- sr-users - kamailio.org

kamailio db
by Slava Bendersky 18 Apr '14

18 Apr '14

Hello Everyone, Is possible run kamailio without database ? I got ldap working and extension is registered, but can't place calls between two extensions. Getting 404 all the time. Slava.

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invite always contains "s"
by Vasco Mendes 18 Apr '14

18 Apr '14

Hi, I created an incoming trunk between iptel and my asterisk. Whatever the account is on iptel I always have in the from of received INVITE < sip :s@mypublicip :5060 >. < s > is not a recognized DID on my asterisk. Why do I receive this < s > in the from ?? Many thanks for your help, Vasco --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

2 1

Editings in kamailio.cfg for routing decision
by aawaise 18 Apr '14

18 Apr '14

Hi All, When a user sends an "invite" packet to make a call. I want to make a decision about call routing. Whether to route call to other server or route it to local user. I want to base this decision on "domain" of called user. This "domain" is to be picked from "Location List" in database. Both servers are using the same database. What changes are to be made in kamailio.cfg ?? Actually in my environment my user can connect to any of the servers. I want to know to which server it is connected and then make the call routing decision. Any help will be highly appreciated. Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Editings-in-kamailio-cfg-for-routin… Sent from the Users mailing list archive at Nabble.com.

1 0

Custom fr timer for dialog's OPTIONS keep alives
by Vassilis Radis 17 Apr '14

17 Apr '14

Hello, I would like to try customizing the fr_timer for the OPTIONS keep-alive msgs transactions that the dialog module sends. I tried to call t_set_fr() function from inside the dialog modules's code (in dlg_req_within.c , in dlg_send_ka() function after the call to d_tmb. t_request_within(&uac_r); But it seems that this tm function is not available from inside other module's code. Is the above is the correct place to try and do it, and how can I programmatically ,inside a module's code, adjust fr_timer? On a side note, i followed the call flow from dlg_send_ka() down to the t_uac_prepare() function of tm module, where the fr_timer is set. In there, there is this comment+code: /* init timers hack, new_cell->fr_timer and new_cell->fr_inv_timer * must be set, or else the fr will happen immediately * we can't call init_new_t() because we don't have a sip msg * => we'll ignore t_set_fr() or avp timer value and will use directly the * module params fr_inv_timer and fr_timer -- andrei */ new_cell->fr_timeout=cfg_get(tm, tm_cfg, fr_timeout); new_cell->fr_inv_timeout=cfg_get(tm, tm_cfg, fr_inv_timeout); new_cell->end_of_life=get_ticks_raw()+lifetime; Does this mean that it cant be done in a straighforward way? Any advice on how I can achieve the customized fr_timers for those messages? Thanks.

2 2

Basic question regarding Routing
by aawaise 17 Apr '14

17 Apr '14

Hello, I want to ask some basic questions. Anyone please guide me to links with answers. I am sure such basics must be explained online. 1. What is exactly achieved by "route(RELAY)" ?? Routing calls from local users to other domains ?? 2. If user 1 is caller and user 2 is being called. Which pseudo variable will be used to catch value of called user i.e user 2 and his domain ?? I want to pick domain name from location table in database and not from the authorization module. Thanks, Cheers, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Basic-question-regarding-Routing-tp… Sent from the Users mailing list archive at Nabble.com.

2 1

Routing decision based on domain entry in Location Table
by aawaise 17 Apr '14

17 Apr '14

Hello, I have deployed two kamailio servers, both sharing the same database. I want my kamailio servers to make a decision on the routing of incoming call by seeing the domain of "called user" from location table of online users. If the "called user" is connected to same server, the call should be routed to user directly. In other case if the user is connected to other server, the call should be forwarded to other user for further processing. Will someone please guide me how can this be achieved ?? P.S : I am using kamailio-3.1.1. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Routing-decision-based-on-domain-en… Sent from the Users mailing list archive at Nabble.com.

2 4

rls multipart body does not contain watcherinfo+xml
by Mihai Marin 16 Apr '14

16 Apr '14

Hello Sirs, I'm trying to accomplish a rls subscribe but I have some miss-understandings regarding the multipart publish response that should come. I'm using the integrated xcap from the kamailio with all the xml files added. My problem is that event if I request a multipart containing application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml I only get the first two. The request contains the following headers: Event: presence Supported: eventlist Accept: multipart/related, application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml The response only sends the first two: --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:bob@93.187.138.214.681064155> Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list uri="sip:bob@93.187.138.214" xmlns="urn:ietf:params:xml:ns:rlmi" version="4" fullState="false"><resource uri="sip:alice@93.187.138.214"><instance id="Scf8UhwQ" state="active" cid="1397508890.sip:alice@93.187.138.214.1552580371"/></resource></list> --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:alice@93.187.138.214.1552580371> Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8" standalone="no"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:alice@93.187.138.214"> <dm:person id="p6850"> <rpid:activities/> <rpid:status-icon>http://93.187.138.214:5060/xcap-root/oma_status-icon/users/sip:alice@93.187…</rpid:status-icon> </dm:person> <tuple id="t7673"> <status> <basic>open</basic> </status> <contact>sip:alice@93.187.138.214</contact> <note>Online</note> </tuple> </presence> --4kSPpRGyJQct4DIhyGnyobqO-- What do I missing? I need the body part with the watchers info in oder to see the state of the watcher (active, pending) - to accomplish the authorization for users. Thank you in advance for you help. Best regards, Mihai M

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Register request forwarding from kamailio to asterisk
by betul.cerit＠netgsm.com.tr 16 Apr '14

16 Apr '14

Hi to all We integrated kamailio and asterisk as explained here http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb When kamailio receives a register request it authenticates and forwards the request to asterisk.Why do we need to do this?What happens if we dont pass registration requests to asterisk? Thanks in Advance Betül

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Security
by Keith 16 Apr '14

16 Apr '14

Hi All, I am looking to apply some security on invites so I only accept from IP addresses in the address table or authenticated users. I have the address table bit working but I am struggling with the authenticated users bit. Wondering if anyone has done it? I was thinking of using the location table in some way as those users have authenticated? Any ideas would be appreciated. Keith

2 3

uac_auth problem
by neumann 16 Apr '14

16 Apr '14

Hi all sorry for my english! Iam trying to authenticate as uac on 3rd party server(Asterisk). Kamailio get 401, send next invite with DIGEST but Asterisk send 401 again((( Asterisk don’t write in log what auth is failed. 115.600045 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP/SDP 1093 Request: INVITE sip:321321231@xxx.xxx.xxx.xxx, with session description 115.601818 yyy.yyy.yyy.yyy -> xxx.xxx.xxx.xxx SIP 674 Status: 401 Unauthorized 115.602470 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP 389 Request: ACK sip:321321231@xxx.xxx.xxx.xxx 115.604310 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP/SDP 1267 Request: INVITE sip:321321231@xxx.xxx.xxx.xxx, with session description 115.605537 yyy.yyy.yyy.yyy -> xxx.xxx.xxx.xxx SIP 674 Status: 401 Unauthorized 115.606431 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP 389 Request: ACK sip:321321231@xxx.xxx.xxx.xxx My aster peer: [kamailio] type=peer defaultuser=kamailio secret=1234 host=dynamic disallow=all allow=alaw nat=no qualify=yes My kamailio config: modparam("uac","auth_username_avp","$avp(s:uac_user)") modparam("uac","auth_password_avp","$avp(s:uac_pass)") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") route[DISPATCH_OUT] { if(!ds_select_dst( 4, 8)){ send_reply("404", "No destination"); exit; } t_set_fr(0,2000); t_on_failure("RTF_DISPATCH_OUT"); t_relay(); exit; } failure_route[RTF_DISPATCH_OUT] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(s:uac_user) = "kamailio"; $avp(s:uac_pass) = "1234"; $avp(s:uac_realm) = "asterisk"; if (isflagset(7)) { t_reply("513","Authentication failed"); xlog("L_INFO", "[%ci]: Remote authentication failed\n"); break; } if (uac_auth()) { xlog("L_INFO", "401/407 message recived"); # mark that auth was performed setflag(7); # trigger again the failure route t_on_failure("RTF_DISPATCH_OUT"); # repeat the request with auth response this time append_branch(); #t_relay_to_udp("94.25.100.198","5060"); t_relay(); break; } else { xlog("L_INFO", "uac_auth filed!!!!!!!!!!!"); } t_reply("514","Unknow authentication peer"); xlog("L_INFO", "[%ci]: Unknown authentication peer.\n"); break; } # next DST - only for 500 or local timeout if (t_check_status("500") or t_check_status("480") or (t_branch_timeout() and !t_branch_replied())){ ds_mark_dst("p"); if(ds_next_dst()){ t_set_fr(0,2000); t_on_failure("RTF_DISPATCH_OUT"); route(RELAY); exit; } } } ———————————— Timofeev Dmitry VoIP Engineer Linux, Asterisk, Freeswitch, Cisco solutions Skype: itsroot icq: 227227933

2 1

client provisioning.
by Slava Bendersky 15 Apr '14

15 Apr '14

Hello Everyone, Is possible do phone provisioning with kamailio ? Slava.

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PERMISSIONS module issue
by Samuel Ware 15 Apr '14

15 Apr '14

I having issue updating my allow list for the PERMISSIONS module. I added an address to the ADDRESS table. I have tried to do a service restart, kamctl address reload, and kamcmd permissions.addressReload. The kamctl address show displays the new address; however kamcmd permissions.addressDump does not neither does kamcmd permissions.subnetDump. The messages from this new address return a false to the !allow_source_address("1”) command in my routing logic. I am wondering if this is a bug or I am doing something wrong. I am on the most recent GIT version to the best of my knowledge. Sam

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SDP body rewritten issue at RTPproxy
by Wingsravi R 15 Apr '14

15 Apr '14

Dear All, I am working with Kamailio (4.1.2) and RTPproxy (1.2.1) servers and Jitsi as SIP clients. I have a problem in Audio/Video calling with Kamailio+RTPproxy server combination that Video is not streaming to other side through RTPproxy(also no audio both sides), I am suspecting something is rewritten in SDP body at RTPproxy, concerning to audio/video parameters (I am unable to rectify what is causing problem there) . Please find the Tcpdump based SIP capture of video call with this Kamailio+RTPproxy set-up. And with only kamailio server running (without RTPproxy instance running), audio/video call is fine on both sides. (find SIP capture for the same call). Also find my kamailio.cfg file for any reference. Please anybody help me in rectifying problems at RTPproxy server. Awaiting somebody's comments on this issue. Regards, Ravi

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Re: [SR-Users] SDP c= IP doesn't get rewritten
by Ciprus, Daniel 14 Apr '14

14 Apr '14

Yes I do Sent via the Samsung GALAXY S®4, an AT&T 4G LTE smartphone -------- Original message -------- From: Daniel-Constantin Mierla Date:04/11/2014 6:13 AM (GMT-05:00) To: "Ciprus, Daniel" ,"Kamailio (SER) - Users Mailing List" Subject: Re: [SR-Users] SDP c= IP doesn't get rewritten Hello, are you using a rtp relay (e.g., rtpproxy)? Cheers, Daniel On 11/04/14 02:46, Daniel Ciprus wrote: All, I am facing problem with rewriting SDP part of the INVITE, in particular following is in my configuration: #!subst "/RTP_INT_ADDR/10.96.173.15/" #!subst "/RTP_EXT_ADDR/168.7.24.17/" #!subst "/BOT_RTP_ADDR/10.96.173.52/" subst_body('/RTP_INT_ADDR/RTP_EXT_ADDR/g'); subst_body('/BOT_RTP_ADDR/RTP_EXT_ADDR/g'); and this is what I ended up with: c=IN IP4 10.96.173.15168.7.24.17 Apparently 2nd rule doesn't work rewrite, it's doing append instead. What I want to achieve is : if SDP part contains BOT_RTP_ADDR -> rewrite it. In case SDP part contains RTP_INT_ADDR-> rewrite it, so 2x if. I have tried to use sdpops module, it's coring in my case and I have a reason to not upgrade, I just can't. Any other ideas how to handle my problem ? Thanks Dan Server:: kamailio (4.2.0-dev0 (x86_64/linux)) Build:: mi_core.c compiled on 17:42:57 Dec 18 2013 with gcc 4.4.7 Flags:: STATS: Off, USE_TCP, USE_TLS, 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 GIT:: unknown -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

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LUA - UAC package implementation
by Daniel Ciprus 11 Apr '14

11 Apr '14

Hi, How much of UAC package is implemented in LUA embedded module ? According to wiki seems like only "sr.uac.replace_from" is implemented. My goal is to "emulate" 3rd party registration by using UAC and LUA meaning that I will have to recreate register, tweak existing attributes and send it over to application server to process. thanks Dan -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

3 4

about tls client certificates
by Juha Heinanen 11 Apr '14

11 Apr '14

while doing some tls tests, i noticed that if tls.cfg has a section like this [server:default] verify_certificate = no require_certificate = no tls_method = SSLv23 private_key = /etc/sip-proxy/certs/sip-proxy/key.pem certificate = /etc/sip-proxy/certs/sip-proxy/cert.pem ca_list = /etc/ssl/certs/cacert.org.pem then client does not give its certificate to kamailio server during tls connection setup even if it had one. if i specify: require_certificate = no then client sends its certificate to kamailio server, but if another client does not have a client certificate, then it cannot connect at all. one way to solve this would be making kamailio listen on two tls ports, one for clients that are required to present a a certificate and another port for clients that do not have a certificate. unfortunately, it is not possible to add a mask to ip address in tls.cfg section like this: [server:0.0.0.0/0:5062] does anyone have a solution to this problem (other that running two kamailio instances)? -- juha

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SDP c= IP doesn't get rewritten
by Daniel Ciprus 11 Apr '14

11 Apr '14

All, I am facing problem with rewriting SDP part of the INVITE, in particular following is in my configuration: #!subst "/RTP_INT_ADDR/10.96.173.15/" #!subst "/RTP_EXT_ADDR/168.7.24.17/" #!subst "/BOT_RTP_ADDR/10.96.173.52/" subst_body('/RTP_INT_ADDR/RTP_EXT_ADDR/g'); subst_body('/BOT_RTP_ADDR/RTP_EXT_ADDR/g'); and this is what I ended up with: c=IN IP4 10.96.173.15168.7.24.17 Apparently 2nd rule doesn't work rewrite, it's doing append instead. What I want to achieve is : if SDP part contains BOT_RTP_ADDR -> rewrite it. In case SDP part contains RTP_INT_ADDR-> rewrite it, so 2x if. I have tried to use sdpops module, it's coring in my case and I have a reason to not upgrade, I just can't. Any other ideas how to handle my problem ? Thanks Dan Server:: kamailio (4.2.0-dev0 (x86_64/linux)) Build:: mi_core.c compiled on 17:42:57 Dec 18 2013 with gcc 4.4.7 Flags:: STATS: Off, USE_TCP, USE_TLS, 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 GIT:: unknown -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 1

Disjoint REGISTER and INVITE username/uri sets
by James Cloos 10 Apr '14

10 Apr '14

Given a proxy where the only registrations will be next-hop pbxes, where those names will not be INVITEable, what is the ideal way to tell kama the list of accepted usernames and authentication info for REGISTER? It is important to ensure that any INVITE to any of the REGISTERable usernames 404s. Thanks, -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 1024D/ED7DAEA6

3 3

carrierroute prefix regex?
by Igor Potjevlesch 10 Apr '14

10 Apr '14

Hello, I'm wondering if we can set into < scan_prefix > an entrier with regex like : 3[0-9]{3}? If not, what would be the best module to manage in a DB the prefix that can be composed by SIP clients before call cr_route? Something equivalent to a dialplan I guess. Thanks for your feedback. Regards, Igor.

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Re: [SR-Users] How Do you Stack up Against Cisco's Call Manager
by Marjorie Lawrence 10 Apr '14

10 Apr '14

Hi, Not sure if this is a fair question. But how do you compare your call processing engine against Cisco's Call Manager? Appreciate your time. Marjorie Lawrence Global Wireless Networks, Inc. Senior Voice Engineer/Unified Communications Email: <mailto:Marjorie@gwnwireless.com> Marjorie(a)gwnwireless.com Website: <http://www.gwnwireless.com/> http://www.gwnwireless.com Office (301)674-5933

1 0

how to track down issue with shared lines
by mark li 10 Apr '14

10 Apr '14

I have a scenario where calls that are transferred to a shared line doesn't trigger the "in use" light to show up on both phones that have the shared line. but when i make an outgoing call, let's say, using the shared line, the "in use" light lights up on both phones that have the shared line. Phone - polycom 650s. Just a noob with SIP, kamailio and telecom, so not sure where to start debugging. Any suggestions would be appreciated. thanks in advance for your time.

1 0

Load fr_inv_timer from DB
by Igor Potjevlesch 10 Apr '14

10 Apr '14

Hello, I try to set the "fr_inv_timer" dynamically from DB. Under 60s it works fine. But I can't set more than 60s. Here are the TM parameters: # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 5000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 60000) Here are the instructions to set dynamically: avp_db_load("$ruri/username", "$avp(s:inv_timeout)/usr_timeout"); t_set_fr($avp(s:inv_timeout)*1000); Is there anything wrong? Regards, Igor.

1 0

topoh and onhold reinvites
by Daniel Tryba 10 Apr '14

10 Apr '14

I'm having some trouble with kamailio that uses the topology hider module and an incoming call being put onhold by the caller. The INVITE containing the sdp a=sendonly is to the 172.19.162.1 (the topoh mask_ip) contact header from the 200 OK to the initial INVITE. U 109.235.36.178:5060 -> 109.235.32.40:5060 INVITE sip:172.19.162.1:5060;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*;user=phone SIP/2.0. This is handled within dialog (has_totag() && !loose_route()), which in the default config ends in a sl_send_reply("404","Not here") Explicitly handing INVITE (just relaying), results in the INVITE being sent to 172.19.162.1 (which doesn't exist) U 109.235.32.40:5060 -> 172.19.162.1:5060 INVITE sip:172.19.162.1:5060;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*;user=phone SIP/2.0. How to get this INVITE to the actual destination? To get the same result for the INVITE as for the ACK to the 200 OK? U 109.235.32.40:5060 -> 109.235.36.178:5060 SIP/2.0 200 OK. ... Contact: <sip:172.19.162.1;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*>. U 109.235.36.178:5060 -> 109.235.32.40:5060 ACK sip:172.19.162.1;line=pcs- mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy* SIP/2.0. U 109.235.32.40:5060 -> 109.235.34.107:5060 ACK sip:+31756418030@109.235.34.107:5060;transport=udp SIP/2.0. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

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ldap SSL
by Slava Bendersky 10 Apr '14

10 Apr '14

Hello Everyone, How I can specify ldap CA cert in ldap configuration ? The directive ca_list bellow is ignored. [07/Apr/2014:13:40:37 -0400] conn=20 op=-1 fd=70 closed - Peer does not recognize and trust the CA that issued your certificate. [07/Apr/2014:13:40:37 -0400] conn=22 op=-1 fd=64 closed - Encountered end of file. Current config ldap_server_url = "ldaps://ds389.network.com" ldap_bind_dn = "uid=adm,ou=People,dc=network,dc=com" ldap_bind_password = "password" authtype = simple tls = on ca_list = /etc/kamailio/cert/ca-cert_ldap.pem

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$sel(tls.peer.subject.cn) error
by Juha Heinanen 10 Apr '14

10 Apr '14

i have these two lines in config: $var(common_name) = @tls.peer.subject.cn; $var(common_name) = $sel(tls.peer.subject.cn); the latter gives error at startup: 0(24214) ERROR: <core> [select.c:316]: resolve_select(): Unable to resolve select 'tls' at level 0 0(24214) ERROR: <core> [select.c:177]: w_parse_select(): parse_select: error while resolve_select 'tls.peer.subject.cn' 0(24214) ERROR: pv [pv_select.c:45]: pv_parse_select_name(): invalid select name [tls.peer.subject.cn] 0(24214) ERROR: <core> [pvapi.c:839]: pv_parse_spec2(): pvar "sel" has an invalid name param [tls.peer.subject.cn] 0(24214) ERROR: <core> [pvapi.c:994]: pv_parse_spec2(): wrong char [)/41] in [$sel(tls.peer.subject.cn)] at [24 (5)] 0(24214) : <core> [cfg.y:3408]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 874, column 29-53: Can't get from cache: $sel(tls.peer.subject.cn) ERROR: bad config file (1 errors) based on pseudo variable wiki page, the syntax should be correct. what is wrong with $sel(tls.peer.subject.cn)? -- juha

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Re: [SR-Users] LCR
by Keith 10 Apr '14

10 Apr '14

Hi, I am having issues with trying to use multiple lcr_ids so I can route to different groups. When I reload lcr it always only shows one group. I have copied the first group exactly and just changed the number but can't work out why it's not working. Any ideas? Thanks Keith

2 1

Exported params of USRLOC module
by aawaise 10 Apr '14

10 Apr '14

Hello, I am using kamailio-3.1 with siremis. I want to ask some questions about the exported params of usrloc module defined in the link given below. http://kamailio.org/docs/modules/3.1.x/modules_k/usrloc.html#id2997066 1. These exported params export value to whom ?? Other params or siremis location table ?? 2. If any export param is not defined in kamailio.cfg. Does its default value considered automatically or it is not used at all ?? 3. In siremis location table, I am not able to show my domain name of selected online user when I click on user. I want to do it. User's contact, received and other columns are filled but not domain name :( Shall I dig deep into usrloc module or I need to follow some other path to resolve this issue ?? Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Exported-params-of-USRLOC-module-tp… Sent from the Users mailing list archive at Nabble.com.

2 1

Show Domain-Name in Siremis
by aawaise 10 Apr '14

10 Apr '14

Hello, I am running kamailio server in my set up. And have got siremis working for monitoring purposes. Problem is that in web interface, domain name of users is not shown in siremis. If a user is online, I get following credentials infront of "Client" column, Client.Name@Client's.IP How can I show domain name in siremis ?? Any help will be highly appreciated. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-Domain-Name-in-Siremis-tp12660… Sent from the Users mailing list archive at Nabble.com.

2 7

Segfault when trying to send MESSAGE through websocket with jsSIP
by Alex Villacís Lasso 09 Apr '14

09 Apr '14

I am currently trying to replace a pure-Asterisk implementation of SIP messaging through Websockets, with a Kamailio-4.1.2-based implementation. However, when I try to send a message with jsSIP, Kamailio crashes: Program terminated with signal 11, Segmentation fault. #0 0x00007f0e5cf31be3 in reg_ht_get_byuuid (uuid=0x7fff59734b00) at uac_reg.c:350 350 slot = reg_get_entry(hash, _reg_htable->htsize); Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.132.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.10.3-10.el6_4.6.x86_64 libcom_err-1.41.12-18.el6.x86_64 libgcc-4.4.7-4.el6.x86_64 libselinux-2.0.94-5.3.el6_4.1.x86_64 libtool-ltdl-2.2.6-15.5.el6.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-14.el6.x86_64 mysql-connector-odbc-5.1.5r1144-7.el6.x86_64 mysql-libs-5.1.73-3.el6_5.x86_64 nss-softokn-freebl-3.14.3-9.el6.x86_64 openssl-1.0.1e-16.el6_5.4.x86_64 unixODBC-2.2.14-12.el6_3.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 0x00007f0e5cf31be3 in reg_ht_get_byuuid (uuid=0x7fff59734b00) at uac_reg.c:350 #1 0x00007f0e5cf36c71 in uac_reg_lookup (msg=0x7f0e6271e790, src=0x7fff59734b00, dst=0x7f0e6267c950, mode=0) at uac_reg.c:924 #2 0x00007f0e5cf2f991 in w_uac_reg_lookup (msg=0x7f0e6271e790, src=0x7f0e6267e0d0 "\a", dst=0x7f0e6267c950 "\006") at uac.c:560 #3 0x0000000000419bf6 in do_action (h=0x7fff59735690, a=0x7f0e6267f1e0, msg=0x7f0e6271e790) at action.c:1111 #4 0x0000000000422878 in run_actions (h=0x7fff59735690, a=0x7f0e6267e510, msg=0x7f0e6271e790) at action.c:1599 #5 0x0000000000417900 in do_action (h=0x7fff59735690, a=0x7f0e62664aa0, msg=0x7f0e6271e790) at action.c:715 #6 0x0000000000422878 in run_actions (h=0x7fff59735690, a=0x7f0e6265d3b8, msg=0x7f0e6271e790) at action.c:1599 #7 0x0000000000423017 in run_top_route (a=0x7f0e6265d3b8, msg=0x7f0e6271e790, c=0x0) at action.c:1685 #8 0x00000000004a5153 in receive_msg ( buf=0x7f0e570d0168 "MESSAGE sip:admin@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: <sip:admin@gatitos.com>\r\nFrom: \"Alex Villacís Lasso\" <sip:avillacisIM@gatitos."..., len=585, rcv_info=0x7f0e570cfe90) at receive.c:212 #9 0x00007f0e5c8e5802 in ws_frame_receive (data=0x7fff59735a40) at ws_frame.c:652 #10 0x000000000045531c in sr_event_exec (type=10, data=0x7fff59735a40) at events.c:254 #11 0x000000000052e04e in ws_process_msg ( tcpbuf=0x7f0e570d0160 "\201\376\002I\032\327\302\344MESSAGE sip:admin@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: <sip:admin@gatitos.com>\r\nFrom: \"Alex Villacís Lasso\" <sip:avillacisIM@"..., len=593, rcv_info=0x7f0e570cfe90, con=0x7f0e570cfe78) at tcp_read.c:1146 #12 0x000000000052e21f in receive_tcp_msg ( tcpbuf=0x7f0e570d0160 "\201\376\002I\032\327\302\344MESSAGE sip:admin@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: <sip:admin@gatitos.com>\r\nFrom: \"Alex Villacís Lasso\" <sip:avillacisIM@"..., len=593, rcv_info=0x7f0e570cfe90, con=0x7f0e570cfe78) at tcp_read.c:1182 #13 0x000000000052f2f3 in tcp_read_req (con=0x7f0e570cfe78, bytes_read=0x7fff59735cac, read_flags=0x7fff59735ca4) at tcp_read.c:1383 #14 0x0000000000530d27 in handle_io (fm=0x7f0e62724d30, events=1, idx=-1) at tcp_read.c:1617 #15 0x00000000005296c0 in io_wait_loop_epoll (h=0x8df220, t=2, repeat=0) at io_wait.h:1092 #16 0x0000000000531650 in tcp_receive_loop (unix_sock=53) at tcp_read.c:1728 #17 0x0000000000523c21 in tcp_init_children () at tcp_main.c:4959 #18 0x000000000046d6a3 in main_loop () at main.c:1702 #19 0x000000000047030b in main (argc=13, argv=0x7fff59736178) at main.c:2533 Is this a known bug?

