- sr-users - kamailio.org

Confusion about various TCP expiry and lifetime parameters
by M S 27 Aug '15

27 Aug '15

Hi, We have a bit of confusion regarding tcp_connecton_timeout core parameter. The documentation says, -- Lifetime in seconds for TCP sessions. TCP sessions which are inactive for longer than *tcp_connection_lifetime* will be closed by Kamailio. -- However we observe a strange behaviour. 1. The connection is NOT closed by Kamailio unless we additionally set "close_expired_tcp" parameter in usrloc module, 2. Secondly, if we set this parameter to a smaller value say 70 seconds while sip register expiry is 600 seconds (and close_expired_tcp parameter enabled in usrloc module), the connection still remains active (tested after keeping it idle for 120 seconds then sending a sip request on this connection, we expected the request to fail but it does not fails). 3. if we set this parameter to greater then sip register value, e.g. 610 seconds and set close_expired_tcp parameter in usrloc, then we observe disconnect after about 8-10 minutes. Whereas we expect it to continue since user is re-registering every 600 seconds. Can you guys explain why this is happening? What keeps a tcp connection active or makes it inactive? We are using websockets (which use TCP at lower layer) and we observe there is no websocket frame sent or received over the tcp connection, yet it remain active after tcp_connection_lifetime in case 2 above and becomes inactive in case 3. We are using Kamailio v4.3.1 SVN Rev. 4717b5 on Debian Wheezy 32bit OS. Please suggest. Thank you.

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IMS Kamailio "477 Unfortunately error on sending to next hop occurred" code sent by the PCSCF
by Victor Huertas Garcia 26 Aug '15

