names will stay the same sr-users and sr-dev, just primary domain will be using kamailio.org (which should 'hopefully' work even now, as this email was addressed with mixed sr-dev(a)lists.sip-router.org and sr-users(a)lists.kamailio.org). If anyone has different opinions or suggestions, let's discuss them! 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 Conference - May 8-10, 2017 - www.kamailioworld.com

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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 -> CANCEL <- 481 (this is sent from my kamailio script when t_check_trans fails) I can't see much that's different between the leading requests/responses of each. They seem to be virtually identical (with the exception of IP addresses and tag values, of course). This makes sense, since they're both based on PJSIP (albeit different versions of PJSIP, but not off by much) Is there anything in particular I should be looking out for that might be obvious? I could provide more detailed logs and traces if required. Best, Colin

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"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 failed at msilo! Searching for this error turned up http://tinyurl.com/lovr4aq , but the link in the answer is broken, and it doesn't appear that STRICT_TRANS_TABLES is relevant to my version of MySQL [1], which is 5.5.54 (running on Trisquel 7, which is based on Ubuntu Trusty). Has anyone else encountered and overcome this problem, or does anyone have any ideas about how I might proceed? Tim <>< [1] http://tinyurl.com/zmnc75c

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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

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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

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