2 8

message 484
by Slava Bendersky 09 Apr '14

09 Apr '14

Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ? <--- SIP read from UDP:192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" <sip:1300@networklab.loc>;tag=as0a530a8d To: <sip:192.168.100.145>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367(a)networklab.loc CSeq: 102 OPTIONS Contact: <sip:1300@192.168.100.145:5062>;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 Slava.

3 19

cnxcc
by Kelvin Chua 09 Apr '14

09 Apr '14

i am trying to use cnxcc for the first time. kamctl kamcmd cnxcc.active_clients client_id:6,number_of_calls:1,concurrent_calls:0,type:1,max_amount:0.000000,consumed_amount:0.000000; i don't think this is correct, seems like the dialog is not disengaging after i hang up. i created dialogs alongside cnxcc just to compare, all dialogs were disengaged. is this an expected behavior? It's also weird that the max_amount is 0 while i checked that there is a value for credit being passed as an argument. Kelvin Chua

4 7

rfc: evapi architecture and message format
by Daniel-Constantin Mierla 09 Apr '14

09 Apr '14

Hello, (cross-posting because the results are impacting users) recently I added a new module named evapi: - http://www.kamailio.org/docs/modules/devel/modules/evapi.html The purpose is to able to send custom messages to applications that connect to kamailio via tcp on a special port (different than sip port) as well as receive messages and react on them. One of the first use case we look at is integration with cgrates for real time billing. For example: - send notification when a new call comes in and cgrates will return allow/reject - send notification when the call starts so cgrates can start billing - send notification when the call ends so cgrates will close bllling The module is now in quite good shape, sending and receiving messages. Sending is done from config file, in the future some modules might do it from inside. Received message is passed to config file via an event_route and variables. Now, the point of this discussion is to debate following points: 1) should there be a specific format used for this messages? The examples in the README use json. The module can encapsulate them in netstring format when sending over tcp to be easy to parse and read the message at once. This is optional, the message can be sent as it is 2) now messages are sent to all connected applications, should be there some subscription mechanism per event? If yes, then there has to be a way to specify the event, so point 1) might need a specific format 3) handle response - should be there a defined api that will say allow, reject, drop? I was thinking that the return could be some script for an embedded interpreter (e.g., Lua) that can be executed from config file. Or, alternative, manually scripted in the event route, so each user defines its own api 3) other suggestions for what should be discussed? Looking forward to opinions regarding these matters. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Kamailio IMS dnszone
by Andrea Meroni 09 Apr '14

09 Apr '14

Issue fixed: i just had to add the following line dns_try_naptr=yes to all the kamailio.cfg files of each of my IMS entity. I suggest to modify accordingly the config files in each of the examples (pcscf, icscf, scscf) in the master tree in order to make them properly working with the kamailio_ims_dnszone example Regards, Andrea

1 0

Dialog: Keep-alive issue:
by Vassilis Radis 08 Apr '14

08 Apr '14

Hello, I am trying to use the dialog module for dealing with some cases where clients do not send BYE messages correctly and just disappear. For this reason i use the ka_timer and ka_interval along with ka-dst and ka-src attributes. I am using the master branch. I have setup the following test: I have setup a sipp instance as a callee uas configured to discard OPTIONS messages. Kamailio serves as a proxy between this sipp instance and a VoIP device registered to kamailio. Kamailio is also configured to send every call to the sipp uas with stateful proxing and dialog support: CALLER ------> KAMAILIO -----> CALLEE (sipp / ignores OPTIONS msg) I have the following issue: 1. Dialog does send the OPTIONS messages to caller and callee, and as expected, only the caller responds (with a 200 OK response). But the dialog module never terminates the dialog as it should (because the callee never responds to the OPTIONS msg). Instead it keeps sending those OPTIONS for ever. It is like there is no timer entry in the timer list for those OPTIONS. I looked a bit in the source code and I am trying to find where a timeout for those keep alives is a)set and b)handled . I see that the dlg_timer_routine is called every second but after turning debugging on, i see that the only timer reported is the default timeout for the call which i ve set to 3600 secs in the module parameters. Any insight or advice? Thanks a lot.

2 3

uac_replace_from
by Kelvin Chua 08 Apr '14

08 Apr '14

Has anybody experienced having uac_replace_from() concatenate the old from: value after the new domain? for example: From: "user1" <user1(a)domain.com> becomes From: "user2" <user2(a)domain.comuser1> Kelvin Chua

5 7

Kamailio IMS dnszone
by Andrea Meroni 08 Apr '14

08 Apr '14

Hi all, i have a strange problem: i configured a basic IMS infrastructure Kamailio-based, with a DNS server (bind9) running on a virtual machine with IP address 192.168.100.55. Whenever i try to register a user, the first registration almost always fails, while the second is successful. I attached my dnszone config used with bind9 and the pcaps of the 2 regs; my questions are - is the IMS dnszone configuration correct? - the problems i got are just due to latency issues or have i misconfigured something? Regards, Andrea

1 0

RTP not coming via RTPProxy
by varun pratapsingh 08 Apr '14

08 Apr '14

HI All, I am using the Kamailio 4.1.2 with RTPProxy 1.2.1 as a SIP Proxy. I want to ensure that all my sip and media (RTP) passes through the SIP Proxy (Kamailio + RTP Proxy) while we my SIP Client sends any SIP and RTP to SIP Server (asterisk). So it is like below: SIP Agent <--------SIP+RTP---------->SIP Proxy(Kamailio+RTPProxy)<------SIP+RTP-------->SIP Server(Asterisk). Now it is working fine with my current configuration on Kamailio with RTPProxy with SIP and RTP both if my SIP Agent is on the 802.11 LAN that is my PC have local IP address like 192.168.1.2. Now I am facing the problem in the case where my PC is accessing the internet through the USB data card (or Internet USB dongle). Here my PC IP is like 116.203.51.209. In this case my SIP is successfully passes through the SIP Proxy and works good. But in this case my RTP is does not passes through the SIP Proxy (RTPProxy). The SIP Proxy(Kamailio + RTPPRoxy) and SIP Server is running on the public IP. Please find attached my kamailio.cfg. Please let me know what is the issue. Thanks and Regards Varun

1 0

No audio issue
by jaflong jaflong 08 Apr '14

08 Apr '14

Hi, I am at the point where connection is established and no apparent errors are reported. However audio is not output. The rtp traffic seems to be transfering between the points as conclueded because Asterisk debug log shows Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021868, ts 221760, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001383, ts 1917269534, len 000160) Sent RTP packet to 10.1.xxx.xxx41143 (via ICE) (type 08, seq 021869, ts 221920, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001384, ts 1917269694, len 000160) Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021870, ts 222080, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001385, ts 1917269854, len 000160) And the browser machine on the other endpoint on a tcpdump does shows traffic on the port (41143) What could be causing there to be no audio? This is the connected sdp =0 o=root 350315728 350315728 IN IP4 10.31.xxx.xxx s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.31.xxx.xxx t=0 0 m=audio 24316 UDP/TLS/RTP/SAVPF 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=ice-ufrag:1c5c5d52130f06fd70e1e23f0d6323f2 a=ice-pwd:12611b8146599a9019d59b4b649a7970 a=candidate:Ha1f026f 1 UDP 2130706431 10.31.xxx.xxx 24316 typ host a=candidate:Ha1f026f 2 UDP 2130706430 10.31.xxx.xxx 24317 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8 a=sendrecv

2 2

P-CSCF using MYSQL database
by Andrea Meroni 08 Apr '14

08 Apr '14

Hi all, i am trying to configure an IMS Core Kamailio-based: while configuring the P-CSCF, i set up the parameters 'db_url' and 'db_mode' of the module ims_usrloc_pcscf respectively to 'mysql://kamailio:kamailiorw@localhost/kamailio' and 1 (because i want a DB for persistence). I had created successfully the DB using kamdbctl, and loaded the module 'db_mysql' in kamailio.cfg. Then i tried to run Kamailio, and i got this error: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query: driver error on query: Unknown column 'aor' in 'field list' What am i missing? Consider then when i created the DB i answer yes to all the questions about creating additional tables ... Andrea

2 24

sql_xquery() and xavp checks
by Alex Balashov 08 Apr '14

08 Apr '14

Hi, When using sql_xquery() like this: sql_xquery("ca", "SELECT * FROM gateways", "gateways"); ... what's a good way to check if any rows were returned? Since one does not have a $dbr(gateways=>rows) value in this scenario, what should one do? - is_avp_set("$xavp(gateways=>id)")) does not appear to operate on XAVPs, or at least, the fixup functions reject them: ERROR: avpops [avpops.c:935]: fixup_is_avp_set(): bad attribute name <$xavp(gateways=>id)> - the 'defined' operator does not appear to return a negative condition here: if(!defined $xavp(gateways=>id)) This condition evaluates to true. Much appreciated! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

4 14

kamailio / rtpproxy - remove codec
by Oliver Roth 08 Apr '14

08 Apr '14

Hi all We use kamailio 3.3.7 and rtpproxy for enduser call-termination. In case of a fax call, we get an invite from our carrier for codecs G711a/u and T38. As our termination carrier does not support T38 and because the invite contains G711 and T38 we get back error 488. How is it possible to remove the whole T38 part of this invite? We tried sdp_remove_codecs_by_name(list) without success - what "name" should we use for T38? [T38, t38, t.38, T.38, ...] sdp_remove_line_by_prefix(string) sdp_remove_media(type) None of these functions did really work - best was the last one with type=image but then the sip header is malformed. As we saw with Kamailio version 4.1.x there are a lot of new functions within sdpops. Would an upgrade help? So basically the question is: How to remove the t38 part of the fax invite? (see attachment) KR, Oli [cid:image001.png@01CF523C.988433F0]

2 3

SIP 480 - Service unvailable
by Amogh meshram 07 Apr '14

07 Apr '14

Hello everyone, >From today i started getting SIP 480 error when making outbound call. I checked out with a different voip provider and got SIP 500 error. Internet calling is working fine. The service (outbound calls) was working fine till yesterday. I'm a newbie to kamailio and voip, so need your help. Please guide me to resolve this error. Thanks in advance.

2 11

SIP client TCP behaviour
by Alex Balashov 07 Apr '14

07 Apr '14

Hello, I don't know a lot about SIP with TCP, so I thought I'd ask for some opinions here. I've been testing Kamailio with TCP against the Android SIP client (4.4.2) and have found, to my annoyance, that it sends a RURI with a transport attribute in its INVITEs: INVITE sip:14045551212@sip.evaristesys.com;transport=tcp My upstream gateways are all UDP, so this causes Kamailio to attempt to contact them all via TCP (and fail). I know how to bend the transport with Kamailio, and I can strip this attribute. My question is more methodological: is this "correct" for a client to do? In my opinion, the answer is no. The client should not be so presumptuous. It should specify the transport in its Contact, to indicate how it wants to be reached, and leave other decisions about transport to the UAS. But is there something I am perhaps missing? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 4

TLS connection with short lifetime - how to handle it?
by Dmytro Bogovych 07 Apr '14

07 Apr '14

Greetings. I have the next problem: iOS based clients connect via TLS to kamailio server. They run mostly in background mode - it means connection refresh interval is ~10 minutes. Some of clients reside behind paranoidal routers which considers such idle connections as lost and closes them. I see only way to resolve it is to send OPTIONS/NOTIFY/CRLF from kamailio to clients. Tried to use nat_traversal module by calling nat_keepalive() to every REGISTER message. But there is no incoming keepalive (OPTIONS or NOTIFY) on client side. Is there another way to make "heartbeat" in SIP/TLS from server side? Thank you.

1 0

One-way Relay through RTPproxy
by Nandini madhu 07 Apr '14

07 Apr '14

Dear All, I am running Kamailio (4.0) and RTPproxy server (1.2.1) server in my set-up. When i call from Client A to B , the RTP packets are proxying trhrough RTPproxy service to other end client, but no flow from client B to client A. With this calls Are unsuccessful. I traced a SIP capture (please find in the attachment) and analysed it, as in there only one client 's (A) IP address is changed to RTPproxy IP address in SDP body. What can be do to resolve this issue ? Please anybody help me And also find the attachment for my kamailio config file, and suggest me or point me about any misplacement of NAT traversal function call in the script. Please consider this and help me out as i am newbie in scripting. Regards, Nandini

2 2

Show Domain Name in Siremis
by aawaise 07 Apr '14

07 Apr '14

Hello, I have a Kamailio-3.1.1 running with its DB on MySQL running on a seperate machine. Siremis is installed for interaction with DB. In web interface of Siremis, domain name of users is not shown. How can I show that ?? Like my domain name is "defg" and user is "abc". It shows online users as "abc@HostNameOfUser" instead of "abc(a)defg.com". Secondly does Siremis read information from Kamailio server or directly from DB server ?? Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-Domain-Name-in-Siremis-tp12653… Sent from the Users mailing list archive at Nabble.com.

1 0

Restricting Kamailio to single login
by MrIhaveAnOpinionOnEverything 05 Apr '14

05 Apr '14

Hi Everyone: I am a newbie in kamailio. Basically I am looking for a way to restrict the registration or login in kamailio per user to just one. The experience I want to setup is to unregister an older register/login to kamailio by a new login/register after the register command has already passed the authentication leg. Is there an example for this? Thanks. -- "When I look at you I see two people, ther person you are and the person you are supposed to be. Someday these two people will meet. And when they do, they will achieve great things" - Gene Hackman, The Replacements Catch up on me and check my BLOG at http://mrihaveanopiniononeverything.blogspot.com

3 2

Kamailio Freeswitch
by MrIhaveAnOpinionOnEverything 05 Apr '14

05 Apr '14

To whom it may concern: We are configuring a SIP platform with Kamailio and Freeswitch with this setup: UAC 1 ==> Kamailio ==> Freeswitch UAC 2 <== Kamailio <== Freeswitch We followed the instructions in http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc We are using Kamailio on kamailio 4.0.3 and Freeswitch FreeSWITCH Version 1.5.6b Everything is working fine with the default setting. But when we configured TLS what we noticed some errors. UAC configured with no TLS calling a UAC configured with TLS works ok. UAC configured with TLS calling a UAC configured with TLS gets disconnected 15-30 seconds after answering the call. UAC configured with TLS calling a UAC configured with no TLS gets disconnected 15-30 seconds after answering the call. I cant seem to find any error in the logs or in the ngrep. Can anyone assist me regarding this issue. Good day. -- "When I look at you I see two people, ther person you are and the person you are supposed to be. Someday these two people will meet. And when they do, they will achieve great things" - Gene Hackman, The Replacements Catch up on me and check my BLOG at http://mrihaveanopiniononeverything.blogspot.com

2 3

Problem on dialog and reInvite
by pars3c 05 Apr '14

05 Apr '14

Hi I've a problem of this type: On the first INVITE is all ok, insert record route with cookie of the dialog: Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.1661> The I recieive a 422 for the session timer too small . I resend the new invite with correct expire but with a new cookie dialog value Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.2661> Then the call is connected (Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.2661>). The there is a reInvite for the session timer , the script enter in the routine WITHINDLG . At this point the dialog seems to be lost. The accounting is made with dialog variable and in the 2 records INVITE and BYE , the fields corresponding to these values are missing. In all the other calls without the reInvite for session timers, the dialog is found and the variables are present. I don't understood what is problem ? Thanks for the help

1 1

Kamailio 4.1 and Cisco 79xx Integration Work
by Jonathan Hunter 04 Apr '14

04 Apr '14

Hi Guys, I'm doing some work, with Legacy Cisco phones, its a long story but I need to manipulate signaling due to firmware limitations. I am looking to implement a workaround, whereby I need to modify an ACK, so that it sends the RURI is sent to port 5060, and the destination Port at the IP layer is unchanged, I can do this with static values as below; sethostport("1.2.3.4:5060"); t_relay("1.2.3.4","38596"); In terms of doing this for multiple sites I need to use pseudo variables, so I wondered how this could be achieved, as t_relay or forward dont support dynamic values to this point, or do they? Any comments would be great. Many thanks Jon

2 2

Record-Route headers all over the place
by Phillman25 Kyriacou 04 Apr '14

04 Apr '14

Hi im receiving the following INVITE on my system, you can see that the Record-Route headers are all over the place, is there a way to group the Record-Route headers together followed by the via headers grouped together? INVITE sip:22000017@81.21.42.184;uniq=DBFF66459C85D6171B30016D2422B SIP/2.0 Record-Route: <sip:81.21.32.84;lr=on;ftag=as4c938f35;did=6da.a472> Record-Route: <sip:81.21.38.34;lr=on;ftag=as4c938f35> Via: SIP/2.0/UDP 81.21.32.84;branch=z9hG4bK6a96.dc3413ac122b4d4b8fd61bf19d95b1cf.0 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bK6a96.c81d2c2ad30db0820f05268ea67896f7.0 Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-477861-22030313-22000017-74007 Record-Route: <sip:81.21.38.5;lr=on;ftag=as4c938f35> Record-Route: <sip:81.21.38.34;lr=on;ftag=as4c938f35;did=6da.0ff1> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bK6a96.5d12e421943fbaec75df9a6a8091d336.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7e570efe;rport=5060 Max-Forwards: 16 From: <sip:22030313@192.168.10.189>;tag=as4c938f35 To: <sip:22000017@81.21.38.34> Contact: <sip:22030313@192.168.10.189:5060> Call-ID: 74d7379b710c82f956d7cb31615c2581@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(11.7.0) Date: Fri, 04 Apr 2014 11:33:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 P-hint: outbound P-hint: outbound v=0 o=root 1396887739 1396887739 IN IP4 81.21.38.36 s=Asterisk PBX 11.7.0 c=IN IP4 81.21.38.36 t=0 0 m=audio 42746 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv thanks in advance! Phillip

2 1

What going on this SDP
by jaflong jaflong 04 Apr '14

04 Apr '14

Hi List, Can anyone help me understand why this is getting rejected Please note the specific message further dow the log. "Failed to parse SessionDescription. Failed to parse audio codecs correctly" This is on Chrome. On Firefox There is a further message in the console "Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning: Transport protocol type unsupported (UDP/TLS/RTP/SAVPF). | SDP Parsing Error: Invalid port format (17296) specified for transport protocol (Unsupported), parse failed." JsSIP | RTC SESSION | got local media stream jssip-0.3.0.js:3414 JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | TRANSPORT | sending WebSocket message: INVITE sip:9822@10.1.1.101 SIP/2.0 Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581 Max-Forwards: 69 To: <sip:9822@10.1.1.101> From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 INVITE Contact: <sip:ce5egl03@flogvr403sb2.invalid;transport=ws;ob> Allow: ACK,CANCEL,BYE,OPTIONS,INVITE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 1744 v=0 o=- 3746191339358890844 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 m=audio 65223 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 10.10.10.63 a=rtcp:65223 IN IP4 10.10.10.63 a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0 a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0 a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0 a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0 a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0 a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0 a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0 a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0 a=ice-ufrag:Dgp8HIJdmr1lFPCQ a=ice-pwd:2yYxerrscdbTQhr0vbCTiju9 a=ice-options:google-ice a=fingerprint:sha-256 C8:E9:57:CB:85:63:F7:C5:FC:15:3D:8B:A8:10:94:F4:C9:BB:86:48:E3:EE:A0:5E:FA:42:14:55:6F:68:3F:B7 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:3445528109 cname:QbLF+sVLqHbEqUxY a=ssrc:3445528109 msid:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 768e2e47-bc86-473d-bc2c-6e2340ace772 a=ssrc:3445528109 mslabel:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 a=ssrc:3445528109 label:768e2e47-bc86-473d-bc2c-6e2340ace772 jssip-0.3.0.js:519 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581;rport=56527;received=10.10.10.63 To: <sip:9822@10.1.1.101> From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 INVITE Server: DXI WebRTC Content-Length: 0 Warning: 392 10.10.10.48:6443 "Noisy feedback tells: pid=23455 req_src_ip=10.10.10.63 req_src_port=56527 in_uri=sip:9822@10.1.1.101 out_uri=sip:9822@10.10.10.111:5443 via_cnt==1" jssip-0.3.0.js:670 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WSS flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK9149581 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov To: <sip:9822@10.1.1.101>;tag=as06b3db08 Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 INVITE Server: Easycall Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:9822@10.10.10.111:5443;transport=TLS> Content-Type: application/sdp Content-Length: 801 v=0 o=root 431209641 431209641 IN IP4 10.10.10.111 s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.10.10.111 t=0 0 m=audio 30490 UDP/TLS/RTP/SAVPF 0 126 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=ice-ufrag:5236e84b43b5d10c117e8ead0a340138 a=ice-pwd:51005fde4e6e9d3f1879fbbc15e0f092 a=candidate:Ha1f026f 1 UDP 2130706431 10.10.10.111 30490 typ host a=candidate:S5bec7504 1 UDP 1694498815 91.236.117.4 30490 typ srflx a=candidate:Ha1f026f 2 UDP 2130706430 10.10.10.111 30491 typ host a=candidate:S5bec7504 2 UDP 1694498814 91.236.117.4 30492 typ srflx a=connection:new a=setup:active a=fingerprint:SHA-256 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8 a=sendrecv jssip-0.3.0.js:670 Failed to parse SessionDescription. Failed to parse audio codecs correctly. jssip-0.3.0.js:4512 JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-0.3.0.js:2523 JsSIP | TRANSPORT | sending WebSocket message: ACK sip:9822@10.10.10.111:5443;transport=tls SIP/2.0 Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK431640 Max-Forwards: 69 To: <sip:9822@10.1.1.101>;tag=as06b3db08 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 ACK Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 jssip-0.3.0.js:519 JsSIP | TRANSPORT | sending WebSocket message: BYE sip:9822@10.10.10.111:5443;transport=tls SIP/2.0 Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK7689766 Max-Forwards: 69 To: <sip:9822@10.1.1.101>;tag=as06b3db08 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7436 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 jssip-0.3.0.js:519 JsSIP | RTC SESSION | closing INVITE session 43oclsi0sva6n347bk5c6tmeble9ov jssip-0.3.0.js:4193 JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.js:3392 JsSIP | DIALOG | dialog 43oclsi0sva6n347bk5c6tmeble9ovas06b3db08 deleted jssip-0.3.0.js:2543 JsSIP | EVENT EMITTER | emitting event failed jssip-0.3.0.js:187 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WSS flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK7689766 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov To: <sip:9822@10.1.1.101>;tag=as06b3db08 Call-ID: 43oclsi0sva6n347bk5c CSeq: 7436 BYE Server: Easycall Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0

4 6

Load new module in Kamailio
by Patrik Kristel 04 Apr '14

04 Apr '14

Hello all, I would like to ask, how can I load new module in Kamailio 4.1.2? Actually, I have an issue, when I tried to compile my kamaiio.cfg I've got error: root@kamailio:/usr/local/# kamailio -c kamailio.cfg loading modules under /usr/local/lib64/kamailio/modules/ 0(25392) ERROR: <core> [sr_module.c:587]: load_module(): ERROR: load_module: could not find module <websocket> in </usr/local/lib64/kamailio/modules/> 0(25392) : <core> [cfg.y:3408]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 323, column 12-25: failed to load module 0(25392) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command ws_handle_handshake 0(25392) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 1083, column 27: unknown command, missing loadmodule? ERROR: bad config file (2 errors) when I look physically to the /usr/local/lib64/kamailio/modules/ there is some modules, but websocket.so is missing. So, how can I get and load module in Kamailio? Thank you for help! BR, Patrik Kristel

2 1

Crash on REGISTER
by Igor Potjevlesch 03 Apr '14

03 Apr '14

Hello, I experienced an issue yesterday. I move a customer to Kamailio 4.0.4 (previously he was on an old SER instance without issue). He uses the same SIP account for connecting two IPBX (same platform, same firmware). When the second REGISTER came on Kamailio and, may be 500ms/1s after (I have sufficient time to see 200 OK (2 bindings) on Wireshark), Kamailio crash with coredump generated. I know that I must analyse the coredump but I hope that could be a known issue. In addition, when I try to run "gdb" on the core files, I got the following: Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -m 256 -M 64'. Program terminated with signal 11, Segmentation fault. #0 0x00000030f2230f30 in escape_string_for_mysql () from /usr/lib64/mysql/libmysqlclient.so.16 Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.9-33.el6.x86_64 libcom_err-1.41.12-12.el6.x86_64 libconfuse-2.6-2.el6.rf.x86_64 libselinux-2.0.94-5.3.el6.x86_64 mysql-libs-5.1.61-4.el6.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64 openssl-1.0.0-20.el6_2.5.x86_64 zlib-1.2.3-27.el6.x86_64 I guess that I'm missing packages to troubleshoot. Many thanks. Regards, Igor.