26 Aug '15

Hello everybody, I have tried to perform a call between alice and bob (both registered in the same SCSCF and from the same PCSCF). The INVITE goes all the way as it is supposed: Alice-->PCSCF-->SBC-->SCSCF-->ICSCF-->SCSCF-->SBC-->PCSCF. But when it reaches PCSCF on Network to User direction, this SIP proxy answers with a "477 Unfortunately error on sending to next hop occurred" code just after an exchange with the RTP Proxy (Offer-Answer). At kamailio.cfg (which it is not change from the default config) this exchange is done under Route[RTPRPROXY] in the function: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove"); Does anybody know why is complaining (see below) about the server on localhost while it is not configured so? Here you can find the logs from PCSCF: Aug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Received command 'offer' from 192.168.0.108:57522Aug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Creating new callAug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Replying to 'offer' from 192.168.0.108:57522Aug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: *** cfgtrace:request_route=[RTPPROXY] c=[/etc/kamailio/pcscf/kamailio.cfg] l=813 a=2 n=returnAug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: *** cfgtrace:request_route=[Term_Initial] c=[/etc/kamailio/pcscf/kamailio.cfg] l=1196 a=24 n=t_relayAug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f480e27ed70,1358,0,192.168.0.105:49284,16): Invalid argument(22)Aug 23 15:10:06 ims /usr/sbin/kamailio[3313]: CRITICAL: <core> [udp_server.c:555]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Here you can find the configuration from RTP proxy (/etc/default/ngcp-rtpengine-daemon) and you can see that I am not listening the localhost port at "INTERFACES": RUN_RTPENGINE=yes LISTEN_TCP=25060LISTEN_UDP=12222LISTEN_NG=22222LISTEN_CLI=9900INTERFACES="192.168.0.108"TIMEOUT=60SILENT_TIMEOUT=3600PIDFILE=/var/run/ngcp-rtpengine-daemon.pidFORK=yes# TOS=184TABLE=0# NO_FALLBACK=yes# PORT_MIN=30000# PORT_MAX=50000# REDIS=127.0.0.1:6379# REDIS_DB=1# B2B_URL=http://127.0.0.1:8090/# LOG_LEVEL=6# LOG_FACILITY=daemon# LOG_FACILITY_CDR=daemon# LOG_FACILITY_RTCP=daemon# NUM_THREADS=5# DELETE_DELAY=30# GRAPHITE=9006# GRAPHITE_INTERVAL=60# GRAPHITE_PREFIX=myownprefix# MAX_SESSIONS=5000 And finally the pcscf.cfg: # SIP / UDPlisten=udp:eth0:4060#listen=udp:127.0.0.1:4070listen=udp:127.0.0.1:4060# SIP / TCP (Monitoring)#listen=tcp:127.0.0.1:4060# SIP / TCP/TLS#listen=tls:109.239.57.200:5061# SIP / Websocket #!define MY_WS_PORT 80 listen=tcp:eth0:MY_WS_PORT # SIP / Websocket/TLS#!define MY_WSS_PORT 443#listen=tls:109.239.57.200:MY_WSS_PORT #alias=pcscf-1.imscore.orgalias=pcscf.home-domain.net#alias=proxy.imscore.org#alias=tls:"wss-proxy.imscore.org":443 # Port, where we listen to Traffic#!define PORT 4060 # NUEVO Workshop!#!define SBCPORT 5080#!define PCSCF_URL "sip:pcscf.home-domain.net"#!define TCP_PROCESSES 32 #!subst "/NETWORKNAME/home-domain.net/"#!subst "/HOSTNAME/pcscf.home-domain.net/"#!define HOSTNAME_IP pcscf.home-domain.net#!define HOSTNAME_ESC "pcscf\.home-domain\.net" # SIP-Address of capturing node, if not set, capturing is disabled.##!define CAPTURE_NODE "sip:10.1.8.55" # Allowed IPs for XML-RPC-Queries#!define XMLRPC_WHITELIST_1 "127.0.0.1"##!define XMLRPC_WHITELIST_2 "127.0.0.1"##!define XMLRPC_WHITELIST_3 "127.0.0.1" # Databases:#!define DB_URL "mysql://pcscf:heslo@127.0.0.1/pcscf"##!define DB_URL2 "con2=>mysql://pcscf:heslo@127.0.0.1/pcscf"##!define DB_URL "con1=>mysql://pcscf:heslo@192.168.5.1/pcscf"##!define DB_URL2 "con2=>mysql://pcscf:heslo@10.1.27.217/pcscf" #! Optional: Server-URL for Websocket-Requests##!define WEBSOCKET_WEBSERVER "phone.imscore.org" # NUEVO Workshop!#!subst "/ICID_VALUE_PREFIX/P-CSCFabcd/"#!subst "/ICID_GEN_ADDR/127.0.0.1/" # IP-Adress(es) of the RTP-Proxy##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222 udp:10.1.27.217:22222"#!define RTPPROXY_ADDRESS "udp:192.168.0.108:22222"## Several features can be enabled using '#!define WITH_FEATURE' directives:## *** To run in debug mode:# - define WITH_DEBUG## *** To enable nat traversal execute:# - define WITH_NAT# - define the connection to the RTP-Proxy: RTPPROXY_ADDRESS## *** To force alls calls through the RTP-Proxy# - this will automagically enable NAT-Traversal# - define FORCE_RTPRELAY## *** To enable IPv4/IPv6 Translation (RTPProxy)# - this will automagically enable NAT-Traversal# - define WITH_RTPIPV4## *** To enable TCP support execute:# - define WITH_TCP## *** To enable TLS support execute:# - adjust CFGDIR/tls.cfg as needed# - define WITH_TLS# - this will automagically enable TCP## *** To enable XMLRPC support execute:# - define WITH_XMLRPC# - this will automagically enable TCP## *** To enable anti-flood detection execute:# - adjust pike and htable=>ipban settings as needed (default is# block if more than 16 requests in 2 seconds and ban for 300 seconds)# - define WITH_ANTIFLOOD## *** To enable the Rx-Interface:# - Configure Rx-Diameter-Interface in pcscf.xml# - define WITH_RX## *** To enable a Homer SIP-Capter-Node:# - define CAPTURE_NODE with a proper address## *** To enable support for the SEMS-SBC# - define WITH_SBC# - configure dispatcher-list with a set of SBC's## *** To enable support for Websocket# - define WITH_WEBSOCKET# - this will automagically enable TCP## Enabled Features for this host:#!define WITH_DEBUG#!define WITH_NAT#!define WITH_NATPING#!define FORCE_RTPRELAY##!define WITH_TLS#!define WITH_XMLRPC#!define WITH_ANTIFLOOD##!define WITH_RX##!define WITH_RX_REG##!define WITH_RX_CALL##!define WITH_TCP##!define WITH_SBC##!define WITH_RTPIPV4#!define WITH_SBC_CALL#!define WITH_REGINFO#!define WITH_WEBSOCKET#!define WITH_IMS_HDR_CACHE Thanks a lot for your attention