3 10

NAT Traversal issue
by Wingsravi R 03 Apr '14

03 Apr '14

Dear Kamailio'ns, I am working on Kamailio server (V 4.1.2) with RTPproxy (1.2.1) integrated, in a standalone intranet infrastructure (no any connection with internet). I dont have any NAT settings in my network set-up. even though i will not get Audio/video calls through some times. So in that concern I have installed RTPproxy, now all the audio/Video calls are fine (with some Pixelled). I have the Following Kamailio configuration script, in which it suppose to invoke RTPproxy service when the SIP clients behind NAT. But every time when i do Audio/Video calls, they are proxying through RTPproxy server only. I analysed SIP captures of Audio/video call, i didnt found any IP/port changes in the whole SIP session and with this i assumed that there is no NAT issue in my Network. But why all the Audio/Video calls are proxying through RTPproxy everytime ? Is there any Wrong placement of function call in Kamailio configuration script (below) ? #----------------------------------------------------------------- #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@192.168.2.52") modparam("nathelper", "sipping_method", "INFO") # ----- NAT_traversal ----- modparam("nat_traversal", "keepalive_interval", 60) modparam("nat_traversal", "keepalive_method", "NOTIFY") modparam("nat_traversal", "keepalive_state_file", "/var/run/kamailio/keepalive_state") # ----- params needed for NAT traversal in other modules ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", 6) #!endif #Routing Script # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; #if (msg:len > max_len) { if (msg:len >= 8192 ) { sl_send_reply("513", "Message too big"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #if (method!="REGISTER") { if (!method=="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { # unforce_rtp_proxy(); rtpproxy_answer(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("192.168.2.52","subscriber")) { # proxy_challenge("","0"); proxy_challenge("192.168.2.52", "0"); # break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; rtpproxy_offer("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # if (uri!=myself) { if (!uri==myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { rtpproxy_answer();; }; sl_reply_error(); }; } # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- route[2] { if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("192.168.2.52","subscriber")) { www_challenge("192.168.2.52","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- route[3] { if (!proxy_authorize("192.168.2.52","subscriber")) { proxy_challenge("192.168.2.52","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- route[4] { if (isflagset(6)) { force_rport(); fix_nated_contact(); # force_rtp_proxy(); rtpproxy_offer(); } } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { # force_rtp_proxy(); rtpproxy_offer(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } Please find the Attachment for Tcpdump based Video call Sip capture for your better Understanding. PS: Both Kamailio and RTPproxy are running on same IP (host), i.e 192.168.2.52. Please anybody help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi.

3 2

when dialing another extension, instead of ringing for 30 sec, vmail kicks in right away
by mark li 03 Apr '14

03 Apr '14

I'm using kamailio as a sip proxy and registrar, and freeswitch for all media services like vmail, conference calls, ivrs etc. I just set up vmail ... and it seems to be working. Except that it appears that I've broken something else because now when I call extB from extA, instead of ringing first for a little while and then having the vmail kick in afterwards, vmail kicks in right away. someone mentioned that it could be a forking problem. I'm afraid I don't really understand what this means. just wondering if you can point me in the right direction list, as far as what I should be looking for in the kamailio.cfg file... Thanks.

2 2

error messages in kam log whenever i do anything with my phone
by mark li 02 Apr '14

02 Apr '14

Hi there. I've noticed that no matter what i do with my test phone (call voicemail, call another extension etc) I get error messages like the following in my log file: Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=553 a=26 n=xlog Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=560 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=557 a=25 n=is_method Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=563 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=560 a=25 n=is_method Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=567 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=563 a=25 n=is_method Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=570 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=567 a=24 n=t_relay Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1064 a=25 n=xdbg Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1067 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1071 a=5 n=route Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=854 a=2 n=return Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1088 a=16 n=ifApr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1073 a=24 It doesn't seem to impact the operation of the phone / dialplans etc. I recall reading somewhere that it might have to do with caller ids or location. I'm trying to google to see if I can find the article the made references to similar error messages. But in the meantime, I thought I'd check the list to see if you could point me in the right direction.

2 1

Re: [SR-Users] Regarding Websocket module
by Juha Heinanen 02 Apr '14

02 Apr '14

Premchandiran writes: > Not able to find libunistring-dev rpm for Linux 2.6.18-274.el5. almost 4 > hours I am searching for the rpm. switch to debian, -- juha

2 1

Understunding the routing logic with WebSockets
by DURECU, OLIVIER (OLIVIER) 02 Apr '14

02 Apr '14

Hi experts, Using websockets, I am facing a behavior I can't manage explaining : on reception of an initial REGISTER (over websocket), it seems that the routing logic in the request_route method is not invoked Extract of my Kamailio.cfg to explain =================================================================== Request_route { xlog("L_DBG", "Entering request_route for $rm") ... Onreply_route[REGISTER_reply] { If (t_check_status("200")) xlog("L_DBG", "REGISTER OK") Event_route[xhttp:request] { xlog("L_DBG", "XHTTP Request received") ws_handle_hanshake(); In my logs, I only see > XHTTP Request received > REGISTER OK No trace of "Entering request route for REGISTER" ! whereas for a subsequent INVITE, I see the trace Any explanation ? because of this, the fix_nated_register() is not called, and I suspect this causes the troubles I meet trying to make call with websockets Olivier DURECU Bell Labs / IP Platform Research Villarceaux Center Tel: +33(0) 130772759 GSM: +33(0) 683776482

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R: Reply route and variable
by pars3c 02 Apr '14

02 Apr '14

Hello, Probably i dont see the dialog variable in the onreply route , because I see on the trace ip that sometime on the reInvite the proxy dont put on the record route the dialog cookie. On the first invite , its ok , but on every reinvite (we have sip session timer) there isnt. Its correct ? Thanks Da: pars3c [mailto:pars3c@gmail.com] Inviato: martedì 1 aprile 2014 17:32 A: 'Kamailio (SER) - Users Mailing List' Oggetto: Reply route and variable Hi , i have a question for all : in the onreply_route is visible a dialog variable ($dlg_var(ring_time)) or a avp variable ? Can I set them , on this kind of route ? On the event route (dialog start) they are all null. I try to set them, but some time (on reInvite) this variables are not setted. Why? Thanks to all

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Re: [SR-Users] Regarding Websocket module
by DURECU, OLIVIER (OLIVIER) 02 Apr '14

02 Apr '14

Have a look at http://rpm.pbone.net/index.php3/stat/3/limit/1/srodzaj/1/dl/40/search/libun…<http://rpm.pbone.net/index.php3/stat/3/limit/1/srodzaj/1/dl/40/search/libun…> and search for your OS version through the pages Olivier De : sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] De la part de Premchandiran Envoyé : mercredi 2 avril 2014 09:17 À : 'Kamailio (SER) - Users Mailing List' Objet : Re: [SR-Users] Regarding Websocket module Hi Olivier, Not able to find libunistring-dev rpm for Linux 2.6.18-274.el5. almost 4 hours I am searching for the rpm. Thanks, Prem From: sr-users-bounces(a)lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org> [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of DURECU, OLIVIER (OLIVIER) Sent: Wednesday, April 02, 2014 12:39 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Regarding Websocket module Hi, You may be missing a library. On Ubunutu/debian, this is the libunistring-dev package which provides the required library This is documented on the module description (modules/websocket/README) Olivier De : sr-users-bounces(a)lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org> [mailto:sr-users-bounces@lists.sip-router.org] De la part de Premchandiran Envoyé : mercredi 2 avril 2014 09:00 À : sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Objet : [SR-Users] Regarding Websocket module Hi All, I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting below error though websocket.so is present in /usr/local/lib64/modules/ load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/websocket.so>: libunistring.so.0: cannot open shared object file: No such file or directory may I know what might be the cause of this error. Regards Prem

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Re: [SR-Users] Regarding Websocket module
by DURECU, OLIVIER (OLIVIER) 02 Apr '14

02 Apr '14

Hi, You may be missing a library. On Ubunutu/debian, this is the libunistring-dev package which provides the required library This is documented on the module description (modules/websocket/README) Olivier De : sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] De la part de Premchandiran Envoyé : mercredi 2 avril 2014 09:00 À : sr-users(a)lists.sip-router.org Objet : [SR-Users] Regarding Websocket module Hi All, I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting below error though websocket.so is present in /usr/local/lib64/modules/ load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/websocket.so>: libunistring.so.0: cannot open shared object file: No such file or directory may I know what might be the cause of this error. Regards Prem

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Regarding Websocket module
by Premchandiran 02 Apr '14

02 Apr '14

Hi All, I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting below error though websocket.so is present in /usr/local/lib64/modules/ load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/websocket.so>: libunistring.so.0: cannot open shared object file: No such file or directory may I know what might be the cause of this error. Regards Prem

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Dimension a kamailio server
by Rizwan Khan 02 Apr '14

02 Apr '14

HI Guys, can you refer me to some resources which help me dimension the hardware for a setup. Total users will be 20,000. What percentage should i assume for concurrent audio and video calls? What is the standard practice? How does this all map to the cpu, ram and storage etc. given that I will be storing the cdrs as well. In the futute this setup should be scalable so that it should support 200,000+ users within a span of 6-8 months. What would be the ideal architecture? Thanks, Rizwan Khan

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loose_route() problem
by Juha Heinanen 01 Apr '14

01 Apr '14

my proxy is listening at udp/tcp ip:port 192.98.102.30:5060 and my script has this for in-dialog requests: route [IN_DIALOG_REQUEST] { # handle in-dialog request if (!loose_route()) { xlog("L_INFO", "Loose route failed on $hdr(route)\n"); ... exit; }; xlog("L_INFO", "destination uri after loose_route: <$du>\n"); if i send to my proxy with sipp this in-dialog bye: <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="In-dialog BYE"> <send> <![CDATA[ BYE sip:test@192.98.102.30:53846;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 192.98.101.40;branch=z9hG4bK63f.e9d25102.0 Via: SIP/2.0/UDP 192.98.101.157:5080;branch=z9hG4bK6955a179;rport=5080 Route: <sip:192.98.102.30;r2=on;lr;n1>,<sip:192.98.102.30;transport=tcp;r2=on;lr;n1> Max-Forwards: 16 From: <sip:jh@test.tutpro.com>;tag=as0cb9dd4f To: "Test" <sip:test@test.tutpro.com>;tag=3feea37d Call-ID: d745a79b66919309ba2ef68fde0ca13e@0:0:0:0:0:0:0:0 CSeq: 102 BYE Content-Length: 0 ]]> </send> </scenario> i correctly get to syslog: Apr 1 20:47:28 siika /usr/sbin/sip-proxy[9583]: INFO: destination uri after loose_route: <<null>> that is, the proxy removed both route uris that pointed to itself from the route set leaving empty $du. however, if i send to my proxy with sipp this in-dialog bye (where the only difference to the previous one is request uri that now contains alias param): <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="In-dialog BYE"> <send> <![CDATA[ BYE sip:test@192.98.102.30:5060;alias=192.98.102.30~53846~2;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 192.98.101.40;branch=z9hG4bK63f.e9d25102.0 Via: SIP/2.0/UDP 192.98.101.157:5080;branch=z9hG4bK6955a179;rport=5080 Route: <sip:192.98.102.30;r2=on;lr;n1>,<sip:192.98.102.30;transport=tcp;r2=on;lr;n1> Max-Forwards: 16 From: <sip:jh@test.tutpro.com>;tag=as0cb9dd4f To: "Test" <sip:test@test.tutpro.com>;tag=3feea37d Call-ID: d745a79b66919309ba2ef68fde0ca13e@0:0:0:0:0:0:0:0 CSeq: 102 BYE Content-Length: 0 ]]> </send> </scenario> i get to syslog: Apr 1 20:51:20 siika /usr/sbin/sip-proxy[9540]: INFO: destination uri after loose_route: <sip:192.98.102.30;transport=tcp;r2=on;lr;n1> can someone explain, how it is possible that when request uri changes, loose_route() does not anymore consume both route uris that point to the proxy itself? could this be a bug in loose_route()? -- juha

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rls no notify generated
by Mihai Marin 01 Apr '14

01 Apr '14

Hello Sirs, I managed to make it work but with the resource list embedded in the rls-services xml. The xml looks like that: <?xml version="1.0" encoding="UTF-8"?> <rls-services xmlns="urn:ietf:params:xml:ns:rls-services" xmlns:rl="urn:ietf:params:xml:ns:resource-lists" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"> <service uri="sip:bob-list@93.187.138.214"> <list name="bob-list"> <rl:entry uri="sip:alice@93.187.138.214"/> </list> <packages> <package>presence</package> </packages> </service> </rls-services> Now, I'm trying to make it work using an "external" resource list but it's not working. The subscribe is working, but the rls_handle_notify raise "rls_handle_notify(): cannot find content type header header". My xml looks something like this: <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="rcs"> <display-name>All Contacts</display-name> <external anchor=" http://93.187.138.214:5060/xcap-root/resource-lists/users/sip:bob@93.187.13… "/> </list> <list name="doubango"> <display-name>My Default Contacts</display-name> <entry xmlns="urn:ietf:params:xml:ns:resource-lists" uri=" sip:alice@93.187.138.214"> <display-name>bob</display-name> </entry></list> </resource-lists> OR <?xml version="1.0"?> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="rcs"> <display-name>All Contacts</display-name> <entry xmlns="urn:ietf:params:xml:ns:resource-lists" uri=" sip:alice@93.187.138.214"> <display-name>alice</display-name> </entry> </list> </resource-lists> with <rls-services xmlns:rl="urn:ietf:params:xml:ns:resource-lists" xmlns="urn:ietf:params:xml:ns:rls-services"> <service uri="sip:alice@93.187.138.214;pres-list=rcs"> <resource-list> http://93.187.138.214:5060/xcap-root/resource-lists/users/sip:alice@93.187.… </resource-list> <packages> <package>presence</package> </packages> </service> Do you know if the internal module xcap and rls has support for "external" resource list like in my previous xml - <resource-list> ...</resource-list>. Any clue why is not working? I-m using the same configuration like I did (and worked) with embedded rl in rls. Thank you. Best regards, Mihai M ---------- Forwarded message ---------- From: Mihai Marin <marinmihai(a)gmail.com> Date: Mon, Mar 31, 2014 at 8:06 PM Subject: rls, rls_get_by_service_uri(): match not found To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Hello Sirs, I'm trying from some days to make RLS work but without success. I checked every character from kamailio.cfg, xml files from xcap but I can't find the problem and I'm out of ideas. My problem is the the subscribe is not done => 0 notifications with status. I think the key error is that: DEBUG: rls [subscribe.c:125]: rls_get_by_service_uri(): match not found, service-uri = [sip:bob@93.187.138.214] 2(29025) DEBUG: rls [subscribe.c:125]: rls_get_by_service_uri(): match not found, service-uri = [sip:bob@93.187.138.214] 2(29025) DEBUG: rls [subscribe.c:125]: rls_get_by_service_uri(): match not found, service-uri = [sip:bob@93.187.138.214] 2(29025) DEBUG: rls [subscribe.c:279]: rls_get_service_list(): service uri sip:alice@93.187.138.214 not found in rl document for user sip:bob@93.187.138.214 Request: SUBSCRIBE sip:bob-list@93.187.138.214 SIP/2.0 Via: SIP/2.0/WS kvvj9i9uhvbp.invalid;branch=z9hG4bK4622972 Max-Forwards: 69 To: <sip:bob-list@93.187.138.214> From: "Bob Test" <sip:bob@93.187.138.214>;tag=e9144sdk2v Call-ID: 52k3b9gheoj6gk2vmjlrr0 CSeq: 81 SUBSCRIBE Expires: 300 Event: presence Contact: <sip:v8jmlmeu@kvvj9i9uhvbp.invalid;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:152abb8b-085d-4ca5-b658-4ecb38d1b79f>";expires=60000 Supported: eventlist Accept: multipart/related, application/pidf+xml, application/rlmi+xml, application/rpid+xml Allow-Events: refer, presence, presence.winfo, xcap-diff, conference Require: recipient-list-subscribe Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER User-Agent: JsSIP 0.3.7 Content-Length: 0 Receive: NOTIFY sip:v8jmlmeu@188.215.94.132:25702;transport=ws SIP/2.0 Via: SIP/2.0/WS 93.187.138.214:5040;branch=z9hG4bKf43b.cc179644000000000000000000000000.0 To: <sip:bob@93.187.138.214>;tag=e9144sdk2v From: < sip:bob-list@93.187.138.214>;tag=67d32ce551425daaac3772fa02abe99f-daa0 CSeq: 1 NOTIFY Call-ID: 52k3b9gheoj6gk2vmjlrr0 Content-Length: 394 User-Agent: kamailio (4.2.0-dev3 (x86_64/linux)) Max-Forwards: 70 Event: presence Contact: <sip:rls@93.187.138.214> Subscription-State: active;expires=300 Require: eventlist Content-Type: multipart/related;type="application/rlmi+xml";start="<1396291389.sip:bob-list@93.187.138.214.834374984>";boundary="TnOoxHgwMlltoGjpMlgd1kNc" --TnOoxHgwMlltoGjpMlgd1kNc Content-Transfer-Encoding: binary Content-ID: <1396291389.sip:bob-list@93.187.138.214.834374984> Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list uri=" sip:bob-list@93.187.138.214" xmlns="urn:ietf:params:xml:ns:rlmi" version="1" fullState="true"> <resource uri="sip:alice@93.187.138.214"/> </list> --TnOoxHgwMlltoGjpMlgd1kNc-- I have 2 users: bob & alice. Alice xcap files: <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="Default"> <display-name>All Contacts</display-name> <entry xmlns="urn:ietf:params:xml:ns:resource-lists" uri=" sip:bob@93.187.138.214"> <display-name>bob</display-name> </entry> </list> </resource-lists> <rls-services xmlns:rl="urn:ietf:params:xml:ns:resource-lists" xmlns="urn:ietf:params:xml:ns:rls-services"> <service uri="sip:alice-list@93.187.138.214"> <resource-list> http://93.187.138.214:5060/xcap-root/resource-lists/users/sip:alice@93.187.… </resource-list> <packages> <package>presence</package> </packages> </service> </rls-services> Bob xcap files: <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="Default"> <display-name>All Contacts</display-name> <entry xmlns="urn:ietf:params:xml:ns:resource-lists" uri=" sip:alice@93.187.138.214"> <display-name>alice</display-name> </entry> </list> </resource-lists> <rls-services xmlns:rl="urn:ietf:params:xml:ns:resource-lists" xmlns="urn:ietf:params:xml:ns:rls-services"> <service uri="sip:bob@93.187.138.214"> <resource-list> http://93.187.138.214:5060/xcap-root/resource-lists/users/sip:bob@93.187.13… </resource-list> <packages> <package>presence</package> </packages> </service> </rls-services> And, global file: <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> </namespaces> </xcap-caps> KAMAILIO log is here: http://pastebin.com/4XGSCPkc Kamailio route presence from config is here: http://pastebin.com/LYSr5yPz What I'm doing wrong? Are xml configured properly? Thank you. Best regards, Mihai M

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Re: [SR-Users] UAC Reg
by Keith 01 Apr '14

01 Apr '14

Hi all, I am trying to do an on-registration to a Freeswitch box, to to this I am trying to use uac_reg_request_to(user, mode) Does anyone have any examples with this working? The request is going to the freeswitch box OK, I can see the challenge header being built but it is not sent back to the freeswitch box. Any help appreciated. Cheers. Keith

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webrtc (websockets) to UDP and vice versa
by ik 01 Apr '14

01 Apr '14

Hello, I'm a newbie with Kamailio, and I require to connect webrtc (websockets) based phones, into a VoIP PBX that does not support websockets. I wish to create/use Kamailio rules that will translate UDP to websockets and vice versa. I have found few examples over the internet, but as it seems (to me), they are just doing normal SIP operations under websockets (registration, routing, voicemails etc). Is there a way to make Kamailio a broker that understand both transports, and translate them ? If so, can you please point me to a documentation/example that does it that might help me better understand it ? Please note that I do not have any experience with Kamailio, and just getting started with it. Thank you very much, Ido

2 1

Reply route and variable
by pars3c 01 Apr '14

01 Apr '14

Hi , i have a question for all : in the onreply_route is visible a dialog variable ($dlg_var(ring_time)) or a avp variable ? Can I set them , on this kind of route ? On the event route (dialog start) they are all null. I try to set them, but some time (on reInvite) this variables are not setted. Why? Thanks to all

1 0

Notify registered users of primary server status change
by Neven Boric 01 Apr '14

01 Apr '14

Hi, I'm using kamailio as an intermediate proxy with survivability. That means I'm monitoring the primary server using the dispatch module, and if it goes down, I take control of all further requests. The main server acts as registrar, but I keep a copy a of the registered users list. There's a feature of Siemens (now Unify) phones and SBC where in this same scenario, the SBC will send a NOTIFY packet informing the status change to the phones. The phones then display a message indicating "limited service" and re-register. To indicate that they support this feature, phones send a "Supported: X-Siemens-Proxy-State" header when registering. The NOTIFY packet from the SBC includes the header "Event: server-mode-backup" or "Event: server-mode-normal" according to the main server status. I would like to replicate this functionality in kamailio. I looked at the presence module, as it already sends NOTIFY requests, but UACs have to subscribe to events first, and Siemens phones do not. I am trying to do it manually in the event_route[dispatcher:dst-down/up] triggers, but I can't find a way to iterate through all registered users. I would appreciate some assistance: - Is there some module I'm missing that implements something similar? - If not, how to iterate through all registered users to manually send a request to each of them? - If I send the requests using uac_req_send, will they be sent sequentially from the same thread? Thanks Neven Boric

1 0

IMS and SIP over WebSocket 500/Error
by DURECU, OLIVIER (OLIVIER) 01 Apr '14

01 Apr '14

Hello all, I have installed Kamailio 4.1.2 in IMS configuration (3 Kamailio instances to run in P/I/S-CSCF modes) Everything works fine with Native IMS clients (Boghe) I want now to introduce Web clients using WebSockets. I have added to the P-CSCF Kamailio.cfg the websockets parts Registration is Ok but I can't manage placing a call. With SIPoWS, the contact header is sip:olivier@132.invalid. So when the P-CSCF receives a terminating INVITE, its destination URI is sip:olivier@132.invalid , it tries to make a DNS resolution of 312.invalid and it doesn't forward to the client saying the domain is unknown and can't be routed I guess that something is missing in my configuration around aliases/nathelper/outbound but didn't find what... Thanks for any help Olivier DURECU Bell Labs / IP Platform Research Villarceaux Center Tel: +33(0) 130772759 GSM: +33(0) 683776482

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Re: [SR-Users] Kamailio ICSCF strange forking
by Richard Good 01 Apr '14

01 Apr '14

Hi Apologies for late response on this - you are correct, there is an unnecessary append_branch() in the example config file. I've removed it from master: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=9606520… Thanks for the find - let us know if you have any other problems (or successes!) Regards Richard. On 1 April 2014 12:00, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Forward copy of Register information to FreeSWITCH > (Alexandr Usov) > 2. Re: Forward copy of Register information to FreeSWITCH > (Alexandr Usov) > 3. Re: Forward copy of Register information to FreeSWITCH > (Daniel-Constantin Mierla) > 4. Re: Forward copy of Register information to FreeSWITCH > (Frank Carmickle) > 5. Re: Forward copy of Register information to FreeSWITCH > (Daniel-Constantin Mierla) > 6. R: ERROR reply code Failure route (pars3c) > 7. Re: message 484 (Slava Bendersky) > 8. Re: message 484 (Slava Bendersky) > 9. Re: Kamailio and Asterisks work on HP-UX ? (Rizwan Khan) > 10. Re: Kamailio and Asterisks work on HP-UX ? (Alex Balashov) > 11. Re: How to turn on debugging in kamailio? (mark li) > 12. Re: Kamailio and Asterisks work on HP-UX ? (Rizwan Khan) > 13. Re: Kamailio and Asterisks work on HP-UX ? (Alex Balashov) > 14. Re: message 484 (Pedro Ni?o) > 15. Re: message 484 (Slava Bendersky) > 16. Re: Kamailio ICSCF strange forking (DURECU, OLIVIER (OLIVIER)) > 17. Re: Crash on REGISTER (Igor Potjevlesch) > 18. (no subject) (info(a)vintageelectronics.ca) > 19. rls, rls_get_by_service_uri(): match not found (Mihai Marin) > 20. Re: message 484 (Slava Bendersky) > 21. Kamailio on Android (jay binks) > 22. Social Configuration module for Kamailio released (Alex Balashov) > 23. Re: Social Configuration module for Kamailio released > (Kelvin Chua) > 24. Re: Intermittent Audio/video call (Wingsravi R) > 25. Re: Dimension a kamailio server (Rizwan Khan) > 26. Re: Dimension a kamailio server (Alex Balashov) > 27. Re: Dimension a kamailio server (Alex Balashov) > 28. Re: Dimension a kamailio server (Rainer Piper) > 29. SIP - WebRTC gateway (Zappasodi Daniele) > 30. Re: Dimension a kamailio server (Rainer Piper) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 31 Mar 2014 14:07:13 +0300 > From: Alexandr Usov <blessendor(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] Forward copy of Register information to > FreeSWITCH > Message-ID: > <CAEaKXwDcmT5urkuKxLS50fbznmTKnc=1fknAX_iCMBxHq= > igZA(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > 2014-03-28 18:16 GMT+02:00 Frank Carmickle <frank(a)carmickle.com>: > > > > > Freeswitch does not require registration. What are you trying to use > > freeswitch for? Voicemail, B2BUA (transcoding), ivr? See the acl > > configuration or don't use the directory at all. > > > > > I want use Kamailio only for registrations and use rtpproxy for registered > peers. > So all PBX features I want to use on FS (and/or Asterisk). > > So acl style can't serve Voicemail and presence features for not registered > (on FS) users? >

1 0

Intermittent Audio/video call
by Wingsravi R 01 Apr '14

01 Apr '14

Dear All, I am working on Kamailio server (V 4.1.2) with RTPproxy (1.2.1) integrated, in a standalone intranet infrastructure (no any connection with internet). I have a strange behavior with my set-up here is : --> Audio/Video calls are just go fine with only kamailio server (Without running RTPproxy instance). But with the same script and set-up sometimes i'm experiencing one-way audio/video !!. What could be the reason for this behavior ? Is it a NAT problem ? How can i solve this issue ? --> I was just suspecting it as NAT problem and installed RTPproxy server, with this all the audio/video calls are going fine. But i'm curious to know about this intermittent audio/video calls that too without any changes to Kamailio configuration and my set-up. Please below find the attachment for kamailio config file. Please anybody help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi

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Social Configuration module for Kamailio released
by Alex Balashov 01 Apr '14

01 Apr '14

For immediate release: ATLANTA, GA (1 April 2014)--Evariste Systems LLC, an Atlanta-based consultancy and software vendor specialising in Kamailio-based VoIP infrastructure solutions for the ITSP (Internet Telephony Service Provider) market, has announced the public release of its groundbreaking "Social Configuration" module for the Kamailio open-source SIP server and SIP proxy. Evariste principal Alex Balashov explained: "It's no secret that writing Kamailio configuration in its embedded scripting language is complex. Social Configuration integrates with Twitter and Facebook to allow tens of thousands of people to write a config simultaneously through social networks, over a variety of devices and media, including PCs, tablets, smartphones and SMS. This lets Kamailio users harness the benefits of crowdsourcing." Continued Balashov: "The process begins when the Kamailio administrator writes the desired intentions of their setup--yes, in natural English prose. This isn't for propellerheads. The module sends this to our proprietary QNLP (quantum natural language parsing) server, which splits these desiderata into Business Action Items (BAIs). These are then sent out to participating Facebook and Twitter users via our expansive middleware, and the users return their piecemeal script implementations, which are usually one line of code or less. These are consolidated back into a fully functional Kamailio configuration." Brooks Bridges, Director of Engineering at Evariste, explained how Social Configuration handles reconciliation of conflicting implementations from across the Twittersphere and Facebook: "This is really the heart of the technology. It's handled via a deeply recursive, Rails-based social A/B testing and multiple-choice polling system. It uses hashtags, trending and RSS and stuff. The days of merge conflicts with unified diffs are gone! Throw out your legacy version control, 'cause this is Conflict Resolution 2.0: The Feed!" Another frequently asked question lingered: assembling the contributions in a particular order. On this topic, Balashov offered a surprising answer: "Plot twist: that's social, too! Twitter is an infinitely dynamic collaboration platform. The question isn't so much 'what can social networks do for us?' as 'what can't they do?' You can see that there are many strata of social clouds here: cumulus, stratus, cirrus--we've got all the clouds!" "Xzibit would make it easier to understand for the layman," chortled Balashov as he shifted his position slightly. "The whole process uses infinitely differential recursion manifolds. It's like: 'Yo dawg, I heard you like crowds, so we put a crowd in your crowd, so you can crowd while you crowd', you know? All stages of the process happen simultaneously and in real time. It's consensus-driven development by a Committee of the Ultimate." Balashov also stated that total development time for Social Configuration was only three days, rather than the thousands of man-hours normally required to develop complex technology. When asked how this was possible, he deferred to Bridges: Bridges explained: "It's all because we embraced the Agile Full Vapour method at our company. Maybe you've heard of it. Classic Agile and 'customer-driven development' says 'release early, fail often'. That wasn't enough for us. Agile Full Vapour is Agile extended to its logical maximum. Our slogan here at Evariste Engineering is 'Release Too Early, Fail More Often'. And stand-ups? Ain't nobody got time for that. We've done away with all team communication or collaboration of any kind. We've slashed overhead to a new level, and the numbers speak for themselves. We get three thousand percent more out of our developers than anyone, anywhere else." Balashov concurred, gesturing to the Evariste office "Vapour Board": "I really don't know how legacy Agile companies get anything done!" Initial reactions to the new technology from inside the VoIP community were positive. Sean McCord, Vice President of Wholesale Hashtag and viral marketing expert at Atlanta-based social media anaytics giant CyCORE Systems, agreed that Social Configuration is an essential leap forward for the Kamailio ecosystem: "The Social Configuration module heralds the long-overdue marriage of Kamailio with the Social Web, the biggest trend of the Internet in this millenium so far, and probably for the rest of the millenium. Everyone knows that the wisdom of crowds is greater than the wisdom of one." On the broader theme of Kamailio's ascendancy to the mainstream of the Social Web, McCord said: "Anyway, the whole reason Kamailio hasn't gone viral so far is due to lack of network effects. We have planking, owling, Grumpy Cat, so why not Kamailio? Kamailio's imagery is green like Philosoraptor. What's stopping it from going big like Philosoraptor? No network effects, no multiplier--that is, until now." J.R. Richardson, Chief Technology Officer of Dallas-based Ntegrated Solutions, reported positive results from his initial testing: "Down here in Texas, we're a real social bunch. But I always felt so lonely writing Kamailio script by myself. I don't even know what I'm doing half the time. But now, with this Social Configuration business, the whole neighborhood can help out!" John Knight, Chief Engineer at Ringfree Communications in Hendersonville, North Carolina, also spoke highly of the new social technology: "It's great! The config just writes itself! But the best part for me, as a civically conscious businessman, is the economic boost. It's a much better employment opportunity for stay-at-home moms in Appalachia than Mechanical Turk. I'm paying out 2 cents per core function call, 4 cents per modparam, 10 cents per Record-Route parameter, and 5 cents per module function invocation. Stimulate this economy!" Carlos Alvarez, an international business consultant at The Himley Dock Table Agency, a Phoenix, Arizona-area think tank focused on sociology of the World Wide Web, was keen to note that the technology brought forth by Evariste was far more than just a different way of implementing Kamailio solutions. "We were talking with Evariste about building an XMPP server and selling it to Facebook for $20 billion, but that train left the station. It was a blessing in disguise, though, because it told us that we weren't thinking big enough. Social Configuration is way more transformative." Asked further about what it meant, Alvarez said: "This is going to have implications way beyond esoteric programming. A crowdsourced config is nothing like an individual's config; it's e-cosmopolitanism, it's Humanity 2.0. It will open up the floor to new kinds of first movers and lead to new assumptions about how scripts of all kinds should be written. It's going to challenge established notions of what it means to program, and it will make the playing field more level by disintermediating the supply chain of session initiation. A lot of middlemen are going to hurt. One does not simply program the program anymore; the program programs you, too. It programs all of us. For the first time, request routes and SRV load balancing are part of a living, breathing organism. And that's worth way more than an XMPP server. That's disruption right there." After a moment's thought, Alvarez added: "I can probably flip it to Facebook for at least $3 trillion in Zuckerberg funny money--whoops, did I say that? I meant Facebook stock." The transformative impact forecasted by Alvarez is seemingly borne out in the enthusiasm for Social Configuration shown around the world by a variety of societal constituencies not traditionally thought to be associated with software development and telecommunications. Jakub Klausa, a self-described member of an "anarcho-syndicalist reading group" in Wroclaw, Poland, confidently offered a hypothesis about the` new era brought forth by Social Configuration: "Let's start with the deeply-ingrained assumption of Kamailio consultant or trainer as 'teacher', as a purveyor of knowledge that he has and you don't," said Klausa. "That's a very oppressive, authoritarian construction. It's quite presumptuous and dehumanising. Oh, does the Kamailio 'expert', His Royal Highness, deign to 'teach' us, to 'share' what he knows with the little people? Oh, thank you, massa', thank you!" lamented Klausa. "You see, it's not so much about what the consultants and self-anointed mailing list demigods can 'teach' us. It's more about what we can all learn from each other! Social Configuration is a tool that finally allows us, the workers, the so-called 'newbies', to take the power of Kamailio into our hands and master our own destiny. It's time to put a knife through the tyranny of 'expertise' and 'knowledge'." The Chairman of the Plenary Session of the Central Committee of the Revolutionary Communist Party of Poland, Magdalena Boksa, issued the following communiqué from the party's headquarters, thought to be located in an underground bunker beneath a Starbucks: Social Configuration empowers the revolutionary vanguard of the activated, class-conscious proletariat to commandeer the machinery of Kamailio, in order to advance a scientific understanding of the objective laws of motion of the capitalist system, together with the tension of its internal contradictions, in keeping with the essential truth of dialectical materialism. The global imperialist division of labour, as revealed by V.I. Lenin in Selected Works, Vol. 1, shall meet its end through the thunderous destruction of the petit-bourgeois monopoly on knowledge-as-capital and the extraction of surplus value from it. Through the dogma of Social Configuration, the means of production in Information Technology shall be socialised among the working class, distributed according to the principle of "from each according to his ability, to each according to his need". When asked about the revolutionary potential of Social Configuration, a young student member of the RCP-Poland Youth Brigade concurred with Comrade Boksa's recitation: "¡Venceremos! ¡No pasarán! ¡Hasta la victoria siempre!" The impact of Social Configuration has even played out on a national, and perhaps global level of significance. A brief statement from the Ministry of Foreign Affairs of the Democratic People's Republic of Korea read: Social Configuration of Red Star SER [as it is known inside North Korea] is consistent with the teachings of Kim Il-Sungism and Kim Jong-Ilism. Great Leader Kim Jong-Un, Supreme Commander of the Korean People's Army, First Secretary of the Workers' Party of Korea, will chart the glorious course to the maximum realisation of its full military and civil potential in all spheres of public and private life. It will boost the resilience of our prosperous socialist nation in the face of a continual onslaught by the imperialist menace and its ROK puppet regime. Motivated by chronic energy shortages, the secretive North Korean state has been rumoured to be working on a Kamailio-based personal VoIP appliance that is--the first of its kind--powered not by conventional electricity, but by The Juche Idea. Social Configuration has also garnered attention from scholars attentive to the role of gender, race and privilege in the configuration scripting process. Moriah Trostler, a resident feminist scholar at the Phoenix, Arizona-based Scorpion Bay Institute of Gender Studies, gave some insight into the ways that Social Configuration potentially broadens the inclusiveness of the Kamailio tapestry: "If people would check their cisprivilege and open their eyes, they would see that Kamailio, and SIP itself, is full of oppressive, psychologically violent gendered constructions. For instance, there is the infamous 'loose routing vs. strict routing' dichotomy, which is clearly a dark, sadomasochistic allegory for capricious--dare I say schizophrenic?--sexual humiliation. It's a lurid invitation to indulge the prurient whims of the boss-with-secretary." Trostler gave another example: "And From and To 'tags'? Please, the idea of ownership of another's corporeal being, of branding--tagging--like cattle, is transparent. That's straight out of the Larry Flint smut-peddling playbook. And what's more, all this makes use of 'globally unique identifiers', the signature scent of industrial alienation--an anonymising, dystopian, Kafkaesque labyrinth. This has all the markers of masculinised depersonalisation. Need I go on?" Trostler went on: "Social Configuration opens the door to speaking truth to some of these discriminatory, alienating, gendered mental categories in which brogrammers think. Finally, women can write the Kamailio config as they might have written it, without compromising their identity or their dignity. Maybe we can even make some headway in influencing the male-dominated IETF standards process." Asked to summarise the impact of emergent minority voices and critical discourse enabled by Social Configuration, Fred Posner, a literary theorist based in Gainesville, Florida, offered the following comments: "The larger power of Social Configuration is essentially meta. It offers a new language, a new word-prism through which to look at the neosemiotic discursive space of SIP express routing. It's about reader as text, and text as reader, and the joy and wonder of their sizzling interplay." "It is evocative of Nietzsche," said Posner, gesturing to an open manhole cover on a busy Jacksonville thoroughfare. "If you stare too long into the Kamailio configuration, it, in turn, stares back into you! But now it speaks with a palpable collective voice, giving shape--definition!--to a bold, muscular subtext of configuration as culture, and culture as configuration. It is a semantic fusion of coder as critic, and critic as coder. Without knowing it--and it was inconceivable that they could have known it--Evariste has set into motion a wondrous, intricate and mesmerising ballet, a shower of sparks, a thrifty, yet spirited rejoinder to the supposed futility of poststructuralist rationalism. As we speak, a new 'Social Text' is being written!" Alistair Cunningham, a physicist from the University of Cambridge, provided an appraisal from a different angle: "Social Configuration, as interpreted by Balashov, embodies the kind of nonlinear thinking that is noticeably missing in software. At first glance, it does seem to invite a deafening cacophany, an undifferentiated chaos. But if one phase-shifts it slightly, the shadows of mathematical signatures emerge, waiting to be unlocked. The quantum resonance of social entropy doing its work gives us valuable clues into the origins of the universe at the sub-particle level." Asked about what Evariste has in store next, Balashov was uncharacteristically tight-lipped: "I can't say too much. But I can tell you it's big. Though he doesn't know it yet, we're even going to hire Leif Madsen for this, because the database is going to be at least 6 GB. That's almost a byte for every person on the planet. Definitely Big Data territory." Pressed for details, Balashov said: "Okay, okay, but this is all I'm going to say. I've got three words for you: IMS With Friends. It's a proprietary mash-up of Snapchat in the front, Hadoop in the back, and a DIAMETER billing interface in the middle. What could possibly go wrong?" -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

Kamailio on Android
by jay binks 31 Mar '14

31 Mar '14

I realise this is a little left field, but has anyone compiled kamailio to run on Android ? ( and rtpproxy etc if you know ) would be really handy if there was some experience here. -- Sincerely Jay

1 0

rls, rls_get_by_service_uri(): match not found
by Mihai Marin 31 Mar '14

31 Mar '14

Hello Sirs, I'm trying from some days to make RLS work but without success. I checked every character from kamailio.cfg, xml files from xcap but I can't find the problem and I'm out of ideas. My problem is the the subscribe is not done => 0 notifications with status. I think the key error is that: DEBUG: rls [subscribe.c:125]: rls_get_by_service_uri(): match not found, service-uri = [sip:bob@93.187.138.214] 2(29025) DEBUG: rls [subscribe.c:125]: rls_get_by_service_uri(): match not found, service-uri = [sip:bob@93.187.138.214] 2(29025) DEBUG: rls [subscribe.c:125]: rls_get_by_service_uri(): match not found, service-uri = [sip:bob@93.187.138.214] 2(29025) DEBUG: rls [subscribe.c:279]: rls_get_service_list(): service uri sip:alice@93.187.138.214 not found in rl document for user sip:bob@93.187.138.214 Request: SUBSCRIBE sip:bob-list@93.187.138.214 SIP/2.0 Via: SIP/2.0/WS kvvj9i9uhvbp.invalid;branch=z9hG4bK4622972 Max-Forwards: 69 To: <sip:bob-list@93.187.138.214> From: "Bob Test" <sip:bob@93.187.138.214>;tag=e9144sdk2v Call-ID: 52k3b9gheoj6gk2vmjlrr0 CSeq: 81 SUBSCRIBE Expires: 300 Event: presence Contact: <sip:v8jmlmeu@kvvj9i9uhvbp.invalid;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:152abb8b-085d-4ca5-b658-4ecb38d1b79f>";expires=60000 Supported: eventlist Accept: multipart/related, application/pidf+xml, application/rlmi+xml, application/rpid+xml Allow-Events: refer, presence, presence.winfo, xcap-diff, conference Require: recipient-list-subscribe Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER User-Agent: JsSIP 0.3.7 Content-Length: 0 Receive: NOTIFY sip:v8jmlmeu@188.215.94.132:25702;transport=ws SIP/2.0 Via: SIP/2.0/WS 93.187.138.214:5040;branch=z9hG4bKf43b.cc179644000000000000000000000000.0 To: <sip:bob@93.187.138.214>;tag=e9144sdk2v From: < sip:bob-list@93.187.138.214>;tag=67d32ce551425daaac3772fa02abe99f-daa0 CSeq: 1 NOTIFY Call-ID: 52k3b9gheoj6gk2vmjlrr0 Content-Length: 394 User-Agent: kamailio (4.2.0-dev3 (x86_64/linux)) Max-Forwards: 70 Event: presence Contact: <sip:rls@93.187.138.214> Subscription-State: active;expires=300 Require: eventlist Content-Type: multipart/related;type="application/rlmi+xml";start="<1396291389.sip:bob-list@93.187.138.214.834374984>";boundary="TnOoxHgwMlltoGjpMlgd1kNc" --TnOoxHgwMlltoGjpMlgd1kNc Content-Transfer-Encoding: binary Content-ID: <1396291389.sip:bob-list@93.187.138.214.834374984> Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list uri=" sip:bob-list@93.187.138.214" xmlns="urn:ietf:params:xml:ns:rlmi" version="1" fullState="true"> <resource uri="sip:alice@93.187.138.214"/> </list> --TnOoxHgwMlltoGjpMlgd1kNc-- I have 2 users: bob & alice. Alice xcap files: <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="Default"> <display-name>All Contacts</display-name> <entry xmlns="urn:ietf:params:xml:ns:resource-lists" uri=" sip:bob@93.187.138.214"> <display-name>bob</display-name> </entry> </list> </resource-lists> <rls-services xmlns:rl="urn:ietf:params:xml:ns:resource-lists" xmlns="urn:ietf:params:xml:ns:rls-services"> <service uri="sip:alice-list@93.187.138.214"> <resource-list> http://93.187.138.214:5060/xcap-root/resource-lists/users/sip:alice@93.187.… </resource-list> <packages> <package>presence</package> </packages> </service> </rls-services> Bob xcap files: <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="Default"> <display-name>All Contacts</display-name> <entry xmlns="urn:ietf:params:xml:ns:resource-lists" uri=" sip:alice@93.187.138.214"> <display-name>alice</display-name> </entry> </list> </resource-lists> <rls-services xmlns:rl="urn:ietf:params:xml:ns:resource-lists" xmlns="urn:ietf:params:xml:ns:rls-services"> <service uri="sip:bob@93.187.138.214"> <resource-list> http://93.187.138.214:5060/xcap-root/resource-lists/users/sip:bob@93.187.13… </resource-list> <packages> <package>presence</package> </packages> </service> </rls-services> And, global file: <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> </namespaces> </xcap-caps> KAMAILIO log is here: http://pastebin.com/4XGSCPkc Kamailio route presence from config is here: http://pastebin.com/LYSr5yPz What I'm doing wrong? Are xml configured properly? Thank you. Best regards, Mihai M

1 0

Kamailio ICSCF strange forking
by DURECU, OLIVIER (OLIVIER) 31 Mar '14

31 Mar '14

Hi all, I have deployed an IMS based on Kamailio 4.1.2 and I don't manage establishing a call. Users registration is OK but when I initiate the call from A(a)ims.net to B(a)ims.net, I notice a strange I-CSCF behavior The INVITE message goes OK from P-CSCF to S-CSCF then to I-CSCF. Till this point everything is fine... Then the I-CSCF performs its LIR diameter request and gets an Ok response from the HSS with a S-CSCF URI The I-CSCF forwards the INVITE at the received S-CSCF. BUT at the same time the I-CSCF generates a DNS request to resolve ims.net SIP SRV (it gets itself as answer...) and forwards a second INVITE (exact copy from the one sent to the S-CSCF) to itself So I have 2 copies of the same INVITE dialog travelling in the network and messing everything. Any idea of the cause of this behavior . Thanks for your support... Olivier

1 1

Kamailio and Asterisks work on HP-UX ?
by Rizwan Khan 31 Mar '14

31 Mar '14

Can they be installed on an HP UX platform. Considering they are written in C, it shouldnt be a porblem, but has anyone tried that? Thanks, Rizwan Khan

2 4

SIP - WebRTC gateway
by Zappasodi Daniele 31 Mar '14

31 Mar '14

Hello, I'm figuring out the best approach to deploy a bridge between Websocket\Webrtc and SIP\rtp. Can Kamailio (+mediaproxy-ng or something else) operate as a full Webrtc\SIP gateway (signaling, audio or video transcoding, ICE and so on)? Some months ago I found the architecture described here http://www.kamailio.org/wiki/devel/rtcweb_breaker that proposes to introduce a new RTCWeb Breaker. Is it just a proposal or is Kamailio moving following this approach? If Kamailio really requires a RTCWeb Breaker, what are the main issues against using Doubango webrtc2sip with Kamailio? Performance? Interoperability? License? ... Thanks Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

1 0

How to turn on debugging in kamailio?
by mark li 31 Mar '14

31 Mar '14

is there a way to add debug statements in kamailio.cfg? I'd like to be able to dump some of the variables and also see which path the calls are taking. thanks

5 11

ERROR reply code Failure route
by pars3c 31 Mar '14

31 Mar '14

Hello, i receive many error on the failure route like this " t_should_relay_response(): ERROR: t_should_relay_response: status rewrite by UAS: stored: 500, received: 487" . Do you know why ? My version of Kamailio is the 4.1.2 and the code is this: failure_route[MANAGE_FAILURE] { if (t_is_canceled()) { exit; } revert_uri(); if (!cr_next_domain("$avp(s:carrier)", "$avp(s:domain)", "$rU", "$avp(s:host)", "$T_reply_code", "$avp(s:domain)")) { exit; } if (!cr_route("$avp(s:carrier)", "$avp(s:domain)", "$rU", "$rU", "call_id")) { exit; } $avp(s:host)= $rd; t_on_failure("MANAGE_FAILURE"); t_on_branch("MANAGE_BRANCH"); append_branch(); if (!t_relay()) { exit; } } Thanks to all

2 2

Forward copy of Register information to FreeSWITCH
by Alexandr Usov 31 Mar '14

31 Mar '14

I am already have some practice to integrate Kamailio with Asterisk, when all users creates and registers in Kamailio, and calls go to/from Asterisk with static "host=kamailio_ip" settings for each user on Asterisk side. I can't (don't know - how to) use in same way integration with FreeSWITCH. Can't create in FS directory structure a user with "host=kamailio_ip", FS require registration. Maybe I can register user on Kamailio and send additional registration request to FS with src ip changed to kamailio (lan ip)? P.S. Reading FreeSWITCH 1.2 book in progress...

4 7

Patch to support symmetric UDP socket in mi_datagram
by Muhammad Shahzad 31 Mar '14

31 Mar '14

Hi, After wasting most of the day trying to make mi_datagram over UDP socket work. I eventually realize that it does asymmetric UDP communication, which creates a lot of trouble for writing a useful MI script using PERL or Python etc. Anyhow, i go through the module code and was able to write a patch for symmetric UDP communication using mi_datagram. I have tested the patch using both Kamailio 4.1 and trunk branches and it seems to compile and run flawlessly. I am attaching here with this patch for Kamailio community. It would be great if Kamailio developers would review and commit this patch to source trunk. Thank you.

2 5

dialog expire, acc
by Kelvin Chua 28 Mar '14

28 Mar '14

I have a situation here, when a dialog expires, it sends a BYE to both call legs and this does not generate an entry on acc table. however, when i use acc_db_request() on event_route[tm:local-request], i get 2 entries on acc, 1 for the downstream, and 1 for the upstream. is there a way of handling this so i can only get 1 BYE on acc? Kelvin Chua

2 6

Kamailio custom header stored in acc db_extra
by Ryan Brindley 28 Mar '14

28 Mar '14

Hey community, I'm trying to get a 2nd leg custom header value stored in my acc table. Right now I'm appending the header with append_hf in a failure route and trying to use: modparam("acc","db_extra","custom=$hdr(Custom)") A sip trace shows the header was properly added and xlog output shows the proper $hdr(Custom) value just after I append the header, but the value doesn't show in acc. Does acc only read from the INVITE of the incoming leg? Any suggestions on how to get this 2nd leg INVITE header in acc is greatly appreciated. Thanks in advance, Ryan

2 5

Trying to dial 4000 into freeswitch's vmail system - Kam Proxy returning Status 403: Not allowed
by mark li 28 Mar '14

28 Mar '14

Hi there. I'm still trying to integrate Kamailio and freeswitch... where kamailio acts as a proxy and registrar ... and freeswitch provides conference calls and voicemail. I have calls between two polycoms working and conference calls work. But when I try to leave a voice message for a user by dialing 44+ext, the sip proxy never replies to the polycom. I did a tcpdump and i can see the INVITE from the polycom to the sip proxy multiple times... but no response back. The phone eventually disconnects itself. Here's what my config looks like: http://pastebin.com/wWgyVcxc Just do a search for "route[FSDISPATCH]". You will see how I check for the "44" prefix, and then send the call to a route called "FSVBOX". Any suggestions would be appreciated. 1. route[FSVBOX] { 2. if(!($rU=~"^1[0-9][0-9]+$")) 3. return; 4. prefix("vb-"); 5. route(FSRELAY); 6. } 7. 8. # Send to FreeSWITCH 9. route[FSRELAY] { 10. $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" 11. + $sel(cfg_get.freeswitch.bindport); 12. route(RELAY); 13. exit; 14. } 15.