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Kamailio dead after INVITE from SIPDROID
by Kowalski Piotr 3 - Hurt 26 Aug '15

26 Aug '15

Hello Community. We have openIMS nodes in our lab. We have Kamailio 4.0 based solution. I registered SIPDROID client (iptel) on Android and JITSI on WINDOWS. When I call from SIPDROID to JITSI following happens: SIP DROID - iptel - client - registration ok, but INVITE attempt towards JITSI (and other clients) does what's below: HSS (every 30 seconds): [root@hss1 kowalp00]# 2015-08-25 12:16:55,468 WARN de.fhg.fokus.diameter.DiameterPeer.transport.Communicator - run Read 3021 before Exception. 2015-08-25 12:16:56,980 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer scscf1.hss.ims.test.pl:3868 not responding to connection attempt 2015-08-25 12:17:26,985 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer scscf1.hss.ims.test.pl:3868 not responding to connection attempt SCSCF: Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8879]: INFO: <core> [main.c:854]: INFO: signal 15 received Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8865]: INFO: <core> [main.c:854]: INFO: signal 15 received Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: INFO: snmpstats [snmpstats.c:410]: The SNMPStats module got the kill signal Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: INFO: snmpstats [snmpstats.c:414]: Shutting down the AgentX Sub-Agent! Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Aug 25 12:25:18 scscf1 /usr/sbin/kamailio[8803]: : <core> [main.c:697]: BUG: shutdown timeout triggered, dying... Kamailio becomes dead and needs restart. Clients become unregistered. A call from JITSI to SIPDROID it's fine. Any ideas? Piotr Kowalski Starszy Architekt Techniczny Tel. +48 22 699 50 93 Kom. +48 519 123 289 Orange Orange Polska Hurt Badania i Rozwój Sieci i Platform Usługowych Wydział Usług Konwergentnych Obrzeżna 7 /p. 309, 02-691 Warszawa Fax. +48 22 857 99 86 <http://www.hurt-orange.pl/> www.hurt-orange.pl - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Czy musisz drukować tę wiadomość? Pomyśl o środowisku. Korzystaj z e-faktury. Treść tej wiadomości jest własnością Orange Polska i zawiera informacje stanowiące tajemnicę przedsiębiorstwa Orange Polska. Jeżeli nie jesteście Państwo jej adresatem, bądź otrzymaliście ją przez pomyłkę, prosimy o powiadomienie o tym nadawcy oraz trwałe jej usunięcie. Orange Polska Spółka Akcyjna z siedzibą i adresem w Warszawie (02-326) przy Al. Jerozolimskich 160, wpisana do Rejestru Przedsiębiorców prowadzonego przez Sąd Rejonowy dla m.st. Warszawy XII Wydział Gospodarczy Krajowego Rejestru Sądowego pod numerem 0000010681; REGON 012100784, NIP 526-02-50-995; z pokrytym w całości kapitałem zakładowym wynoszącym 3.937.072.437 złotych.

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Rewrite Invite w/o sdp
by Serge Yuriev 25 Aug '15

25 Aug '15

Hello, Our PBX (Cisco CallManager) uses "late offer" sending invites w/o sdp. After receiving provisional reply 18x it answers prack with sdp I'm trying to realize following scenario: 1 Pbx (invite no sdp) 2 kamailio hold and saves invite 3 send 183 to Pbx 4 get prack with sdp 5 construct new invite from saved one and sdp from prack 6 send it to calee Is this possible? How to save invite for future use? -- Serge via mobile

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dispatcher gateway probing in 4.3
by Joseph Dickson 24 Aug '15

24 Aug '15

Greetings.. I'm fairly new to Kamailio, and am having fun so far... I'm using the latest 4.3 rpm version on Centos 7. I'm using Kamailio to front end a pair of FreeSwitch SBC boxes in an active/active config.. My plan is to use dispatcher to load balance calls between them. This is working fine.. the problem that I'm bumping into is that the gateway probing options in dispatcher seem not to fit my use case.. In addition to the two SBC gateways, I also have a few external gateways that I want to load balance between also using dispatcher. The problem is that I can't send options packets to these gateways -- they don't respond to them. This means I can't just configure dispatcher to probe all gateways (probing_method=1) because it will mark these gateways as inactive.. It looks like every other configuration of probing_method only probes a gateway until the state is determined, and then it stops probing. probing_method=2 does not appear to be used at all in the code.. Am I doing this wrong? Or is this just an area of dispatcher that needs some improvement? Looking at the code, I can probably add the functionality that I need, but I don't want to go that route if I'm just missing how this is usually done.. Thanks, -- *Joseph Dickson* E: jdickson(a)evolvetsi.com