1 1

rport
by Marc Soda 28 Mar '14

28 Mar '14

I have a Kamailio server forwarding REGISTERs to an Asterisk box, similar to the way Daniel shows here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb Can I force Kamailio to append rport=5060 to the topmost Via header, prior to forwarding the REGISTER to Asterisk? I tried add_local_rport() but it only included 'rport' and not 5060. Thanks, Marc

2 3

SIP Connection questions
by GRIBANOV, Oleg 28 Mar '14

28 Mar '14

Dear sirs, I would like to make some tests of the SIP connection using your server. Could you please confirm me the following details: - IP of the server? Is it 212.79.11.155 ? - How could I register and use the STUN server ? What is the address of STUN server ? Thank you, Oleg Gribanov ________________________________ Important Notice: THIS EMAIL (INCLUDING ANY ATTACHMENTS) IS CONFIDENTIAL AND MAY BE PRIVILEGED. IF YOU ARE NOT THE INTENDED RECIPIENT, PLEASE CONTACT THE SENDER, AND REFRAIN FROM COPYING, FORWARDING, DISCLOSING, DISTRIBUTING OR OTHERWISE USING THIS EMAIL OR ANY PART OF IT. DOING OTHERWISE MAY CONSTITUTE AN OFFENCE OR CONTRACTUAL BREACH. ALL DISCUSSIONS AND DRAFTS CONTAINED IN OR ATTACHED TO THIS EMAIL ARE WITHOUT COMMITMENT AND ARE SUBJECT TO CONTRACT, UNLESS EXPRESSLY STATED OTHERWISE. CET EMAIL (AINSI QUE TOUTES SES PIECES JOINTES) EST CONFIDENTIEL ET PEUT ETRE COUVERT PAR LE SECRET PROFESSIONNEL. SI VOUS N'ETES PAS LE DESTINATAIRE D'ORIGINE, VEUILLEZ CONTACTER L'EXPEDITEUR, ET ABSTENEZ-VOUS DE LE COPIER, TRANSFERER, REVELER, DIFFUSER OU FAIRE UN QUELCONQUE USAGE DE CET EMAIL, EN TOUT OU PARTIE. ENTREPRENDRE UNE TELLE ACTION PEUT CONSTITUER UN DELIT OU UNE FAUTE CONTRACTUELLE. TOUTES DISCUSSIONS ET PROJETS QUI SONT INCLUS OU JOINTS A CET EMAIL SONT SANS ENGAGEMENT DES SOCIETES DU GROUPE EUROSPORT ET DOIVENT FAIRE L'OBJET D'UN CONTRAT, SAUF ACCORD EXPRES CONTRAIRE. ________________________________

1 0

Rx interface
by Dees 27 Mar '14

27 Mar '14

Does Kamailio support full Rx interface?

1 0

Integrating Kamailio and Freeswitch
by mark li 27 Mar '14

27 Mar '14

Hi there. I'm a noobie to Kamailio and Freeswitch... but I'm trying to follow the article located here: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc I've tried to add all the sections marked with WITH_FREESWITCH in the sample config in the article into my own kamailio-advanced.cfg file. Here's what my cfg file looks like: http://pastebin.com/KsvrYVN7 I've restarted kamailio after making these changes. Then I tried to dial 41 to listen to vmail or 44999 to leave a message for user 999 but both return a busy tone. Any suggestions would be appreciated. Thanks.

1 2

Need to confirm the changes for the Kamailio.cfg
by varun pratapsingh 27 Mar '14

27 Mar '14

HI All, This is my first post to this community and I am new user of the Kmailio. I want to use the Kamailio and RTPProxy on the same machine. I have installed and rtpproxy succesfully and I have started both the kamailio and RtpProxy. My RtpProxy is listening on localhost:7722 port of my local machine. Now this machine is on the public IP and I want that all sip and RTP passes though this machine from SIP Softphones before going to the SIp Server. This means the machine on which I am running the Kamailio and RTPProxy will act like a SIP proxy from my SIP Softphones through which the SIP and RTP relays. SIP is relaying fine with defult kamailio.cfg but can anyone let me know the changes in Kamailio.cfg so that my RTP also passes or rleays through this Kamailio + RTPProxy machine. Thanks in advance. Thanks and Regards Varun

2 1

Preventing information about my sip network
by Arya Farzan 27 Mar '14

27 Mar '14

I'm concerned about others reverse engineering their way into my project's sip network. Is there anyway to prevent others from finding out that the SIP protocol is being used and prevent others to reverse engineer their way into my sip network?

3 6

Rewrite SIP INVITE
by Mickael MONSIEUR 26 Mar '14

26 Mar '14

Hello, I installed kamailio and asterisk with the tutorial of asipto. For alias numbers I configured the module alias_db. Everything works because Asterisk outgoing call is directed at Kamailio, then Kamailio is sending to the alias registered (table location) Only the number alias appears in the "To" (02XXXXXX) and not in the INVITE URI. It only shows "s" ... is it possible to force writing 02XXXXXX instead of "s" ? Example: INVITE sip:s@10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: "0475XXXXXX" <sip:1053212@sip.domain.com>;tag=as7df9ab18 To: <sip:02XXXXXX@kamailioIP:5060> Contact: <sip:1053212@asteriskIP:5060> Call-ID: 344d42bd16975a54141d11f635bdfc71(a)sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 252 Because without it, Asterisk servers behind Kamailio not will route the call to the correct extension but to the "s". Asterisk ignores the "To" apparently this is strange ... Thank you, Mickael

2 3

mediaproxy-ng, invalid header version
by Mihai Marin 26 Mar '14

26 Mar '14

Hellor Sirs, Sir Richard, I saw some updates in the last 2 weeks that were working with Firefox - I also did some tests as it was working. Now, I tried to get the latest version and I'm getting the following error: mediaproxy-ng[2747]: Got valid command from 127.0.0.1:35127: answer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-29.0a2 13431 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:8a9a3649#015#012a=ice-pwd:80b3b796ecdd0addb079fc531c7e0afa#015#012a=fingerprint:sha-256 A3:D7:1F:F6:60:1D:91:27:64:89:2E:09:CE:42:19:DB:7E:5F:02:06:A3:DA:E0:26:0F:F7:30:4C:1E:65:86:2B#015#012m=audio 50990 RTP/SAVPF 109 101#015#012c=IN IP4 188.215.94.132#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=recvonly#015#012a=setup:active#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50990 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50990 typ srflx raddr 192.168.0.5 rport 50990#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50991 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50991 typ srflx raddr 192.168.0.5 rport 50991#015#012a=rtcp-mux#015#012m=video 50992 RTP/SAVPF 120#015#012c=IN IP4 188.215.94.132#015#012a=rtpmap:120 VP8/90000#015#012a=sendrecv#015#012a=rtcp-fb:120 nack#015#012a=rtcp-fb:120 nack pli#015#012a=rtcp-fb:120 ccm fir#015#012a=setup:active#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50992 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50992 typ srflx raddr 192.168.0.5 rport 50992#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50993 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50993 typ srflx raddr 192.168.0.5 rport 50993#015#012a=rtcp-mux#015#012", "flags": [ "force", "trust-address" ], "replace": [ "origin", "session-connection" ], "call-id": "ufdo7gas8lt6jpco60b1", "received-from": [ "IP4", "188.215.94.132" ], "from-tag": "nj7tuhg5hj", "to-tag": "7lpqrsa1eb", "command": "answer" } mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1] Returning to SIP proxy: d3:sdp1463:v=0#015#012o=Mozilla-SIPUA-29.0a2 13431 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:8a9a3649#015#012a=ice-pwd:80b3b796ecdd0addb079fc531c7e0afa#015#012m=audio 30002 RTP/SAVPF 109 101#015#012c=IN IP4 93.187.138.212#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50990 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50990 typ srflx raddr 192.168.0.5 rport 50990#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50991 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50991 typ srflx raddr 192.168.0.5 rport 50991#015#012a=recvonly#015#012a=rtcp:30002#015#012a=rtcp-mux#015#012a=setup:active#015#012a=fingerprint:sha-1 CA:7D:5B:F1:C8:1A:05:8E:9B:88:0F:36:9B:C4:7A:60:A3:B5:02:A4#015#012a=candidate:kgDINMdmuvprUMd6 1 UDP 2130706432 93.187.138.212 30002 typ host#015#012m=video 30006 RTP/SAVPF 120#015#012c=IN IP4 93.187.138.212#015#012a=rtpmap:120 VP8/90000#015#012a=rtcp-fb:120 nack#015#012a=rtcp-fb:120 nack pli#015#012a=rtcp-fb:120 ccm fir#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50992 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50992 typ srflx raddr 192.168.0.5 rport 50992#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50993 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50993 typ srflx raddr 192.168.0.5 rport 50993#015#012a=sendrecv#015#012a=rtcp:30006#015#012a=rtcp-mux#015#012a=setup:active#015#012a=fingerprint:sha-1 CA:7D:5B:F1:C8:1A:05:8E:9B:88:0F:36:9B:C4:7A:60:A3:B5:02:A4#015#012a=candidate:kgDINMdmuvprUMd6 1 UDP 2130706432 93.187.138.212 30006 typ host#015#0126:result2:oke mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30000] Error parsing RTP header: invalid header version mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30004] Error parsing RTP header: invalid header version mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30002] Error parsing RTP header: invalid header version mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30006] Error parsing RTP header: invalid header version I was dreaming one week ago when I saw Firefox working with chrome :)? Do you know something about this error? Thank you. Best regards, Mihai M

2 3

Using Kamailio with a SIP Trunk
by Rizwan Khan 26 Mar '14

26 Mar '14

I want the following setup: 1 Kamailio server to handle internal calls (A/V), IM and Presence. 1 Asterisk or any other way to communicate with an NGN where I will create the SIP Trunk to route calls outside of the network. Is this the right approach or there is a way to directly communicate with the NGN to make a SIP trunk by using some external modules. Any guidelines will be highly appreciated. Rizwan Khan

3 4

Memory leak when using db_unixodbc, help tracking it down (try 2)
by Alex Villacís Lasso 26 Mar '14

26 Mar '14

I have a CentOS 6 installation with the following packages installed from the RPM build service from Kamailio: kamailio-unixodbc-4.1.2-1.1.x86_64 kamailio-4.1.2-1.1.x86_64 kamailio-presence-4.1.2-1.1.x86_64 kamailio-utils-4.1.2-1.1.x86_64 I am also using db_unixodbc for all database accesses (with MySQL backend). The problem is that after a few hours, a single Kamailio process (never more than one) starts complaining that it has run out of memory: 9(11479) ERROR: uac [uac_reg.c:638]: uac_reg_tm_callback(): got sip response 408 while registering [admin_pbx.elastix.com] 9(11479) ERROR: db_unixodbc [dbase.c:335]: db_unixodbc_fetch_result(): no memory left 9(11479) ERROR: <core> [db_query.c:502]: db_fetch_next(): unable to fetch next rows 9(11479) ERROR: db_unixodbc [dbase.c:224]: db_unixodbc_free_result(): invalid parameter value 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result I have attached the stderr output when I kill the offending process. From what I can see, the memory leak somehow involves database allocations that fail to be freed, consuming more than 3 MB out of 4 MB allocated: 9(11479) NOTICE: <core> [main.c:857]: sig_usr(): Memory still-in-use summary (pkg): 9(11479) NOTICE: qm_sums: summarizing all alloc'ed. fragments: 9(11479) NOTICE: qm_sums: count= 1 size= 10240 bytes from mi_fifo: mi_writer.c: mi_writer_init(57) 9(11479) NOTICE: qm_sums: count= 378 size= 3029384 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(333) 9(11479) NOTICE: qm_sums: count= 1 size= 4096 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(325) 9(11479) NOTICE: qm_sums: count= 1 size= 16 bytes from <core>: sr_module.c: init_modules(1002) 9(11479) NOTICE: qm_sums: count= 1 size= 56 bytes from textops: textops.c: hname_fixup(1634) 9(11479) NOTICE: qm_sums: count= 1 size= 32 bytes from rr: rr_cb.c: register_rrcb(63) I have the setup to compile my own RPMS, and I intend to track down the memory leak on my own. However, I would like your help in focusing my search, on whether this is a known issue, or where in db_unixodbc should I look.

2 3

Regarding SRV lookup
by Swetha Raj 26 Mar '14

26 Mar '14

Hi All, I am trying to communicate between two sip servers.The communication from external sip server user to kamailio user works but message from kamailio to external sip user is not reaching. I get the following error in Jitsi "The above message could not be delivered A network problem occurred. Please check your network configuration and try again. Error was: 408 Request Timeout" Does kamailio do SRV lookup ? How do i force kamailio to use DNS SRV lookup. Any clues ??

1 0

group regex db reload?
by Pete Ashdown 26 Mar '14

26 Mar '14

I'm using the group module to assign groups to users. As the documentation states, "the table content is loaded into memory at startup and all regular expressions are compiled". Having to restart Kamailio is cumbersome every time the table is updated. Is there a method to reload the table via kamcmd? Also, I had no luck in getting the non-regex method of group to recognize a table entry, for example, 159(a)mydomain.com was split into 159 and mydomain.com, yet it get_user_group wouldn't match it. It was only when I started using the regex table that I had success.

2 1

Qualify from Asterisk to Kamailio's peers
by Alexandr Usov 25 Mar '14

25 Mar '14

Peers (from Internet behind NAT) registered on Kamailio (local ip 192.168.182.1), calls from/to routed via Asterisk (192.168.182.24). Can't use qualify info: <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog ' 1efe8f023a80bfb343495d5c4f20ea35@192.168.182.24:5060' Method: OPTIONS <--- SIP read from UDP:192.168.182.1:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.182.24:5060;branch=z9hG4bK4a817c26;rport=5060 From: "asterisk" <sip:asterisk@192.168.182.24>;tag=as34bfcba7 To: <sip:192.168.182.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2d0e Call-ID: 753402f55d9b5ceb6dc402b93e206025@192.168.182.24:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 <------------->

5 7

Unable to make calls after integration with Freeswitch
by mark li 25 Mar '14

25 Mar '14

Hi there. I had a working Kamailio proxy working where i could call from one polycom test phone to another. But then I tried to integrate freeswitch with it... (following the article found at: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc as a baseline. ) Now when I dial a phone, it's just silent for a few seconds... after which i get a busy tone. I did a tcpdump and have been reviewing the file in wireshark. It looks like my two test phones are registered properly... but as you can see, the INVITE is not getting an OK back. It just keeps on retrying the invite. I'm not sure how to go about troubleshooting this. Any suggestions on where to start would be appreciated. In the meantime, I'm googling around to see if I can find anything that might help me figure this out. thanks.

1 0

dialog is not deleted when BYE received
by Tuan Viet Nguyen 25 Mar '14

25 Mar '14

Hello, I disable the rr module and only use SIP elements for dialog match modparam("dialog", "dlg_match_mode", 2) Upon receipt of a BYE request, the request is correctly forwarded and the call ended but the dialog still exists with ref_count = 2 kamctl fifo dlg_list dialog:: hash=731:5167 state:: 3 ref_count:: 2 timestart:: 1394625569 timeout:: 29082806 callid:: 77dd7249530067080de983411e94569c@10.100.8.12:5060 from_uri:: sip:10.100.8.12:5060 from_tag:: as03e52e3f caller_contact:: sip:10.100.8.12:5060 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.100.8.94:5060 callee_bind_addr:: udp:10.100.8.94:5060 Do you have any idea ? Should I remove manually the dialog using a module function? Thanks for your help Best regards,

1 1

Siremis v4.1.0 Released
by Elena-Ramona Modroiu 25 Mar '14

25 Mar '14

Hi, a new release of Siremis is out - v4.1.0 - web management interface that is compatible out of the box with Kamailio v4.1.x series. More details, including link to install tutorial, are available at: - http://siremis.asipto.com/2014/03/25/siremis-v4-1-0-released/ Regards, Ramona

1 0

crash dialog module
by pars3c 25 Mar '14

25 Mar '14

Hi, i have installed the new version ok kamailio 4.1.2. I don't know why Kamailio crash writing this ob the log: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7fc3ab755eb0), called from dialog: dlg_hash.c: dlg_update_cseq(591), first free dialog: dlg_hash.c: dlg_update_cseq(591) - aborting Anybody can help me to undertand what is the problem? thanks

3 2

Ldap auth
by Slava Bendersky 24 Mar '14

24 Mar '14

Hello Everyone, I am trying compare domain part of uri with ldap query result, getting some syntax warning 1. arn_at(): warning in config file /etc/kamailio/kamailio-ldap.cfg, line 992, column 17-39: constant value in if(...) 2. 3. 4. ldap_result("sipExtension/$avp(extension)"); 5. ldap_result("sipDomain/$avp(domain)"); 6. ldap_result("password/$avp(password)"); 7. 8. } 9. 10. if ("$avp(s:domain)" =~ "$fd") { 11. xlog("L_INFO", "Not alllowed $fd"); 12. sl_send_reply("403","Not allowed $fd"); 13. exit; 14. } 15. any help thank you

2 2

presence bug
by Kelvin Chua 24 Mar '14

24 Mar '14

I would like to report a bug on the presence module kamailio 4.0.5 setting custom table names via modparam("presence", "presentity_table", "kam_presentity") modparam("presence", "active_watchers_table", "kam_active_watchers") modparam("presence", "watchers_table", "kam_watchers") does work, i can see entries on the table when logged in. however these things show up from time to time, seems like it is not looking for the custom table name. /usr/local/sbin/kamailio[20406]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Table 'kam.presentity' doesn't exist /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:429]: db_fetch_query_internal(): unable to query db for fetch /usr/local/sbin/kamailio[20406]: ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages /usr/local/sbin/kamailio[20406]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Table 'kam.presentity' doesn't exist /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:291]: db_do_delete(): error while submitting query /usr/local/sbin/kamailio[20406]: ERROR: presence [publish.c:227]: msg_presentity_clean(): failed to delete expired records from DB /usr/local/sbin/kamailio[20406]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Table 'kam.watchers' doesn't exist /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:291]: db_do_delete(): error while submitting query /usr/local/sbin/kamailio[20406]: ERROR: presence [subscribe.c:745]: msg_watchers_clean(): cleaning pending subscriptions /usr/local/sbin/kamailio[20406]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Table 'kam.active_watchers' doesn't exist /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:291]: db_do_delete(): error while submitting query /usr/local/sbin/kamailio[20406]: ERROR: presence [subscribe.c:2220]: update_db_subs_timer(): deleting expired information from database Kelvin Chua

1 0

Memory issue
by Igor Potjevlesch 24 Mar '14

24 Mar '14

Hello, After couple of weeks without issue, my Kamailio 4.0 has begun to reject calls this week-end. The first error : ERROR: <core> [msg_translator.c:2012]: build_res_buf_from_sip_res(): ERROR: build_res_buf_from_sip_res: out of mem. OR ERROR: <core> [msg_translator.c:2012]: build_res_buf_from_sip_res(): ERROR: build_res_buf_from_sip_res: out of mem Followed by ERROR: <core> [forward.c:812]: do_forward_reply(): ERROR: forward_reply: building failed And sometimes, at the end of the line there is a "needs" in addition: ERROR: <core> [msg_translator.c:2164]: build_res_buf_from_sip_req(): ERROR: build_res_buf_from_sip_req: out of memory ; needs 469. The starting options are: OPTIONS="-P /var/run/$prog.pid -m 256". Regards, Igor.

2 5

Re: [SR-Users] Registration error setting pvar
by Nick Raper 24 Mar '14

24 Mar '14

It seems this happens when we try to set the auth_count back to 0 after a successful registration is the following line: $sht(a=>$au::auth_count) = 0; Nick Raper Voice Engineer T +27 (0) 21 710 8400 C +27 (0) 82 777 0855 hymax.co.za <http://www.hymax.co.za/> <http://www.hymax.co.za/> cid:image001.jpg@01CF3399.D9DDFA50 cid:image006.png@01CF339B.A5608F70 This e-mail and any attached files are intended for the named addressee only. It contains information which may be confidential and legally privileged and also protected by copyright. Unless you are the named addressee (or authorized to receive for the addressee) you may not copy or use it, or disclose it to anyone else. If you received it in error please notify the sender immediately and then delete it from your system. We make every effort to keep our network free from viruses. However you do need to check this e-mail and any attachments to it for viruses, as we can take no responsibility for any computer virus which may be transferred by way of this e-mail. From: Nick Raper [mailto:nick.raper@hymax.co.za] Sent: 24 March 2014 11:57 AM To: 'sr-users-bounces(a)lists.sip-router.org' Subject: Registration error setting pvar Hi, Can anyone assist with troubleshooting this error? It happens on client REGISTER requestes. The registration is successful but we get the following errors: 10(2994) ERROR: <core> [lvalue.c:363]: lval_pvar_assign(): setting pvar failed 10(2994) ERROR: <core> [lvalue.c:416]: lval_assign(): assignment failed at pos: (1075,31-1075,31) version: kamailio 4.1.2 (i386/linux) 6fa599 I am using a modified script and DB from version 3.2.1 as we are looking at upgrading our current live servers to ver 4.1.2. I have attached a full debug log of the register. Regards, Nick Raper Voice Engineer T +27 (0) 21 710 8400 C +27 (0) 82 777 0855 hymax.co.za <http://www.hymax.co.za/> <http://www.hymax.co.za/> cid:image001.jpg@01CF3399.D9DDFA50 cid:image006.png@01CF339B.A5608F70 This e-mail and any attached files are intended for the named addressee only. It contains information which may be confidential and legally privileged and also protected by copyright. Unless you are the named addressee (or authorized to receive for the addressee) you may not copy or use it, or disclose it to anyone else. If you received it in error please notify the sender immediately and then delete it from your system. We make every effort to keep our network free from viruses. However you do need to check this e-mail and any attachments to it for viruses, as we can take no responsibility for any computer virus which may be transferred by way of this e-mail.

1 0

kamailio.org website down?
by Isaac McDonald 24 Mar '14

24 Mar '14

Hello, It looks like kamailio.org is having issues. When I try and access it I get a perpetual "waiting for reply". -- Isaac McDonald SusCall, LLC E-Mail: imcdona(a)suscall.com

4 4

Caller ID name
by malik sherif 22 Mar '14

22 Mar '14

Hello, I set SIP users using kamctl add command with username ( I put the phone numbers) doamin (doamin name) and password. Is their a way to add caller id name? when I make a call , i see the phone number but for caller id name I displayline1. Thank you for your help. Abdul

3 4

ldap auth
by Slava Bendersky 22 Mar '14

22 Mar '14

Hello Everyone, Trying setup ldap authentication. Always getting 401. I tested and all avp getting value also domain part check working, look like challenge www in question. Any help thank you in advance. #!ifdef WITH_LDAP route[LDAP] { if(is_method("REGISTER")) { if(!(is_present_hf("Authorization") || is_present_hf("Proxy-Authorization"))) { # no credentials header - send back challenge auth_challenge("$fd", "1"); exit; } # ldap search ldap_search("ldap://sipaccounts/ou=People,dc=networklab,dc=loc?sipDomain,sipMobileExtension,userPassword?one?(&(objectClass=phonesipuser)(sipMobileExtension=$fU))"); $var(rc) = $rc; if ($var(rc)<0) { switch ($var(rc)) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); exit; case -2: # internal error sl_send_reply("500", "Internal server error"); exit; default: sl_send_reply("403", "Not allowed"); exit; } } ldap_result("sipDomain/$avp(domain)"); ldap_result("sipMobileExtension/$avp(s:username)"); if (!ldap_result("userPassword/$avp(password)")) { sl_send_reply("404", "User Not Found"); exit; } if ($fd != $avp(domain)) { xlog("L_INFO", "Got ldap result $avp(domain). For user $avp(s:username) Not allowed $fd"); sl_send_reply("403","Not allowed $fd"); exit; } if (!pv_auth_check("$fd", "$avp(password)", "0", "1")) { auth_challenge("$fd", "1"); exit; sl_send_reply("200", "ok"); exit; } else { sl_send_reply("403","Not allowed"); exit; } if (!is_method("REGISTER|PUBLISH")) { consume_credentials(); } } return; } #!endif

1 0

sqlops mysql URL
by Kelvin Chua 21 Mar '14

21 Mar '14

anybody have idea on how to dynamically access mysql URLs using avp? reason being is, i have the mysql URLs in an htable. how can i avoid using modparam for it? is it doable? Kelvin Chua

1 0

Re: [SR-Users] Module TOPOH: Does not unmask CALL-ID on request BYE in one direction
by Samuel F. 21 Mar '14

21 Mar '14

Hey, No, that is totally correct. What is strange is that it doesn’t trigger on the BYE coming back where direction is (should be) = 1. It never unmasks it when it goes back upstream. I will run a scenario in our setup again on Monday and send you the output. Have a nice weekend! // Samuel

1 0

Memory leak when using db_unixodbc, help tracking it down
by Alex Villacís Lasso 21 Mar '14

21 Mar '14

I have a CentOS 6 installation with the following packages installed from the RPM build service from Kamailio: kamailio-unixodbc-4.1.2-1.1.x86_64 kamailio-4.1.2-1.1.x86_64 kamailio-presence-4.1.2-1.1.x86_64 kamailio-utils-4.1.2-1.1.x86_64 I am also using db_unixodbc for all database accesses (with MySQL backend). The problem is that after a few hours, a single Kamailio process (never more than one) starts complaining that it has run out of memory: 9(11479) ERROR: uac [uac_reg.c:638]: uac_reg_tm_callback(): got sip response 408 while registering [admin_pbx.elastix.com] 9(11479) ERROR: db_unixodbc [dbase.c:335]: db_unixodbc_fetch_result(): no memory left 9(11479) ERROR: <core> [db_query.c:502]: db_fetch_next(): unable to fetch next rows 9(11479) ERROR: db_unixodbc [dbase.c:224]: db_unixodbc_free_result(): invalid parameter value 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result I have attached the stderr output when I kill the offending process. From what I can see, the memory leak somehow involves database allocations that fail to be freed, consuming more than 3 MB out of 4 MB allocated: 9(11479) NOTICE: <core> [main.c:857]: sig_usr(): Memory still-in-use summary (pkg): 9(11479) NOTICE: qm_sums: summarizing all alloc'ed. fragments: 9(11479) NOTICE: qm_sums: count= 1 size= 10240 bytes from mi_fifo: mi_writer.c: mi_writer_init(57) 9(11479) NOTICE: qm_sums: count= 378 size= 3029384 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(333) 9(11479) NOTICE: qm_sums: count= 1 size= 4096 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(325) 9(11479) NOTICE: qm_sums: count= 1 size= 16 bytes from <core>: sr_module.c: init_modules(1002) 9(11479) NOTICE: qm_sums: count= 1 size= 56 bytes from textops: textops.c: hname_fixup(1634) 9(11479) NOTICE: qm_sums: count= 1 size= 32 bytes from rr: rr_cb.c: register_rrcb(63) I have the setup to compile my own RPMS, and I intend to track down the memory leak on my own. However, I would like your help in focusing my search, on whether this is a known issue, or where in db_unixodbc should I look.

1 0

[Sr-users]siptrace duplicate_uri
by Tuan Viet Nguyen 21 Mar '14

21 Mar '14

Hello, I set duplicate_uri to store sip messages in an external server #!ifdef WITH_SIPTRACE modparam("siptrace", "db_url", mysql_conn_data) modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on",1) # disable siptrace modparam("siptrace", "trace_flag",FLT_SIPTRACE) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "duplicate_uri", "sip:siptracesrv:6060") #!endif In the route section setflag(FLT_SIPTRACE); but most of the time I get those error messages in log files when kamailio tries to send the message out /usr/local/sbin/kamailio[11946]: ERROR: <core> [forward.c:220]: get_out_socket(): ERROR: get_out_socket: no socket found /usr/local/sbin/kamailio[11946]: ERROR: siptrace [siptrace.c:1487]: trace_send_duplicate(): can't forward to af 2, proto 1 no corresponding listening socket Do you have any ideas? Thanks for your help,

1 0

Contract support engineer in Americas wanted
by Alistair Cunningham 21 Mar '14

21 Mar '14

Integrics would like to hire a contract support engineer to provide level two support to Enswitch customers. This is a contract, work at home role. It entails: * 10 to 20 hours per week (with the possibility of more in future) answering support questions. This is primarily via web ticketing system, though occasional calls with customers may be required. Questions are submitted by customers' engineers, so are generally highly technical. Many involve substantial investigation, debugging, reading of SIP traces, etc. You will not be dealing with end users or the general public. * Being on call for emergency support up to around 50% of the time. This will normally be daytime in North America as we have staff in other countries to cover night, but may occasionally be night time if necessary. You will be compensated for any callouts. * Optionally, some Perl development. If you're not interested in this, or lack the necessary skills, that's no problem. The following are hard requirements. Please do not reply unless you fulfil all of these: * Strong skills in Linux (Redhat and Debian based distributions), Asterisk, and SIP. * Moderate or better skills in Perl, MySQL, and networking. * At least a basic familiarity with Kamailio and/or OpenSIPS. * Experience of providing technical support to IT professionals. * Strong communication and diplomacy skills. * Be physically located in the Americas. Most of the customers you will be supporting are in the USA and Canada, and expect support during business hours. Please do not reply if located outside the Americas. * Be able to speak and write English fluently, and with a professional demeanour, spelling, and grammar. If you're interested in this role and meet the above requirements, please email me off-list. Please include details of your skills and experience, and a rough indication of hourly rates and availability. -- Alistair Cunningham +1 888 468 3111 +44 20 799 39 799 http://integrics.com/

1 0

extend subscriber profile
by Mihai Marin 21 Mar '14

21 Mar '14

Hello Sirs, How can I extend subscriber profile with some information like avatar, name, surname, other custom attributes in order to consume those information from a WebRTC app? Is there any standard way or we should use custom code for that? Thank you. Best regards, Mihai M

3 5

problem with geoip.
by arun Jayaprakash 21 Mar '14

21 Mar '14

Hello, I am trying to set up my Kamailio config file with geoip but I keep getting the following error: ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <geoip> found I have set my modparam as follows: modparam("geoip", "path", "/usr/share/GeoIP/GeoLiteCity.dat") I am using Ubuntu 12.04 and have loaded the geoip libraries and the database. Is there something I am missing? Thank you, Arun

3 4

ERROR cfgtrace in kamailio.cfg
by Cock Ootec 20 Mar '14

20 Mar '14

I am trying to setup Kamailio server with topoh module (I want to extend topoh module and develop my own module) but when I start kamailio, the errors in attached error.log appear. I attach also my kamailio.cfg. I build Kamailio from source code by this guide: http://www.kamailio.org/wiki/install/4.0.x/git Could someone please help me to resolve these errors? Thanks in advance

2 2

Hash table.
by Slava Bendersky 20 Mar '14

20 Mar '14

Hello Everyone, I am trying set hash table for geoip and put countries code which is not a customers. Currently I have working as static for single country code. #!ifdef WITH_GEOIP route[GEOIP] geoip_match("$si", "src"); if($gip(src=>cc)=="TH") { xlog("SIP message not from allowed customer (ip: $si) - drop it\n"); send_reply("403", "Forbidden"); exit; } #!endif I though replace single value on restricted countries with modparam("htable", "geoip", "country_code=>TH;") modparam("htable", "geoip", "country_code=>IN;") if($gip(src=>cc)=="$sht(geoip=>country_code)"); but not sure if it good approach or which way to go. Any help thank you. Slava.

2 2

Kamailio + TLS
by Adrian 20 Mar '14

20 Mar '14

Hi. I have configured Kamailio with Websockets. I want to add TLS. I generated certifikates with openssl and configured tls.cfg. But when i try to run kamailio, it gives me error, failed to start. This my syslog <http://paste2.org/KdYsGE9f> . Can anybody help me please? Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-TLS-tp126046.html Sent from the Users mailing list archive at Nabble.com.

3 6

Interdomain Routing
by Owais ul Haq 20 Mar '14

20 Mar '14

Hello, I have kamailio implemdented on two machines in my local network. Both are using the same database implemented on a third machine in the same network. As my servers are on local domain so given fake domain names. How can calls be routed between the two domains ?? I asked this question and got reply to use "t_relay ". I read about it but it is for stateful processing of SIP messages. I cant relate my problem to this t_relay module. What can be the solution for this ? Regards.

2 1

Child process exited by a signal 11
by Allen Zhang 19 Mar '14

19 Mar '14

Hi, Scenario: Our kamailio server normally has a debug level of 2. The server gets a segmentation fault and dies when we run a certain demo. The error message is: "[4097]: ALERT: <core> [main.c:789]: child process 4098 exited by a signal 11" And we got a lot of these before it dies: "error reading: Connection reset by peer (104)" "ERROR: tcp_read_req: error reading" Because of the low debug level, I have no idea of what child process 4098 was doing before it died. Here is the wired bit: The problem goes away if I set debug level to 5. But always occurs at debug level 2. Two questions: 1, I noticed that the main thread was 4097 and 4098 died. what is the child process created straight after main? My guess is this 4098 child process manages TCP connections. Is this correct? 2, Why does debugging level has an impact on this? Is it because higher debugging level introduces some delay? Regards, Allen

2 6

Legacy Cisco 7945 and 7960 Handset Integration with Kamailio
by Jonathan Hunter 19 Mar '14

19 Mar '14

Hi Guys, Hope your all well? Apologies for the post, however I wondered if anyone had successfully integrated Cisco 7945/7960 handsets running SIP firmware with Kamailio? I appreciate these handsets have some issues, whereby they expect responses on port 5060, even though they initiate transactions from a dynamic port, I just wondered if anyone had a working model or some pointers they could share? As we are pondering a VPN connection to remove NAT, however Id like to consider other alternatives before then. Many thanks Jon

2 3

Module not up to date in apt-get repository
by Oliver Roth 19 Mar '14

19 Mar '14

Hi all We updated from kamailio 3.1 to kamailio 3.3. As shown in module description the function rtpproxy_manage() should be available starting version 3.2 But if the update is done to 3.3 - we get the following error when trying to start kamailio * Not starting Kamailio SIP server: invalid configuration file! * * 0(24558) ERROR: <core> [cfg.y:3455]: cfg. parser: failed to find command rtpproxy_manage 0(24558) : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 905, column 17: unknown command, missing loadmodule? In request_route we have the command : Rtpproxy_manage() ; Any idea ? Kind regards, Oli

2 1

Installing Kamailio on Ubuntu
by Swetha Raj 19 Mar '14

19 Mar '14

Hi, I am trying to install Kamailio on Ubuntu 12.04,but hitting the below issue. root@test:/usr/local# apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package kamailio E: Unable to locate package kamailio-mysql-modules E: Unable to locate package kamailio-tls-modules Can anybody point me the right steps to do the same Thanks

3 3

Debian Kamailio Shell Script Installer
by Joel White 19 Mar '14

19 Mar '14

*https://gist.github.com/zhacker98/9643861 <https://gist.github.com/zhacker98/9643861>*

1 0

Does tls/wss actually work or What is required for tls/wss
by jaflong jaflong 19 Mar '14

19 Mar '14

Hi, What are the requirements for connecting with tls/wss. I have not come across any information or example for this. My config is working when the client uses ws. However if I change this to use wss, (this is it only paramter I change) it does not work. I understand Kamailio does not support DTLS, I set the jssip client DtlsSrtpKeyAgreement to false to disable this, I also set the tls option to not require or verify certicficates and it still does not work. What if other considaerstion do I need to check? thanks

6 8

db logging for unsuccessful calls
by betul.cerit＠netgsm.com.tr 19 Mar '14

19 Mar '14

Hi to all, I could hardly make kamailio log cdr to db but it just logs successful calls.I need cdr of unsuccessful calls also.I used to use asterisk so in cdr I need duration and billsec separately.As far as I understand kamailio can give only one of them.Is there a way to get them? Here is my acc configuration: #!ifdef WITH_ACCDB modparam("acc", "cdr_enable", 1) modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 0) modparam("acc", "db_url", "mysql://root: secret@localhost/mydb") modparam("acc", "log_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_user=$rU;dst_domain=$rd") #!endif What should I exactly do?Can someone help? Thanks in Advance. BETÜL CERİT

2 2

Kamailio vs Asterisk - Not found in usrloc
by Alexandr Usov 19 Mar '14

19 Mar '14

Kamailio (192.168.182.1) and Asterisk (192.168.182.24) realtime integration by http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. Kamailio have a public interface also. Grep from debug on kamailio 4.2.1 on call from asterisk extensions/peers: Mar 19 14:51:50 netbox /usr/sbin/kamailio[547784]: DEBUG: usrloc [udomain.c:602]: db_load_urecord(): aor 101(a)192.168.182.1 not found in table location Mar 19 14:51:50 netbox /usr/sbin/kamailio[547784]: DEBUG: registrar [lookup.c:158]: lookup(): '101(a)192.168.182.1' Not found in usrloc Mar 19 14:51:50 netbox /usr/sbin/kamailio[547785]: DEBUG: usrloc [udomain.c:602]: db_load_urecord(): aor 102(a)192.168.182.1 not found in table location Mar 19 14:51:50 netbox /usr/sbin/kamailio[547785]: DEBUG: registrar [lookup.c:158]: lookup(): '102(a)192.168.182.1' Not found in usrloc Kamailio forwards registration to Asterisk and store location of peer with domain name sip.domain.tld. But Asterisk dials peer to peer in local dialplan - to IP of kamailio. asterisk1*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 101/101 192.168.182.1 D No No A 5060 Unmonitored Cached RT 102/102 192.168.182.1 D No No A 5060 Unmonitored Cached RT asterisk1*CLI> sip show peer 102 * Name : 102 Description : Realtime peer: Yes, cached Secret : <Not set> MD5Secret : <Not set> Remote Secret: <Not set> Context : ext-local Record On feature : automon Record Off feature : automon Subscr.Cont. : <Not set> Language : Tonezone : <Not set> AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromUser : 102 FromDomain : sip.domain.com Port 5060 Callgroup : Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 6/0 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 252 Insecure : no Force rport : No Symmetric RTP: No ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Path support : No Path : N/A Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.182.1:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 102 SIP Options : (none) Codecs : (gsm|ulaw|alaw|h263|testlaw) Codec Order : (none) Auto-Framing : No Status : Unmonitored Useragent : kamailio (4.1.2 (x86_64/linux)) Reg. Contact : sip:102@192.168.182.1:5060 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 Encryption : No /etc/asterisk/extensions.conf: ... [ext-local] ; our phones use 3 digit extensions, starting with 1 exten => _1XX,1,DumpChan(verbose) exten => _1XX,n,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,101,Voicemail(${EXTEN},b) exten => _1XX,102,Hangup Also strabge why I see user.pub.lic in Variables on channel during call: ... SIPCALLID=GOB1.VSLU0SOTjG-u8jjarzXv.lJTjin SIPDOMAIN=netbox.teleservice.com.ua SIPURI=sip:101@user.pub.lic.ip:35835 Do I need rewrite To header with change IP to domain name? Maybe in this section: #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; }

1 0

Ugrade Kamailio 4.0.2 to 4.1.2
by Helena Garcia-Nieto 19 Mar '14

19 Mar '14

Hi! I am not used to manage the software upgrade and I would like to have a more detailed explanation… I've searched in the web and in the mailing list but I need a more step-by-step explanation for newbie that what I found. I downloaded the kamailio version 4.0.1 some time ago (http://www.kamailio.org/pub/kamailio/latest/src/) when it was latest release. I untar it on /opt/kamailio-4.0.2 and did make FLAVOUR=kamailio include_modules="db_mysql " cfg, Make all and make install. Now I would like to upgrade to the 4.1.2 version but I am not sure how. When you say: " just replace the binaries." What do you mean? Should I delete the /opt/kamailio-4.0.2 folder and untar the (http://www.kamailio.org/pub/kamailio/latest/src/) on a new /opt/kamailio-4.1.2 and make again? Should I remove anything else? I have modified the config file, and the kamctlrc to use mysql as dbengine. Then after making it should I execute the sql script for adapting the db? And then connecting it by modifying the config file? I am sorry if that seems a really silly question but I'd appreciate a really step by step guide help! Thanks in advanced Helena -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Moritz Graf Sent: jueves, 05 de diciembre de 2013 13:48 To: mohammed alyaseen; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Ugrade Kamailio 4.0.4 to 4.1.0 Hi Medo, just replace the binaries. see http://www.kamailio.org/w/kamailio-v4-1-0-release-notes/ http://www.kamailio.org/wiki/install/4.1.x/git greetz ps i assume u mean [...]to the new version 4.1.0 or Do I [...] Am 05.12.2013 10:29, schrieb mohammed alyaseen: > Hello > > I am using kamailio 4.0.4. I want to change to kamailio 4.1.0. Can the > version 4.0.4 be upgraded to the new version 4.0.4 or Do I have to > uninstall it and install the 4.1.0? > > Thanks in advance > > Medo -- Moritz Graf, B.Sc. Betrieb NGN-Plattform G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg Telefon +49 (9 41) 69 85 - 1 86 Telefax +49 (9 41) 69 85 - 2 86 mailto:moritz.graf@g-fit.de http://www.g-fit.de G-FIT Gesellschaft für innovative Telekommunikationsdienste mbH & Co. KG, Kommanditgesellschaft, Sitz Regensburg, Registergericht Regensburg, HRA 7626; Geschäftsführer: Dipl.Inf. (FH) Alfred Rauscher

1 0

TLS websocket problem
by jaflong jaflong 19 Mar '14

19 Mar '14

Hi List, Any suggestions on why I am getting the following issues. I can get a successful tls connection when I connect with http Tested by having this in kamailio.cfg event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); xhttp_reply("200", "OK", "text/html","<html><body>Received HTTP request to $hu from [$si:$sp] with protocol $proto</body></html>"); xlog("L_INFO", "HTTP Request Received\n"); ...... Going to https://10.1.2.3:6443 gives this Received HTTP request to / from [10.1.1.1:58179] with protocol tls This seems ok so far. However problems are encounter when trying to connect by WSS. It appears to not accept wss connections. Any ideas please JSSIP settings var configuration = { 'ws_servers': 'wss://10.1.2.3:6443/ws', 'uri': 'sip:webrtc@10.1.2.3', 'stun_servers': '10.1.2.3', 'turn_servers': null, 'register': true, 'password': 'webrtc' ...... Browser Console log JsSIP | UA | configuration parameters after validation: jssip-0.3.0.min.js:11 · via_host: "m3lmqpohnnin.invalid" jssip-0.3.0.min.js:11 · password: "webrtc" jssip-0.3.0.min.js:11 · register_expires: 600 jssip-0.3.0.min.js:11 · register_min_expires: 120 jssip-0.3.0.min.js:11 · register: true jssip-0.3.0.min.js:11 · registrar_server: sip:10.1.2.3 jssip-0.3.0.min.js:11 · ws_server_max_reconnection: 3 jssip-0.3.0.min.js:11 · ws_server_reconnection_timeout: 4 jssip-0.3.0.min.js:11 · connection_recovery_min_interval: 2 jssip-0.3.0.min.js:11 · connection_recovery_max_interval: 30 jssip-0.3.0.min.js:11 · use_preloaded_route: false jssip-0.3.0.min.js:11 · no_answer_timeout: 60000 jssip-0.3.0.min.js:11 · stun_servers: ["stun:10.1.2.3"] jssip-0.3.0.min.js:11 · turn_servers: [] jssip-0.3.0.min.js:11 · trace_sip: false jssip-0.3.0.min.js:11 · hack_via_tcp: false jssip-0.3.0.min.js:11 · hack_ip_in_contact: false jssip-0.3.0.min.js:11 · uri: sip:webrtc@10.1.2.3 jssip-0.3.0.min.js:11 · ws_servers: [{"ws_uri":"wss://10.1.2.3:6443/ws","sip_uri":"<sip:10.1.2.3:6443;transport=ws;lr>","weight":0,"status":0,"scheme":"WSS"}] jssip-0.3.0.min.js:11 · instance_id: "5d2898f4-af42-41cf-865b-848d2ba9a3f2" jssip-0.3.0.min.js:11 · jssip_id: "fagpa" jssip-0.3.0.min.js:11 · hostport_params: "10.1.2.3" jssip-0.3.0.min.js:11 · authorization_user: "webrtc" jssip-0.3.0.min.js:11 JsSIP | EVENT EMITTER | adding event newMessage jssip-0.3.0.min.js:9 JsSIP | EVENT EMITTER | adding event newRTCSession jssip-0.3.0.min.js:9 JsSIP | EVENT EMITTER | adding event registrationFailed jssip-0.3.0.min.js:9 JsSIP | EVENT EMITTER | adding event unregistered jssip-0.3.0.min.js:9 JsSIP | EVENT EMITTER | adding event registered jssip-0.3.0.min.js:9 JsSIP | EVENT EMITTER | adding event disconnected jssip-0.3.0.min.js:9 JsSIP | EVENT EMITTER | adding event connected jssip-0.3.0.min.js:9 JsSIP | UA | user requested startup... jssip-0.3.0.min.js:10 JsSIP | TRANSPORT | connecting to WebSocket wss:/10.1.2.3:6443/ws jssip-0.3.0.min.js:9 s {cache: Object, configuration: Object, dialogs: Object, registrator: null, applicants: Object…} call.html:62 JsSIP | EVENT EMITTER | new listener added to event registered jssip-0.3.0.min.js:9 JsSIP | EVENT EMITTER | new listener added to event registrationFailed jssip-0.3.0.min.js:9 JsSIP | EVENT EMITTER | new listener added to event newRTCSession jssip-0.3.0.min.js:9 JsSIP | TRANSPORT | WebSocket connection error: [object Event] jssip-0.3.0.min.js:9 JsSIP | TRANSPORT | WebSocket disconnected (code: 1006) jssip-0.3.0.min.js:9 JsSIP | TRANSPORT | WebSocket abrupt disconnection jssip-0.3.0.min.js:9 JsSIP | UA | transport wss://10.1.2.3:6443/ws failed | connection state set to 2 jssip-0.3.0.min.js:11 JsSIP | EVENT EMITTER | emitting event disconnected jssip-0.3.0.min.js:9 JsSIP | UA | next connection attempt in 2 seconds Kamailio [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/kamailio-selfsigned.key certificate = /usr/local/etc/kamailio/kamailio-selfsigned.pem DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 10.1.1.1 14(17989) DEBUG: <core> [tcp_main.c:1096]: tcpconn_new(): tcpconn_new: on port 50261, type 3 14(17989) DEBUG: <core> [tcp_main.c:1408]: tcpconn_add(): tcpconn_add: hashes: 1825:534:2831, 39 14(17989) DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x89af40, 25, 2, 0x7f42527a75b8), fd_no=19 14(17989) DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x89af40, 25, -1, 0x0) fd_no=20 called 14(17989) DEBUG: <core> [tcp_main.c:4302]: handle_tcpconn_ev(): tcp: DBG: sending to child, events 1 14(17989) DEBUG: <core> [tcp_main.c:3973]: send2child(): selected tcp worker 2 12(17987) for activity on [tls:10.1.2.3:6443], 0x7f42527a75b8 12(17987) DEBUG: <core> [tcp_read.c:1510]: handle_io(): received n=8 con=0x7f42527a75b8, fd=5 12(17987) DEBUG: tls [tls_server.c:178]: tls_complete_init(): Using TLS domain TLSs<default> 12(17987) DEBUG: tls [tls_domain.c:609]: sr_ssl_ctx_info_callback(): SSL handshake started 12(17987) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 12(17987) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f42527a75b8 n=1099 fd=5 12(17987) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= 12(17987) DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x8df1e0, 5, 2, 0x7f42527a75b8), fd_no=1 12(17987) DEBUG: tls [tls_domain.c:621]: sr_ssl_ctx_info_callback(): SSL handshake done 12(17987) DEBUG: tls [tls_domain.c:624]: sr_ssl_ctx_info_callback(): SSL disable renegotiation 12(17987) DEBUG: tls [tls_server.c:348]: tls_accept(): TLS accept successful 12(17987) DEBUG: tls [tls_server.c:355]: tls_accept(): tls_accept: new connection from 10.1.1.1:50261 using TLSv1/SSLv3 AES256-SHA 256 12(17987) DEBUG: tls [tls_server.c:358]: tls_accept(): tls_accept: local socket: 10.1.2.3:6443 12(17987) DEBUG: tls [tls_server.c:369]: tls_accept(): tls_accept: client did not present a certificate 12(17987) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 12(17987) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f42527a75b8 n=250 fd=5 12(17987) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= 12(17987) DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7f42527a75b8, FD 5 12(17987) DEBUG: <core> [tcp_read.c:1293]: tcp_read_req(): tcp_read_req: EOF 12(17987) DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8df1e0, 5, -1, 0x10) fd_no=2 called 12(17987) DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f42527a75b8, state -1, fd=5, id=39 12(17987) DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f4252793e90 14(17989) DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f42527a75b8, -1 from 2 14(17989) DEBUG: tls [tls_server.c:597]: tls_h_close(): Closing SSL connection 0x7f4252793e90

1 0

Re: [SR-Users] [Kamailio-Users] [Kamalito-Users] Sip MESSAGE
by Peter Reid 19 Mar '14

19 Mar '14

Hi Alex, Please let me make a comment here in support of Abdul Hakeem's comment in respect of Kamailio supporting a SIP to SMPP translation module and in furtherance of this thread (conversation). I believe Kamailio, as an open source community would glean the benefit of supporting a feature that could see better funded players use Kamailio, with SIP to SMPP, as an alternative strategy to supporting proprietary solutions. As networks look towards 4G technology the support of such a feature could see Kamailio used as a gateway between the origination technology and the legacy SMSC gateway. This is a solution, I believe, many will need to consider looking forward. Best Regards, Peter Reid Morodo Operations Director Manager - Mobile Software Development - ET Mob: +34603002300 E-mail: peter.reid(a)morodo.co.uk 04-04-2013 9-12-55Morodo Limited <mailto:support@morodo.co.uk> support(a)morodo.co.uk / <http://www.mo-call.com> www.mo-call.com Place of Registration: Companies House, England & Wales / Registered Office: 7 Breasy Place, 9 Burroughs Gardens Hendon, London NW4 4AU --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

1 0

Query About IMS request fetching
by Muhammad Nadeem 19 Mar '14

19 Mar '14

Hi, I am working on an educational research project, regarding IMS. I want to capture IMS request from IMS client and compare it with my defined rules. If the request satisfy my rules then it will be forwarded to IMS core. Could you please tell me, where i can do changing in IMS server and accomplish my goal. I am currently looking in files on following link. http://svn.berlios.de/svnroot/repos/openimscore/ Thanks in advance -- Best Regards Muhammad Nadeem International Islamic University Islamabad

1 0

Problem with TLS config
by Rene Montilva 19 Mar '14

19 Mar '14

Hi List i'm trying config sip with tls module, my config for module in kamailio.cfg is this: #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif in tls.cfg is this: [server:192.168.1.1:5061] method = SSLv23 verify_certificate = no require_certificate = no private_key = /etc/kamailio/key.pem certificate = /etc/kamailio/cert.pem [client:default] verify_certificate = no require_certificate = no But when a reload kamailio i get the follow error: Mar 18 16:36:50 softswitch /usr/sbin/kamailio[23330]: ERROR: tls [tls_domain.c:906]: load_private_key(): TLSs<default>: Unable to load private key file '/etc/kamailio/cert.pem' Why try to find '/etc/kamailio/cert.pem' for a private key??