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Request timout - Change value
by Loic Chabert 24 Aug '15

24 Aug '15

Hello everybody, I'm using kamailio as a dispatcher in front of asterisk boxes and i use a failure route if asterisk box does not respond or send 500error. failure_route[RTF_DISPATCH]{ if(t_is_canceled()){ exit(); } if(t_check_status("500") || (t_branch_timeout() && !t_branch_replied())){ xlog("L_WARN","[$fU@$si:$sp]{$rm} Asterisk Box $du is down\n"); ds_mark_dst("ip"); if(ds_next_dst()){ xlog("L_WARN","[$fU@$si:$sp]{$rm} Sending to Asterisk Box - $du\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } As you can see, i check if "t_branch_timeout() && !t_branch_replied()", but where has been stored timeout value ? How can i change it ? For exemple, i need to set 1s, and then, request expired. Thanks, Regards.

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Dialog module - db_fetch_rows
by jay binks 24 Aug '15

24 Aug '15

Hey All, Im just after someone else thoughts on if this is a bug. After reading : http://kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#idp148408 Im lead to understand that setting *modparam("dialog", "db_fetch_rows", 0)* should instruct the dialog module to write dialogs to the DB, but not bother loading them on startup. This sounds like what I'm after. However ever time I set the value to 0, dialogs are still loaded from the DB. a quick dig in the source takes me to the function select_entire_dialog_table in dlg_db_handler.c this chunk of code does not appear to properly implement " if fetch_num_rows == 0 then do nothing " Im after a couple of things from you guys. a) validation that my understanding of what SHOULD happen is correct b) possibly validating that I'm not crazy, and this is actually a bug :) -- Sincerely Jay

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Ims Kamailio issue - "403 Forbidden - You must register first with a S-CSCF"
by Victor Huertas Garcia 23 Aug '15

23 Aug '15

Hello everybody, I have installed an IMS core using Kamailio and FHoSS. In this environment, I am able to successfully register Bob and Alice with the same S-CSCF. However, when I try to make a voice call between Bob and Alice it is not established. The code received is "403 Forbidden - You must register first with a S-CSCF". This code is generated by the P-CSCF because the Contact Header that is checked against the database ("location" table) it is not found. When I look up manually the "location" MySQL table, there are two things that surprise me: - The Contact header (AOR column content) that is included in the INVITE Requet URI when entering in the P-CSCF is present in the table. So why the pcscf_is_registered ("location") returns false? - When Bob registers two Contacts are stored, the first one is the one included in the REGISTER message generated from Bob IMS Client. However, the second Contact is the one present in the 200 OK message (coming from the S-CSCF) which is different from the one present in the REGISTER (the only thing different is the UDP port). Why the S-CSCF (with the default kamailio.cfg) doesn't the Contact header and changes the UDP port? Is this the standard behaviour? I don't think so... Thanks for your attention Regards,

2 3

A question to you, SIP providers: what is your policy on PBXes calling themselves
by Antonio Gómez Soto 23 Aug '15

23 Aug '15

Hello, This is question on PBX behavior, what is the right thing to do, and how do PBX's generally behave. If a user on a phone, dials a number, which happens to be configured on the same phone system (for example another tenant), there are two options: 1. The PBX notices this, and directly connects the phone to the DID on that system (breaking separation of tenants) 2. The PBX sends the call out on the SIP trunk, and the provider-routing sends the call back as an incoming call. What are the pros and cons of each option from the SIP provider point of view? How do PBX's generally behave? Thanks, Antonio PS: I reposted, because the original question was apparently not phrased in a clear way.

3 2

IPSec supporting open source SIP Server
by Priyaranjan Nayak 23 Aug '15

23 Aug '15

Hi All, I need one IPSec supported opensource SIP server for testing my sip client endpoint.Could you please tell me which SIP server supporting IPSec ? Thanks in advance. Thanks Priyaranjan

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Re: [SR-Users] Presence and NAT
by Pramod Venugopal 21 Aug '15

21 Aug '15

I realize this last email has no useful information, so let me elaborate. I have a Kamailio server on AWS with an internal IP of 10.20.30.40 and it has an external elastic IP of 55.66.77.88. When a SIP client subscribes it sends the initial subscription to user@realm (where realm resolves to 55.66.77.88) , then it gets a notify with a new address <sip:10.20.30.40:5060;transport=udp> . The client attempts to refresh the subscription, but we are now sending the subscribes to this new contact which will time out. 300 seconds later, we subscribe again to user@realm and contacts show up. Other stuff like rtp, inbound/outbound calling all work properly. Pramod Venugopal pramod(a)dvnull.org From: sr-users on behalf of Pramod Venugopal Reply-To: "Kamailio (SER) - Users Mailing List" Date: Thursday, August 20, 2015 at 10:38 PM To: "Kamailio (SER) - Users Mailing List" Subject: [SR-Users] Presence and NAT Hello everyone, I have a Kamailio 4.2 system on Amazon AWS. When subscribing to presence the contact is sent out with the internal IP rather than the external public IP. The subscribes are being sent to it with the external hostname ( which resolves to the external IP ). Where would I be able to change the IP sent in the Contact header ? Pramod Venugopal pramod(a)perseus-tech.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Problems with RTPPROXY module
by Michael Nielsen 21 Aug '15

21 Aug '15

Hello everyone, I'm new to Kamailio and are trying this tutorial: http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I've installed everything and are trying to launch Kamailio, but it fails with the following error: ----- systemctl status kamailio.service ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2015-08-21 06:26:26 CDT; 8s ago Process: 1474 ExecStart=/etc/init.d/kamailio start (code=exited, status=1/FAILURE) Aug 21 06:26:26 sip kamailio[1474]: error in config file /etc/kamailio/kamailio.cfg, line 234, column 12-25: failed to load module 0(1483) ERROR: <core> [sr_module.c:563]: load_module(): could not find module <rtpproxy> in </usr/lib64...3432]: yyerror_at(): Aug 21 06:26:26 sip kamailio[1474]: r in config file /etc/kamailio/kamailio.cfg, line 270, column 35: Can't set module parameter 0(1483) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <registrar> found 0(148...ailio/kamailio.cfg, Aug 21 06:26:26 sip kamailio[1474]: : <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(1483) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 295, column ...mod_param_regex(): N Aug 21 06:26:26 sip kamailio[1474]: 1483) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 321, column 36: Can't set module parameter 0(1483) ERROR: <core> [modparam.c:150]: set_mod_para...]: yyerror_at(): par Aug 21 06:26:26 sip kamailio[1474]: file /etc/kamailio/kamailio.cfg, line 378, column 50: Can't set module parameter 0(1483) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command is_method (params 1) 0(1483) : <co...mailio.cfg, line 417 Aug 21 06:26:26 sip kamailio[1474]: fg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 436, column 24: unknown command, missing loadmodule? 0(1483) ERROR: <core> [pvapi.c:810]: pv_parse_spec2(): erro...): wrong char [U/85] Aug 21 06:26:26 sip kamailio[1474]: failed! Aug 21 06:26:26 sip systemd[1]: kamailio.service: control process exited, code=exited status=1 Aug 21 06:26:26 sip systemd[1]: Failed to start LSB: Start the Kamailio SIP proxy server. Aug 21 06:26:26 sip systemd[1]: Unit kamailio.service entered failed state. Hint: Some lines were ellipsized, use -l to show in full. ----- It seems that if it cannot find the module RTPPROXY. I have installed it via apt-get install rtpproxy and it seems to be running. What to do?

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Debian nightly builds
by Jon Bonilla (Manwe) 21 Aug '15

21 Aug '15

Hi all Trying to install latest nightly build of several branches I find that they are from June. Is the repo outdated? http://deb.kamailio.org/kamailiodev-nightly/pool/main/k/kamailio/ cheers, Jon

2 1

Presence and NAT
by Pramod Venugopal 21 Aug '15

21 Aug '15

Hello everyone, I have a Kamailio 4.2 system on Amazon AWS. When subscribing to presence the contact is sent out with the internal IP rather than the external public IP. The subscribes are being sent to it with the external hostname ( which resolves to the external IP ). Where would I be able to change the IP sent in the Contact header ? Pramod Venugopal pramod(a)perseus-tech.com

1 0

Back to the dead communications: No outgoing audio
by Manuel Camargo Lominchar 20 Aug '15