2 1

login to siremis failed 'siremis.user' doesn't exist
by malik sherif 18 Mar '14

18 Mar '14

2 3

Redirect, maddr, and domain
by Marc Soda 18 Mar '14

18 Mar '14

I'm trying to handle a redirect with get_redirects(). It seems that Kamailio is ignoring the maddr param on the contact header. Is there a way to force maddr to be used? The Contact header on the 302 looks like this: <sip:2404441112@domain.com:5060 ;maddr=1.1.1.1>;q=0.5,<sip:2404441112@comain.com:5060;maddr=2.2.2.2>;q=0.25 The message is then being sent to domain.com, rather than 1.1.1.1.

3 4

Query About Request Fetching
by Muhammad Nadeem 18 Mar '14

18 Mar '14

Hi, I am working on an educational research project, regarding IMS. I want to capture IMS request from IMS client and compare it with my defined rules. If the request satisfy my rules then it will be forwarded to IMS core. Could you please tell me, where i can do changing in IMS server and accomplish my goal. I am currently looking in files on following link. http://svn.berlios.de/svnroot/repos/openimscore/ Thanks in advance -- Best Regards Muhammad Nadeem Muhammad Ali Jinnah University

1 0

error messages from pv functions
by Juha Heinanen 18 Mar '14

18 Mar '14

in order to avoid scripting errors, it would be nice if pv functions would return an error message when applied to non-applicable pvs. for example, pv_isset() and pv_unset() applied on a script variable. -- juha

1 0

login to siremis failed 'siremis.user' doesn't
by Hugo Servellon 17 Mar '14

17 Mar '14

Hi There, A fast solution would be to re-run the installation procedure. To do so , follow the steps below : a)Remove siremis-4.0.0/siremis/install.lock b)While reinstalling and setting up the database properties , check all of the options available at the bottom of the form . That worked for me , Cheers

2 1

Transfer Problem
by Noriyuki Hayashi 17 Mar '14

17 Mar '14

Hi, Now I am going to use Asterisk-11.7.0 and Kamailio-4.1.1 on CentOS-5.10 x86-64. Backend is PostgreSQL-9.2.6 as Realtime. Conditons, Sip register is working. Extention to Extention is working. Echo test and meetme is fine. Only has the problem is When I Tranfer to other extention number as same context. In this case, Kamilio is using default context transfer is aborted. If anyone has any idea with much appriciated. Thank you. Noriyuki Hayashi Japan

2 1

where does iceserver configuration get set
by jaflong jaflong 17 Mar '14

17 Mar '14

Hi List Is ice settings initiaised on the browser side or Kamailio websocket side? Where are the setting parameters created for a=candidate In the debug I get such logs as a=candidate:3164634682 1 tcp 1509957375 xxx.xxx.xxx.xxx 0 typ host generation 0 a=candidate:3164634682 2 tcp 1509957375 xxx.xxx.xxx.xxx 0 typ host generation 0 a=ice-options:google-ice I am connecting with with jssip like the snippet below. There is nothing set for stun/turn. So how then does it use a=ice-options:google-ice. Does it set google as default var configuration = { .... .... stun_servers: '', turn_servers: null, .... .... }; // User Agent instance var coolPhone = new JsSIP.UA(configuration); Does anyone know are these the only options is relation to ice onfig, stun_servers: turn_servers: null,

1 0

evapi module question
by Alex Balashov 17 Mar '14

17 Mar '14

Hi, Does evapi_relay() have to emit JSON blobs? Or can it be used to emit generic strings? Thanks! -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

Endpoint registration tracking and reporting
by Nick Raper 17 Mar '14

17 Mar '14

Hi all, What is the best way to monitor endpoint registrations? When one of our clients has a link failure or for some reason they lose their SIP registration with Kamailio, we'd like to be able to see this so that we can respond and get the trunk re-registered. We do monitor the physical links but endpoints, (usually Asterisk based), often don't re-register after connectivity is restored. I was thinking of a script that dumps the 'location' table every 10 minutes and compares it to the previous 'location' table dump and reports on the changes. Is there a better way of doing this? Regards, [Description: cid:image001.jpg@01C8D7A0.651BD080] Nick Raper 4th Dimension Technology | South Africa Office: +27 21 506 6260 | 24 hour helpline: +27 21 506 6260 | Fax: +27 86 610 5792 Email: nick(a)4dt.co.za<mailto:nick@4dt.co.za> | Web: www.4dt.co.za<http://www.4dt.co.za/> P Please consider the environment before printing this email.

3 4

Softphone cant connect
by aawaise 17 Mar '14

17 Mar '14

Hello, I have deployed a kamailio server. Now I am trying to set up a voip call between two soft phones (EyeBeam) located on local network. I am using "Outbound proxy option" on softphone. It shows "peer to peer mode enabled". Settings are shown in figure attached <http://sip-router.1086192.n5.nabble.com/file/n125931/1.jpg> But when I make a call. It doesnot connect. Using wireshark on kamailio server, I found that in response to "SIP : Request" packet, it sends ICMP packet "Destination Unreachable: Port Unreachable". Firewall is disabled on all concerned systems. What can be the possible reason. Any one will help please. Regards, Awais -- View this message in context: http://sip-router.1086192.n5.nabble.com/Softphone-cant-connect-tp125931.html Sent from the Users mailing list archive at Nabble.com.

1 0

alias behaving differently than droute
by Pete Ashdown 15 Mar '14

15 Mar '14

I presumed that aliases could be used for quickie destination routing, but for some reason my RTP isn't connecting up after. If I put the same alias into droute, it works. Am I understanding aliases wrong?

3 3

sip code and sip reason read only
by Kelvin Chua 14 Mar '14

14 Mar '14

How can i alter at least the sip reason in transit? $rr is read only for example: what i wanted to do is when an egress switch returned, 404 i am ugly i wanted to return this to ingress 404 he is ugly Kelvin Chua

1 1

bug in textops module
by Juha Heinanen 14 Mar '14

14 Mar '14

according to textops readme: 2.1. Kamailio Modules The following modules must be loaded before this module: * sl -- Stateless replies. however, i get these errors if pv module is not loaded before textops module: Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: <core> [pvapi.c:790]: pv_parse_spec2(): error searching pvar "avp" Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: <core> [pvapi.c:994]: pv_parse_spec2(): wrong char [s/115] in [$avp(s:rpid)] at [5 (5)] Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: siputils [rpid.c:73]: init_rpid_avp(): malformed or non AVP $avp(s:rpid) AVP definition Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: siputils [siputils.c:235]: mod_init(): failed to init rpid AVP name Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: <core> [sr_module.c:970]: init_mod(): init_mod(): Error while initializing module siputils (/usr/lib/sip-proxy/modules/siputils.so) please fix textops module so that it does not depend on pv module if rpid_avp is not set. -- juha

2 7

SIREMIS redirecting to login prompt issue
by malik sherif 14 Mar '14

14 Mar '14

I configured apache web server and also installed siremis 4.1.1 successfully but i couldn't login and my problem is after " welcome to Siremis installation wizard" I went through steps 1to 4 without issue. On the last step I even receive installation complete congratulation also display username and login password but after 10 seconds it goes back to step 1. I was expecting to see a login window but I am stack. Did anybody experience this issue? Any help greatly appreciated. Thanks Abdulmalik

5 11

Kamailio World 2014 - Interactive sessions
by Daniel-Constantin Mierla 14 Mar '14

14 Mar '14

This year we scheduled more interactive sessions, to reveal better how Kamailio and open source RTC applications can be used out there. During the afternoon of April 2, a group of technical tutorials target to teach how to use various platforms for typical use cases: sip:provider CE, Kamailio for IMS and prepaid systems, SEMS. At the end of April 3, we have the open debate moderated by Randy, the host of weekly VoIP Users Conference. Before noon on April 4, James Body challenges everyone around to do a dangerous demo, impress the audience and win some goodies -- prepare yours in time! More details are presented in the schedule: - http://conference.kamailio.com/k02/schedule/ About two weeks and a half till the start of the event, there is no other meeting alternative in Europe for networking with professionals in large real time communications deployments and open source telephony, be sure you don't miss it. Registration details are available at: - http://conference.kamailio.com/k02/registration/ Credits to event sponsors that ensured a good location and excellent content by participation of world wide speakers - Asipto, FhG Fokus, Sipwise, IT Center, Sipgate, Simwood, Digium, NG Voice, Core Network Dynamics: - http://conference.kamailio.com/k02/sponsors/ See you very soon in Berlin! Meanwhile work on your idea for a dangerous demo! -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, Berlin, Germany http://www.kamailioworld.com

1 0

401 after a 302
by Marc Soda 14 Mar '14

14 Mar '14

Can someone tell me how to handle a 401 from a 302 redirect? I am attempting to register with the uac module. Normally, I set a failure route for the 401 and call auc_auth(). In this case, I am receiving a 302, sending the register with get_redirects(), then getting a 401. However, Kamailio seems to be ignoring the 401. I see it in an onrely_route, but not a failure route. failure_route[REG_AUTH] { xlog("L_NOTICE","route[REG_AUTH]\n"); # Handle redirects if ($T_reply_code == 302) { get_redirects("*", "Redirected"); t_on_failure("REG_AUTH"); route(RELAY); } else if ($T_reply_code == 401) { uac_auth(); t_on_reply("REG_REPLY"); route(RELAY); } } route[RELAY] { xlog("L_NOTICE","route[RELAY] ($rm)\n"); if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if (!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if (!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if (!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } xlog("L_NOTICE","t_relay()'ing ($rm)\n"); if (!t_relay()) { sl_reply_error(); } exit; } Thanks! Marc

1 1

siremis doesn't works with postgresql
by Rene Montilva 14 Mar '14

14 Mar '14

Hi List i want to config siremis with postgresql and really to much things fail for the install, at the end y left the siremis db with mysql and kamailio with postgresql but the problems continue, when i want to show subscriber list from siremis interface, i get an error and check the postgresql log and found this LOG: sentencia: SET NAMES 'utf8' ERROR: error de sintaxis en o cerca de «`» en carácter 101 SENTENCIA: SELECT T0.id, T0.username, T0.domain, T0.password, T0.email_address, T0.ha1, T0.ha1b, T0.rpid FROM `subscriber` T0 LIMIT 10 what could be the right config?

2 1

Inter domain routing
by Owais ul Haq 14 Mar '14

14 Mar '14

Hello, I have two kamailio servers running independently with different domain names. I want to route calls between those two servers. Can it be done ? If yes then How ? Thanks.

5 6

Soft phone Issue
by Owais ul Haq 14 Mar '14

14 Mar '14

Hello, I have configured a kamailio server on my local network, gave it a dummy domain like "abcd.tk". I have a number of computers on my local network. I want to run eyebeam on them and connect to my kamailio server. Server is running fine. When I try to connect Eyebeam to Server, it goes to DNS to find IP for "abcd.tk". I have entered the value "192.168.1.10 = abcd.tk" in "hosts" file of all machines. File is located at "C:\WINDOWS\system32\drivers\etc\hosts". On other hand, when I ping abcd.tk from client computers. They ping 192.168.1.10. So Operating systems verify the entry "192.168.1.10 = abcd.tk", but problem lies with soft phones. I have tried X-Lite and Eyebeam. How can I make softphone understand that abcd.tk = 192.168.1.10 and please go to this adress, not DNS ?? Help please :( Regards, Awais.

4 3

Softphone not connecting
by Owais ul Haq 14 Mar '14

14 Mar '14

Hello, I have deployed a server. It is starting fine. Connection with MySQL database is also okay. Siremis is also working fine. But softphone "Eyebeam" can't login as a user to the server. Client functionality is ok. As same client with same setting is working good with other kamailio server. How can I trace down the error ?? Please guide. Regards, Owais

2 1

Two kamailio servers sharing database
by Owais ul Haq 14 Mar '14

14 Mar '14

Hello, I have deployed two kamailio servers. Both are sharing a common database deployed on Windows 2008 Server using MySQL. Now domain of one kamailio server is lets say "abc.com" and other's is "xyz.com". I want to know that if a user is created by user id "user1(a)abc.com" with password "qazwsx@123" in the database. And a user try to login from 2nd server "xyz.com" using user id "user1(a)xyz.com" with password "qazwsx@123". Will he able to login using this shared database ? Also please tell which modules controls this ? As I have few more questions also concerning this.

2 1

help with SIP Notify message, to make a endpoint (SPA504G and similar) to reboot.
by Pedro Niño 13 Mar '14

13 Mar '14

Sorry for intruding, but this is puzzling me a lot. I followed the guide to make Kamailio work with asterisk and realtime, Kamailio version 4 and asterisk 11. (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ) up until now its working, and now want to get the phones (CISCO SPA504G, former linksys) to get the SIP notify command and reload to get their new config, from the asterisk cli the cli command is "sip notify spa-reboot <username>" The problem is that, using the setup in the guide, the phones registers with kamailio, kamalio then strip the password and register with no password to the asterisk, as supposed to be. by that, when asterisk sends the SIP Notify with the reboot commands, it arrives to the kamailio, and is forwarded to phone, and the phone answers with a '401' and a digest. In the flow, the kamailio (in the middle) receives the 401 and sends again the sip message to asterisk, and asterisk uses the digest to cipher the password and reply, but as asterisk is configured with no password, the reply contains as 'wrong' password (the empty/blank one) is there any way to make kamailio intercepts again that '401' and reply with a crafted message, with the right digest, password, (the one in the ul table) and user, so the phone can accept the order and reboots? The other (ugly) option, is to remove the auth from the phone, for the Sip Provisioning, but that would leave and open door to a reboot attack without auth needed from any IP. And I dont like that option. As we talk, I managed to create a routing logic, but is that the right approach? : =================== request_route { if (is_method("NOTIFY")) { #xlog("NOTIFY received: $rs"); t_on_failure("MANAGE_FAILURE_401"); #try to intercep a 401 } ..... } failure_route[MANAGE_FAILURE_401] { if (t_check_status("401")) { #401 detectado dentro de un Notify, #probablemente ha sido por una orden de reboot. #hay que construir un paquete e intentar enviar, con el digest adecuado.... $uac_req(method)="NOTIFY"; $uac_req(ruri)="sip:"; $uac_req(furi)="sip:"; $uac_req(turi)="sip:"; $uac_req(hdrs)="Contact: <sip:"+ ">\r\n"; #Paquete construido, debe ser enviado uac_req_send(); xlog("401 detectado"); # exit; } =================== Thanks in advance.....

2 2

Problem install siremis and kamailio with postgresql
by Rene Montilva 13 Mar '14

13 Mar '14

Hi list i tried install siremis with postgresql db, but when i was in step 2 and select postgresql driver and fill all the required fields, but i saw siremis installer try to connect to postgresql with www-data user, why this happend

1 0

Bizarre dialog profile tracking behaviour
by Alex Balashov 13 Mar '14

13 Mar '14

Hello, We've got an issue we've been trying to track for days with 'dialog', where the dialogs go through the following life cycle: 1. INVITE --> 2. <-- 407 challenge 3. ACK --> 4. INVITE --> 5. <-- 100 Trying 6. <-- 180 Ringing 7. <-- 183 Session Progress 8. <-- 200 OK 9. ACK --> 10. BYE --> 11. <-- 200 OK Somewhere around steps #6-7 (18x messages), the dialogs no longer appear to be counted in the dialog profile they're attached to, as gleaned from 'kamctl fifo profile_get_size <profile> <key>'. The dialog count in the profile falls to 0. Naturally, I suspected a SIP issue that was preventing the dialog state from being tracked correctly. However, subsequent investigation revealed that these very same dialogs are tracked just fine throughout their entire lifetime. The 'kamctl fifo dlg_list' command shows them to have the right states at the right times. So, the dialog itself isn't going away from tracking. Its attachment to the given profile seems to be the issue. I can't find any logical explanation for this in the code, nor any technical reasoning to support this hypothesis. However, that's what seems to be happening. I'm not sure that this is the issue; it might be another issue masquerading as having this effect. I'm just trying to rule possibilities out. Are there any situations that can cause a dialog that is otherwise being normally tracked to be dumped from a profile of which it was previously part? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 8

Problemas with kamctl ul show - Kamailio 4.1
by Raphael Borges 13 Mar '14

13 Mar '14

Hello, I'm having a problem with the command kamctl ul show, and I do not understand what is happening. On my server when I run just "kamctl ul show" command works fine but when I run "kamctl ul show [<username>]" an error happens. Before when I was using Kamailio 3.1, its work ok. I looked for some documentation and found nothing about this. For example: *$ /home2/local/kamailio/sbin/kamctl ul show* Domain:: location table=512 records=1 max_slot=1 AOR:: 2501 Contact:: sip:2501@192.168.XXX.XXX:5060;transport=udp Q= Expires:: 3594 ... Domain:: aliases table=512 records=0 max_slot=0 *$ /home2/local/kamailio/sbin/kamctl ul show 2501* 404 AOR not found Thanks for Help. Best regards. Raphael

2 6

Reminder: about private emails
by Daniel-Constantin Mierla 12 Mar '14

12 Mar '14

A message sent almost one year ago, but based on volume of private emails related to free support on SIP, Kamailio or other applications, perhaps it is time to refresh, with couple of extra bits for clarification: - http://lists.sip-router.org/pipermail/sr-users/2013-April/077700.html >>> I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is not recommended. The rule is also suggested in the mailing lists presentation page: - http://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. There are several common sense as well as technical reasons. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine. Also, other people subscribed to the list can learn from conversation or contribute to the discussion, resulting in better outcome - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. They are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else if you don't get this instantly, or this application is missing what so ever tutorial giving exactly the solution you need, or simply asking for full config of complex requirements, is not helping at all, but actually making people to ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, nobody here owe you anything. Therefore, if anyone sent me a private message asking about kamailio or continuing a discussion started on mailing lists, then the messages have to be resent on mailing list if one wants an answer. A quick look in my unfiltered folder revealed hundreds of messages during the past year, soon many will be deleted anyhow. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, Berlin, Germany http://www.kamailioworld.com

3 3

trixbox or asterisk as a media server for kamailio
by malik sherif 12 Mar '14

12 Mar '14

I am looking documentation as to how to integrate kamailio and Asterisk, I am not sure if I use the correct term "media server" but I would like to configure 3-way call, call waiting, call transfer , and call forwarding. Your help is greatly appreciated. Thanks Abdul

2 2

Kamailio LCR unable to relay new calls, memory issue
by Camila Troncoso 12 Mar '14

12 Mar '14

Hi, Last Friday I presented a increase of load in kamailio by a client. The load was almost twice as usual. By that time I encountered with many issues in kamailio. The calls were not relay to gateways, and a lot of messages about memory were appearing in log file. Kamailio didn't crash. When I remove the extra traffic all went back to normal. Is there a issue about memory leak or about to few memory resources to manage this amount of traffic? ERROR: tm [t_reply.c:1799]: ERROR: relay_reply: cannot alloc reply shmem ERROR: <core> [msg_translator.c:1782]: ERROR: build_req_buf_from_sip_req: out of memory ERROR: tm [t_fwd.c:431]: ERROR: print_uac_request: no shm mem ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches ERROR: tm [tm.c:1369]: ERROR: w_t_relay_to: t_relay_to failed ERROR: tm [t_reply.c:598]: ERROR: _reply_light: cannot allocate shmem buffer ERROR: tm [t_msgbuilder.c:371]: ERROR: cannot allocate shared memory ERROR: tm [t_msgbuilder.c:521]: ERROR: build_local_reparse: cannot build ACK request MERROR: tm [t_msgbuilder.c:371]: ERROR: cannot allocate shared memory ERROR: tm [t_msgbuilder.c:521]: ERROR: build_local_reparse: cannot build ACK request ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: No error (2/SL) ERROR: <core> [msg_translator.c:1782]: ERROR: build_req_buf_from_sip_req: out of memory ERROR: tm [t_fwd.c:431]: ERROR: print_uac_request: no shm mem ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches ERROR: tm [tm.c:1369]: ERROR: w_t_relay_to: t_relay_to failed ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: No error (2/SL) ERROR: <core> [sip_msg_clone.c:506]: ERROR: sip_msg_cloner: cannot allocate memory ERROR: tm [t_lookup.c:1338]: ERROR: new_t: out of mem: ERROR: tm [t_lookup.c:1478]: ERROR: t_newtran: new_t failed ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Regards, Camila *Camila Troncoso Solar **|* Ingeniero de Desarollo +56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com <http://www.redvoiss.com>*

2 12

Re: [SR-Users] Routing based on destination and source
by Keith 12 Mar '14

12 Mar '14

Hi Yeah I do already use the lcr module for routing to external carriers but is probably a little overkill for what I need to route that traffic internally. Is that my only real option? Cheers Keith

1 0

Kamailio Proxy for Freeswitch
by jay binks 12 Mar '14

12 Mar '14

So im using Kamailio as a load balancer and proxy in-front of a few Freeswitch boxes. Im not going to use RTP Proxy, as my FS boxes will be public on the internet ( Direct RTP connection is fine ). I want ALL SIP to go via Kamailio, but the issue im running into is that the SIP Contact header has the FS IP address in it in some cases. ( 180's etc ) Every bit of documentation I can find about re-writing the Contact header is geared around NAT, and thats not what im trying to do here.. Can anyone give some pointers on how to basically topology hide the FS boxes on the SIP front but leave their IP in the SDP. im going to have the firewalls setup so the SIP Can ONLY come from kamailio, so its quite important I get this all right. Any pointers would be appreciated. Thanks Jay

2 3

(no subject)
by malik sherif 12 Mar '14

12 Mar '14

I send couple of email to Asipto.com about Siremis login issue but I didn't get any response. I configured apache web server and also installed siremis 4.1.1 successfully but i couldn't login and my problem is after " welcome to Siremis installation wizard" I went through step 1to 4 without issue on the last step I even receive installation complete congratulation also display username and login password but after 10 seconds it goes back to step 1. I was expecting to see a login window but I stack. Did anybody experience this issue? or if there is a mailing list, can someone point to the right group. Any help greatly appreciated. Thanks Abdulmalik

2 1

Re: [SR-Users] Routing based on destination and source
by Keith 12 Mar '14

12 Mar '14

Hi all, I am looking for the best module to route calls based on the authenticated user that call came from and the destination number. Is there any current module to do this? Many thanks Keith

2 1

Fail Safe Environment for kamailio
by Owais ul Haq 12 Mar '14

12 Mar '14

Hello, Guys, I need some help. I am tasked to deploy "Fail Safe Architecture" for already running Kamailio Server in our organization. Presently one Kamailio Server is serving the cause with one internet connection and database on MySQL running on Windows 2008 Server R2(which is attached to kamailio server on other LAN card). Now the requirement is to deploy redundancy for internet connection and kamailio server. Like they want two kamailio servers running parallel, so if one fails, other will automatically take over and service will be available to users without any obstruction. At any cost, my organization want to absolutely minimize any possible blackout. To further elaborate my point. Here is the network we want. 2 incoming internet connections to 2 kamailio servers. Both attached to a single database placed at the backend. Now is it possible ?? If not what are constraints ?? What workaround is possible ?? I seriously need help as it is matter of job. Thanking in advance for help. P.S : kamailio server is running of Fedora machine in our environment. Cheers.

3 2

Web interface for kamailio
by malik sherif 11 Mar '14

11 Mar '14

Any recommendation web interface for gathering statistics other than SIREMIS? I installed SIREMIS but unable to resolve login problem and my request for help on ASIPTO didn't get any response also unable to find siremis mailing list. Your help is greatly appreciated. Thanks Abdulmailk

3 4

kamailio 4.1.X and ast 11.8
by lore 11 Mar '14

11 Mar '14

hello folks, i'm really new with kamailio (a couple fo days) and I need some suggestion/samples/idea/HELP to realize my environment. I've been starting to read the docs on website but there are a lot and i don't know where start and which kind of modules need to use I''d like to realize a trunk from kamailio and asterisk. Each one has their own sip users. kamailio sip users are some inside network and some outside network (behind nat) Could someone be son kind to show me the right way to configure kamailio for incoming/ougoing calls from/to asterisk and for using sip peers (behind nat and not)?. Also same samples are really preciated. Thanks a lot in advance Kind regards Lorenzo -- "Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non conta un cazzo, 1941 ... sono anche un autore"

1 0

milliseconds on $Ts
by Kelvin Chua 11 Mar '14

11 Mar '14

Hi guys, Is there a code somewhere in the project i can reuse to implement milliseconds on $Ts? for example introduce a "$Tms" or a "$TmS" to refer to a pv that stores milliseconds instead of seconds? Kelvin Chua

2 2

how to get value of route param?
by Juha Heinanen 10 Mar '14

10 Mar '14

rr module has check_route_param(re) function that can be used to check if local route header has a param that matches re. however, there does not exist a function to find out what is the value of a given rr param. for example, if route header has param ;param=foo, how to extract value foo of param? there is @rr.uri.params select that together with {param.value,name} transformation might be able to do the job? -- juha

3 3

How to encode domain in username for Asterisk REGISTER forwarding (was: How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?)
by Alex Villacís Lasso 10 Mar '14

10 Mar '14

On the previous issue (disjoint networks), I eventually settled for binding Asterisk on localhost:5080, and using rtpproxy to route media to the networks as required. The only snag is that I had to write a script to rewrite the Kamailio configuration in order to take current IPs into account. Now, for the same project. What I want now is to be able to express multiple domains with an Asterisk backend. That is: given pbx.company1.com and pbx.company2.com, two DNS addresses that point to the same server, I want to have separate accounts bob(a)pbx.company1.com and bob(a)pbx.company2.com, with separate registration and accounting, without collisions. I know about the Kamailio "domain" module, and I have enabled WITH_MULTIDOMAIN and filled the kamailio.domain table for it to work. Now the authentication is taking domains into account. Next, I want to express two non-colliding registrations for Asterisk. What comes to mind is username rewrite - Kamailio sees bob(a)company1.com, and forwards a REGISTER to Asterisk using bob_company1_com as the SIP username. I am looking right now at the WITH_ASTERISK block that uses the uac module: # 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(furi)="sip:" + $au + "@" + $var(rip) + ":5060"; $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ":5060"; $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(); } #!endif The above is the non-encoding version. What is the best way to modify it to do a domain encoding in the username? Or, if a better solution exists, what is it?