20 Aug '15

After 6 months of resolving this situation: http://lists.sip-router.org/pipermail/sr-users/2015-February/086980.html Without touching any configs, my server is back to a failure situation I'm not sure if I'm facing the exact same thing, but for some strange reason, without introducing any modification in the servers, one leg is not hearing voice (outgoing traffic), while incoming rtp traffic goes fine, so I have to assume another time that some Update has destroyed it I've mostly done the same checks as before. I started to assume that rtp-engine could be failing by the same cause, but I have updated to the latest version and still the same issue This has happened from 2-3 day to now. Any ideas on how to start trying to find the cause? Versions: Kamailio 4.1.4 RTP Engine: 4.1.0.0 Kind regards, *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com

1 1

Mitel 5000 and 3000
by Fred Posner 20 Aug '15

20 Aug '15

I rarely integrate these with Kamailio and am having some "resistance" in assisting. I was wondering if anyone would be willing to share any configuration recommendations for Mitel when using Kamailio as a SIP trunk (outbound/inbound). Sincerely, Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

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Re: [SR-Users] Mitel 5000 and 3000
by rahul.ultimate 20 Aug '15

20 Aug '15

Hi Fred I too am new to Mitel products but the attached pic might be of some use to you. Thanks RM Sent from Samsung Mobile -------- Original message -------- From: Fred Posner <fred(a)palner.com> Date:18/08/2015 7:08 PM (GMT+05:30) To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Mitel 5000 and 3000 On 08/18/2015 09:23 AM, Alex Balashov wrote: > Are there any specific issues you're encountering? > Mostly hold issues where they try to speak directly with the media server vs the sip server; as well as general audio issues in/out. Most of these seem to be acl related with a set-up issue on the Mitel. I don't work with Mitel enough to discuss in "Mitel speak" what to change. Apparently RTP ports is a different language to the people I am working with. =) --fred _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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new SIP account can not be created ... DB Error : connect failed
by Rainer Böing 20 Aug '15

20 Aug '15

Hello, want to build the following account about Ipel.Org . My old following account no longer works. Iptel.org 19.11.2013 E-Mail rainer.boeing(a)t-online.de Username: boe.rai(a)iptel.org <mailto:boe.rai@iptel.org> Password: xd7ghg Sip address: <sip:boe.rai@iptel.org> sip:boe.rai@iptel.org I can also create no new no account . Please Help !!! clamp THANKS Greetings from Germany Rainer Böing Disclaimer: 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 vernichten 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 destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

2 1

Change ports for outgoing RTP packets in RTPproxy
by Jean-Marie Baran 19 Aug '15

19 Aug '15

Hi, Is there a way to change the ports were RTPproxy sends outgoing packets ? By rewriting the SDP before it gets to RTPproxy, or anything else ? 1 - Kamailio receives the SDP 2 - It somehow transmits the contact information to RTPproxy (ip + ports) 3 - The client start sending packets to RTPproxy which in turn relays them to the contact How to change the ports in step 2 ? Sincerely, -- *Jean-Marie Baran*

2 2

how to use seturi with varibales in kamailio
by SureshTummala 19 Aug '15

19 Aug '15

Hi, I have the following questions 1. How to remove ruri which is already added to the destination set2. How to rewritehost in ruri with variables3. How to change the To header in INVITE which are generated by t_relay based on destination set Regards,Suresh Tummala

2 1

Kamailio with NATted RTPproxy
by Jean-Marie Baran 18 Aug '15

18 Aug '15

Hi, I am trying to have RTPproxy works with Kamailio behind a NAT. As it seems RTPproxy cannot handle NAT natively, I installed RTPproxy 1.2.1 by miconda: https://github.com/miconda/rtpproxy Now I am able to make calls, but neither the video nor the audio are present. Wireshark showed that the RTP packets were sent to wrong IPs (private IPs). And this was because of the SDP which contained the wrong IP (fields o= and c=). Then I tried to play with rtpproxy_manage(), and Kamailio replaced the IP in o= and c= by its advertising IP (i.e. its public IP), so the packets from the client behind the NAT were sent to the public address of Kamailio (and then dropped). My question here is what should be the addresses in the SDP when redirected from Kamailio to the client ? Here is the topology when receiving an ACK with SDP: Client <--(2)-- Kamailio <--(1)-- Router (NAT) <--(1)-- Public Internet <--(1)-- Third party router (21.24.12.24) (1): SDP o= 21.24.12.24, c= 21.24.12.24 (2): SDP o= ??, c= ?? (Should it be Kamailio public IP (actually Router IP), or Kamailio private IP, or third party SIP server public IP ?) Cheer, -- *Jean-Marie Baran*

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rtpengine rewrite sdp for second time
by Virmantas Variakojis 18 Aug '15

18 Aug '15

Hi Carsten, Thanks for your reply! How can I call msg_apply_changes() in right way? If I call msg_apply_changes() for example in route[RELAY] i get error: ERROR: textopsx [textopsx.c:157]: msg_apply_changes_f(): invalid usage - not in request route if I do everything as you suggested in onreply_route (after I get for example SIP 500 message) it doesn't work and my SDP remains unchanged. Thanks! > > *From: *Carsten Bock <carsten(a)ng-voice.com> > *Subject: **Re: [SR-Users] rtpengine rewrite sdp for second time* > *Date: *17 Aug 2015 09:49:12 EEST > *To: *"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org > > > *Reply-To: *"Kamailio $SER$ - Users Mailing List" < > sr-users(a)lists.sip-router.org> > > Hi Virmantas, > > you could try to one or a combination of the following: > - explicitly delete the first session by calling rtpengine_manage() in > the failure route > - call msg_apply_changes() from Textops before calling > rtpengine_manage() the second time (so the already made changes are > applied and the message gets reparsed) > > Thanks, > Carsten > > 2015-08-16 20:13 GMT+02:00 Virmantas Variakojis > <virmantas.variakojis(a)gmail.com>: > > Hi, > > I have a situation like this: > I have kamailio with 3 NIC's. I'm sending INVITE to first provider over > NIC1. Before sending INVITE I use rtpengine_manage to rewrite sdp body with > IP that is on NIC1: > rtpengine_manage ("replace-origin replace-session-connection > direction=intvlan direction=prov1 ICE=remove"); > > If first provider fails i want to send INVITE for the same call to second > provider. But if i use like this: > rtpengine_manage ("replace-origin replace-session-connection > direction=intvlan direction=prov2 ICE=remove"); > or (because ip of prov1 is allready set): > rtpengine_manage ("replace-origin replace-session-connection > direction=prov1 > direction=prov2 ICE=remove"); > rtpengine ignores my changes as said in documentation: "The direction must > only be specified in for initial SDP offer; answers or subsequent offers > can > omit this option." > > and if I try to set ip like this: > rtpengine_manage ("replace-origin replace-session-connection > media-address=X.X.X.X ICE=remove"); > SDP body is doubled: it leaves old values and adds the new one. > > What i'm doing wrong or how can I rewrite SDP for second time? Thanks in > advance for answer! > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Office +49 40 5247593-0 > Fax +49 40 5247593-99 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

Routing re-INVITEs after phone port change
by David Cunningham 18 Aug '15

18 Aug '15

Hello, We have a scenario like this: SUA -> Kamailio with registrar module -> Asterisk A call from the SUA is set up with SIP timers, and after 15 minutes Asterisk sends a re-INVITE to Kamailio to forward on to the SUA. That re-INVITE has a RURI with the address and port of the SUA at the time the call started. Now if the SUA re-registers after the call starts and before the re-INVITE, and is on a new address or port number, then the re-INVITE never gets to the phone. Obviously Kamailio should send the re-INVITE to the new address/port, but is not. The re-INVITE is routed using the lookup() function. Can anyone suggest why this is happening? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

2 5

Kamailio/Asterisk Integration - MySQL driver error
by Ben Fitzgerald 17 Aug '15

17 Aug '15

Has anyone seen this error before? " ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Host '192.168.xxx.xxx is blocked because of many connection errors; unblock with 'mysqladmin flush-hosts'" I fixed it using flush hosts but I'm curious why this popped up? Is there a way to limit connections for Kamailio and why am I reaching that limit with *only 2* SIP endpoints? This is not a heavily utilized database by any means. Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

2 9

Siremis docker
by Sylvain Boily 17 Aug '15

17 Aug '15

Hello, I did a docker image for siremis, if some people want to play easily like me with kamailio and siremis ;) Info here : https://github.com/sboily/dockerfile/tree/master/siremis or : https://hub.docker.com/r/quintana/siremis/ Have a good day. Sylvain

3 2

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