2 1

Debian init script "log_faiulre_msg"
by Corey Edwards 08 Mar '14

08 Mar '14

There's a typo in the Debian init script. Is this the correct place to report packaging bugs? --- /etc/init.d/kamailio 2014-03-06 13:42:23.000000000 -0700 +++ /tmp/kamailio 2014-03-07 08:41:05.000000000 -0700 @@ -52,7 +52,7 @@ log_failure_msg "Not starting $DESC: invalid configuration file!" log_failure_msg log_failure_msg "$out" - log_faiulre_msg + log_failure_msg exit 1 fi } Corey

3 3

Billing solutions for Kamailio?
by Sergey Kharlamov 07 Mar '14

07 Mar '14

Hi, Can someone recommend me good billing solution for kamailio? Ideally it should contain admin panel, user cabinet etc... -- Sergey Kharlamov

2 1

p-cscf - out of mem
by Daniel Ciprus 07 Mar '14

07 Mar '14

Hi, Has anybody seen this : Mar 2 17:25:16 kamailio kam-pcscf[20741]: ERROR: <core> [msg_translator.c:2011]: generate_res_buf_from_sip_res(): out of mem Mar 2 17:25:16 kamailio kam-pcscf[20741]: ERROR: tm [t_reply.c:1943]: relay_reply(): ERROR: relay_reply: no mem for outbound reply buffer followed by: Mar 2 17:27:03 kamailio kam-pcscf[20741]: ERROR: <core> [msg_translator.c:2011]: generate_res_buf_from_sip_res(): out of mem Mar 2 17:27:03 kamailio kam-pcscf[20741]: ERROR: tm [t_reply.c:1943]: relay_reply(): ERROR: relay_reply: no mem for outbound reply buffer Mar 2 17:27:03 kamailio kam-pcscf[20741]: ERROR: <core> [msg_translator.c:2168]: build_res_buf_from_sip_req(): ERROR: build_res_buf_from_sip_req: out of memory ; needs 386 Mar 2 17:27:53 kamailio kam-pcscf[20741]: ERROR: <core> [parser/contact/contact.c:194]: parse_contacts(): parse_contacts(): No memory left Mar 2 17:27:53 kamailio kam-pcscf[20741]: ERROR: <core> [parser/contact/parse_contact.c:59]: contact_parser(): contact_parser(): Error while parsing contacts Mar 2 17:27:53 kamailio kam-pcscf[20741]: ERROR: <core> [parser/contact/parse_contact.c:88]: parse_contact(): parse_contact(): Error while parsing However, seems like we still had enough mem: [root@kamailio kamailio]# vmstat 1 procs -----------memory---------- ---swap-- -----io---- --system-- -----cpu----- r b swpd free buff cache si so bi bo in cs us sy id wa st 3 0 0 9526348 230308 13282472 0 0 0 4 0 1 6 0 94 0 0 1 0 0 9526216 230308 13282524 0 0 0 0 1313 84424 17 8 75 0 0 1 0 0 9526092 230308 13282616 0 0 0 0 1482 86882 16 9 75 0 0 [root@kamailio monit.d]# kamailio -v version: kamailio 4.2.0-dev0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 17:42:42 Dec 18 2013 with gcc 4.4.7 Unfortunately I had to reload p-cscf to get service working again. -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 6

(no subject)
by malik sherif 07 Mar '14

07 Mar '14

Hello, I am using kamailio 4.1.1, I am wondering if kamailio supports call forwarding , three-way call, and call transfer natively? Does it expect the endpoints to do the RTP mixing? Thanks Absul

3 3

500 Forking capacity exceeded (12/SL)
by Camila Troncoso 07 Mar '14

07 Mar '14

Hi, I´m having problems with kamailio as LCR . I have a route with 14 different gateways, each one with priority 1 and all with the same weight. If one gateway is full, busy, down or answers with any failed response to LCR , I jump to the next gateway in the list. I´m presenting the problem that its stops jumping after 12 gateways and returns the message 500 Forking capacity exceeded (12/SL). I need to increase this limit , so all the gateways in the list give the call a try. Regards, Camila *Camila Troncoso Solar **|* Ingeniero de Desarollo +56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com <http://www.redvoiss.com>*

3 5

RV: 500 I'm terribly sorry, server error occurred (1/SL)
by Camila Troncoso 07 Mar '14

07 Mar '14

*De:* Camila Troncoso [mailto:ctroncoso@redvoiss.net] *Enviado el:* jueves, 06 de marzo de 2014 16:03 *Para:* sr-users(a)lists.sip-router.org *Asunto:* 500 I'm terribly sorry, server error occurred (1/SL) Hi, I have kamailio 3.2 as LCR with many gateways, weights and priorities. Out of no ware It presented this error today: 500 I'm terribly sorry, server error occurred (1/SL). This wasn't happening for all the calls , and was arbitrary. I read on google ,that this error comes form t_relay, so the server was unable to relay the call to the gateway. But all the gateways are reachable and active. The server was on load , almost 30 cps. The same configuration have been running for more than 2 years with any problem. Could these happened for heavy load? I have children=32 in config file. Has anyone an idea of what may happened? Regards, Camila Troncoso *Camila Troncoso Solar **|* Ingeniero de Desarollo +56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com <http://www.redvoiss.com>*

1 0

KAMAILIO 4.1.1 Forward SIP REGISTER
by Francesco Maria Magnini 07 Mar '14

07 Mar '14

Hi, I would like to have a Kamailio Registrar/Location server on my private network and another one facing the public side: KAM1 (Private Registrar) <—> KAM2 (Public Proxy) I would like also to configure the proxy in this way: - REGISTRATION TO Public Proxy KAM2 going to Private Registrar KAM1 In the reference guides I’ve seen almost always application specific configuration (WITH_ASTERISK, WITH_FREESWITCH). Basically, how it is possible to forwards REGISTER messages to a generic upstream Registrar? Which modules are involved? I’m using Kamailio 4.1.1 Regards, Francesco

3 4

RPM packages for v4.1.2, v4.0.6 and v3.3.7
by Daniel-Constantin Mierla 07 Mar '14

07 Mar '14

Hello, based on the releases made yesterday, the rpm packages for various Linux distributions are available via opensuse build services - the list of distributions and links to the repositories are published at: - http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, Berlin, Germany http://www.kamailioworld.com

1 0

Newbie looking for resources
by richard.seguin＠marisec.ca 07 Mar '14

07 Mar '14

Hi folks, I'm new to Kamailio, and I am looking for some documentation on how to write config files, what should be included... etc.. I'm pretty much looking for product documentation and tutorials, not info on SIP (already went through the sip school). - How do I write a basic config file - Basic auth and forward examples I'm familiar with the cookbooks, however they don't really provide context as to what the recommendations do. Any resources that people could provide would be greatly appreciated. Thanks, Richard Seguin

3 2

500 I'm terribly sorry, server error occurred (1/SL)
by Camila Troncoso 06 Mar '14

06 Mar '14

Hi, I have kamailio 3.2 as LCR with many gateways, weights and priorities. Out of no ware It presented this error today: 500 I'm terribly sorry, server error occurred (1/SL). This wasn't happening for all the calls , and was arbitrary. I read on google ,that this error comes form t_relay, so the server was unable to relay the call to the gateway. But all the gateways are reachable and active. The server was on load , almost 30 cps. The same configuration have been running for more than 2 years with any problem. Could these happened for heavy load? I have children=32 in config file. Has anyone an idea of what may happened? Regards, Camila Troncoso *Camila Troncoso Solar **|* Ingeniero de Desarollo +56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com <http://www.redvoiss.com>*

1 0

Kamailio v4.1.2 Released
by Daniel-Constantin Mierla 06 Mar '14

06 Mar '14

Hello, Kamailio SIP Server v4.1.2 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.1. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.2. For more details about version 4.1.2 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/03/kamailio-v4-1-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v4.0.6 Released
by Daniel-Constantin Mierla 06 Mar '14

06 Mar '14

Hello, Kamailio SIP Server v4.0.6 stable release is out. This is a maintenance release of the stable branch 4.0, that includes fixes since release of v4.0.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.x. Deployments running previous v4.0.x versions are strongly recommended to be upgraded to v4.0.6. For more details about version 4.0.6 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/03/kamailio-v4-0-6-released/ Note that 4.0.x is previous stable release series, the latest one is now 4.1.x, with v4.1.2 released today as well. In addition, 3.3.7 was packaged in order to have it included in the new APT repository and make transition to newer versions easier for those still on branch 3.3. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Incorrect IP address in path header
by Rob Moore 06 Mar '14

06 Mar '14

Hi All, So an interesting one here. I'm doing pass through registrations using the dispatcher module in Kamailio and I'm using the Path Module to set the Path header In Register request so that Freeswitch knows to follow the route via Kamailio to find the sipPhone in future. In summary: [SIP Phone]------[NAT]------[(externalip)Kamailio(internalip)]-------[FS-Registrar] The problem I'm having is that the Path header is being added with the external ip address of the Kamailio instance not the internal ip e.g. "Path: sip:(externalip);transport=tcp" which mean FS doesn't root new calls out to the SIPPhone correctly via kamailios internal ip. I have noticed that if I remove the ;transport=tcp from the FS-Register address in my dispatcher group so that the last hop (from kamailio to FS-Register) is done over UDP then the problem corrects itself. (although this does present new issues whilst calling out back along the path from FS, and we also have a prerequisite to use TCP so this doesn't help). Has anyone had any experience of this issue? I can't see what I've done wrong here. Many Thanks Rob

2 2

sqlops crash after out of memory
by Juha Heinanen 06 Mar '14

06 Mar '14

pkg memory run out here: Mar 4 09:00:20 localhost /usr/sbin/sip-proxy[31923]: ERROR: sqlops [sql_api.c:318]: sql_do_query(): no more memory then sip proxy crashed here: (gdb) where #0 0xb5fce89b in sql_reset_result (res=0xb6ce3438) at sql_api.c:233 #1 0xb5fd18db in sql_do_query (con=0xb6cd1e28, query=0xbfecd248, res=0xb6ce3438) at sql_api.c:410 #2 0xb5fd630d in sql_query (msg=0xb6ce8788, dbl=0xb6cd1e28 "\230\035Í¶\t", query=0xb70436a8 "\310Y\004\267P", res=0xb6ce3438 "\245\246=|\r3Î¶\006") at sqlops.c:209 ... goto error was executed when if(res->vals[i]==NULL) was true and at error sql_reset_result was called. there the crash happened here: if(res->vals) { for(i=0; i<res->nrows; i++) { for(j=0; j<res->ncols; j++) { if(res->vals[i][j].flags&PV_VAL_STR && res->vals[i][j].value.s.len>0) /* here */ looks to me that it should be tested if res->vals[i] is not null before executing the if statement. -- juha ps. it there any chance to find out from the core dump, why memory run out, i.e., how was most of it used? amount of pkg memory in this host was defined to be 8 MB. -- juha

2 2

dialog module with DB Backend
by jay binks 06 Mar '14

06 Mar '14

Hi all, im using the dialog module with db_cassandra backend.. I dont believe this issue is related to cassandra, but its worth mentioning anyways. so... I run kamailio, make calls, see dialogs in the DB.. and I Can use "kamctl mi dlg_list" and see that dialogs go away when I hangup a call.. When I query the DB Backend, I still see the queries, but they have a state of 5. I Initially thought this was a bug, but it seems dialogs in state 5 get cleaned up after a period. so I moved on. now , lets restart kamailio.. kamailio loads all dialogs on startup, after kamailio starts I call "kamctl mi dlg_list" again, and it shows all my dialogs from the DB. they DO show as "State 5" but for some reason, these dialogs appear to stick around for a long time, and the bigger issue it causes me is that my channel limiting ( using get_profile_size ) seems to consider these dialogs ( in state 5 ) as being active calls. Please someone point me in the right direction... :) what am I doing wrong ? ( or is this a bug somewhere ) Sincerely Jay

3 12

geoip crash
by Juha Heinanen 06 Mar '14

06 Mar '14

i made a typo in my script and $gip(src=>cc) was referenced before geoip_match() call. the reference resulted in crash. to reproduce, execute statement $var(test) = $gip(src=>cc); as first thing in your script. -- juha

2 17

Releasing 4.1.2, 4.0.6 and 3.3.7
by Daniel-Constantin Mierla 06 Mar '14

06 Mar '14

Hello, we are considering to release new versions from stable branches of Kamailio starting with Thursday, March 6. First is v4.1.2 from latest stable branch and if the time allows in the same day, the v4.0.6. The reason to release v3.3.7 is to have a proper debian packages built and hosted on the new repositories -- the specs had a conflict for latest 3.3 version of some sub-packages. If there are known issues that were not addressed, please report them to the tracker. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, in Berlin, Germany http://www.kamailioworld.com

3 4

How to add some text at end of body line with subst_body
by Barthel Marco (CI/AFU1) 05 Mar '14

05 Mar '14

Hi, I'm trying to add some string at the end of some line in the body/sdp of a SIP INVITE for testing some interoperability. The following line a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NwC8z5MsyCWbpJJo1n2BDS8DkGuUU2cIt9KRUqPU should be converted to a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NwC8z5MsyCWbpJJo1n2BDS8DkGuUU2cIt9KRUqPU UNENCRYPTED_SRTCP I tried with this: subst_body('/(a=crypto.*)\r/\1 UNENCRYPTED_SRTCP\r/g'); but it does not match just using subst_body('/(a=crypto.*)/\1 UNENCRYPTED_SRTCP/g'); matches but results in this (carriage return between key and "UNENCRYPTED_SRTCP"): a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NwC8z5MsyCWbpJJo1n2BDS8DkGuUU2cIt9KRUqPU\rUNENCRYPTED_SRTCP using Kamailio 3.3.6 Any ideas? Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) Postfach 30 02 20 70442 Stuttgart GERMANY www.bosch.com<http://www.bosch.com> Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Franz Fehrenbach; Geschäftsführung: Dr. Volkmar Denner, Dr. Stefan Asenkerschbaumer, Dr. Rolf Bulander, Dr. Stefan Hartung, Dr. Dirk Hoheisel, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Dr. Werner Struth, Peter Tyroller

2 1

msilo config
by Peter Villeneuve 05 Mar '14

05 Mar '14

Hi, Can someone share a working msilo config for 4.1.1? I've tried following older tutorials out there but they no longer seem to apply. I just want to test out msilo with the default kamailio config in 4.1.1, but the current example in http://kamailio.org/docs/modules/4.1.x/modules/msilo.html doesn't work for me. Specifically, I get an error regarding the syntax in t_on_failure("1"); Any help or pointers appreciated. Thanks

2 5

Incorrect Contact address
by Marc Soda 05 Mar '14

05 Mar '14

I have Kamailio setup as a proxy in front of a backend server (Asterisk). When I make a call through the proxy, the Contact header in the 200 OK that is returned to the client has the IP of the backend server in it. Thus, the client is sending it's ACK directly to the backend server. Is there a special method to rewrite the Contact header to be Kamailio's IP? Where is a good place in the config to do this? (my config is loosely based on this: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb) Thanks! Marc

4 5

Re: [SR-Users] [Kamailio-Business] [sr-dev] New official Debian and Ubuntu repository
by Vik Killa 05 Mar '14

05 Mar '14

Can someone update the wiki page? I was confused for days on this: http://www.kamailio.org/dokuwiki/doku.php/packages:debs On Wed, Mar 5, 2014 at 8:21 AM, Vik Killa <vipkilla(a)gmail.com> wrote: > Can someone update the wiki page? I was confused for days on this: > http://www.kamailio.org/dokuwiki/doku.php/packages:debs > > > On Tue, Mar 4, 2014 at 6:54 AM, Klaus Darilion < > klaus.mailinglists(a)pernau.at> wrote: > >> Thanks to all involved >> >> >> On 03.03.2014 14:16, Victor Seva wrote: >> >>> The new build system for Debian and Ubuntu packages is now in place. >>> This service is kindly sponsored by SipWise [0] thanks to Andreas >>> Granig [1]. Sipwise is providing the hosting and man power to create >>> and manage this new system. >>> >>> deb.kamailio.org is based on jenkins-debian-glue[2] project running on >>> AWS EC2 environment thanks to Michael Prokop [3] and myself. All the >>> needed files, scripts and info to reproduce this system is kept public >>> at [4]. >>> >>> In [5] is described all the repositories available. >>> >>> * nightly builds are been built if a change is detected in the branch, >>> once by night. >>> - kamailiodev-nightly >>> branch: >>> - 'master' >>> distributions: jessie, wheezy, squeeze, precise >>> >>> - kamailio41-nightly >>> branch: >>> - '4.1' >>> distributions: jessie, wheezy, squeeze, precise >>> >>> - kamailio40-nightly >>> branch: >>> - '4.0' >>> distributions: lenny, squeeze, wheezy, lucid, precise >>> >>> - kamailio33-nightly >>> branch: >>> - '3.3' >>> distributions: lenny, squeeze, wheezy, lucid, precise >>> >>> * tags are been built if a new tag is detected once by night. >>> - kamailio41 >>> branch: >>> - '*/tags/4.1*' >>> distributions: jessie, wheezy, squeeze, precise >>> >>> - kamailio40 >>> branch: >>> - '*/tags/4.0*' >>> distributions: lenny, squeeze, wheezy, lucid, precise >>> >>> - kamailio33 >>> branch: >>> - '*/tags/3.3*' >>> distributions: lenny, squeeze, wheezy, lucid, precise >>> >>> Now there is no 4.1.1 package built so the default kamailio repository >>> is pointing to a empty repository. As soon as we release 4.1.2 this >>> will be fixed. >>> >>> Regards, >>> Victor Seva ( kamailio admin hat on ) >>> >>> [0] http://www.sipwise.com/ >>> [1] http://www.kamailio.org/w/andreas-granig/ >>> [2] https://github.com/mika/jenkins-debian-glue >>> [3] http://michael-prokop.at/ >>> [4] https://github.com/sipwise/kamailio-deb-jenkins >>> [5] http://deb.kamailio.org/index.html >>> >>> _______________________________________________ >>> sr-dev mailing list >>> sr-dev(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>> >>> >> _______________________________________________ >> Kamailio (OpenSER) - Business mailing list >> Business(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/business >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/business >> > >

3 3

Reload memory leaks issues.
by Uri Shacked 05 Mar '14

05 Mar '14

Hi, kamailio ver 3.3.6. I load approximately 100Mb of data into memory. And my shared memory size is 4Gb. It is CARRIERROUTE, DIALPLAN, MTREE and HTABLE combimations. My system has to be synced with DB that is being updated frequently. So, I reload the data every 1-2 minutes. I notice that the shared memory usage is increasing at 1% every 18-19 reloads. I use mem_join = 1 and no mem_join at all. I compiled with C_DEFS+= -DMEM_JOIN_FREE and even with -DF_MALLOC. Tried different options, still, the issue keeps on going and it is very frustrating... Anyone has the same problem? BR, Uri

1 1

MSRP Relay
by Nandini madhu 05 Mar '14

05 Mar '14

Dear All, Can anybody please tel me, are there any MSRP relay (RFC 4976) compliant SIP clients for Android Platform ? Sorry for asking this question which is out of relevant here. Waiting for somebody's comments on this Regards, Nandini

1 0

Kamailio World 2014 - Exhibitors
by Daniel-Constantin Mierla 05 Mar '14

05 Mar '14

During Kamailio World 2014, the companies IT Center, Digium, Sipgate, NG Voice, Asipto and FhG Fokus will provide all day long demos and presentations of their products and services. You can contact them in advance to schedule an appointment if you want to reserve proper time to discuss about their offerings: - http://conference.kamailio.com/k02/exhibitors/ It is less than one month till the start of the event, the agenda is pretty much finalized: - http://conference.kamailio.com/k02/schedule/ It has over 30 sessions presented by world wide speakers, from technical tutorials to business use cases, with topics such WebRTC, IMS, anti-fraud & VoIP security, scalability with OpenStack, real time charging, various Kamailio use cases, Asterisk, FreeSwitch and SEMS. Once again we are grateful to event sponsors -- Asipto, FhG Fokus, Sipwise, IT Center, Sipgate, Simwood, Digium, NG Voice -- without their contributions is impossible to bring such an excellent group of speakers and support event logistics: - http://conference.kamailio.com/k02/sponsors/ Hurry up if you don't want to miss this event, it is no alternative in Europe for meeting professionals in large real time communications deployments and open source telephony. Registration details are available at: - http://conference.kamailio.com/k02/registration/ See you in Berlin! -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

invalid xmlrpc response from pkg.stats?
by Juha Heinanen 05 Mar '14

05 Mar '14

i tried make xmlrpc query on pkg.stats, but my xmlrpc lib was not able to parse the response (below). according to xmlrpc spec: The body of the response is a single XML structure, a <methodResponse>, which can contain a single <params> which contains a single <param> which contains a single <value>. according to the spec <value> is scalar value or <struct> or <array>. in sip proxy response below, there a several <struct>s in the <value>, which according to my understanding is violation of the spec. instead, the response should be like this: <?xml version="1.0"?> <methodResponse> <params> <param> <value> <array> <data> <value> <struct>...</struct> </value> <value> <struct>...</struct> </value> ... </data> </array> </param> </methodResponse> comments? -- juha -------------------------------------------------------------------------- Received the following response: HTTP/1.0 200 OK Via: SIP/2.0/TCP 127.0.0.1:36643 Server: OpenXg SIP Proxy (4.1.1-2 (i386/linux)) Content-Length: 6311 <?xml version="1.0"?> <methodResponse> <params> <param> <value><struct><member><name>entry</name><value><int>0</int></value></member><member><name>pid</name><value><int>5846</int></value></member><member><name>rank</name><value><int>0</int></value></member><member><name>used</name><value><int>4553004</int></value></member><member><name>free</name><value><int>559936</int></value></member><member><name>real_used</name><value><int>5731520</int></value></member></struct> <struct><member><name>entry</name><value><int>1</int></value></member><member><name>pid</name><value><int>5848</int></value></member><member><name>rank</name><value><int>1</int></value></member><member><name>used</name><value><int>4557328</int></value></member><member><name>free</name><value><int>554652</int></value></member><member><name>real_used</name><value><int>5736804</int></value></member></struct> <struct><member><name>entry</name><value><int>2</int></value></member><member><name>pid</name><value><int>5849</int></value></member><member><name>rank</name><value><int>2</int></value></member><member><name>used</name><value><int>4557328</int></value></member><member><name>free</name><value><int>554652</int></value></member><member><name>real_used</name><value><int>5736804</int></value></member></struct> <struct><member><name>entry</name><value><int>3</int></value></member><member><name>pid</name><value><int>5850</int></value></member><member><name>rank</name><value><int>3</int></value></member><member><name>used</name><value><int>4557328</int></value></member><member><name>free</name><value><int>554652</int></value></member><member><name>real_used</name><value><int>5736804</int></value></member></struct> <struct><member><name>entry</name><value><int>4</int></value></member><member><name>pid</name><value><int>5851</int></value></member><member><name>rank</name><value><int>4</int></value></member><member><name>used</name><value><int>4557328</int></value></member><member><name>free</name><value><int>554652</int></value></member><member><name>real_used</name><value><int>5736804</int></value></member></struct> <struct><member><name>entry</name><value><int>5</int></value></member><member><name>pid</name><value><int>5852</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4556356</int></value></member><member><name>free</name><value><int>556296</int></value></member><member><name>real_used</name><value><int>5735160</int></value></member></struct> <struct><member><name>entry</name><value><int>6</int></value></member><member><name>pid</name><value><int>5853</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4556356</int></value></member><member><name>free</name><value><int>556296</int></value></member><member><name>real_used</name><value><int>5735160</int></value></member></struct> <struct><member><name>entry</name><value><int>7</int></value></member><member><name>pid</name><value><int>5854</int></value></member><member><name>rank</name><value><int>4294967294</int></value></member><member><name>used</name><value><int>4563012</int></value></member><member><name>free</name><value><int>549112</int></value></member><member><name>real_used</name><value><int>5742344</int></value></member></struct> <struct><member><name>entry</name><value><int>8</int></value></member><member><name>pid</name><value><int>5857</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4556368</int></value></member><member><name>free</name><value><int>556236</int></value></member><member><name>real_used</name><value><int>5735220</int></value></member></struct> <struct><member><name>entry</name><value><int>9</int></value></member><member><name>pid</name><value><int>5863</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4556368</int></value></member><member><name>free</name><value><int>556236</int></value></member><member><name>real_used</name><value><int>5735220</int></value></member></struct> <struct><member><name>entry</name><value><int>10</int></value></member><member><name>pid</name><value><int>5873</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4557616</int></value></member><member><name>free</name><value><int>554844</int></value></member><member><name>real_used</name><value><int>5736612</int></value></member></struct> <struct><member><name>entry</name><value><int>11</int></value></member><member><name>pid</name><value><int>5876</int></value></member><member><name>rank</name><value><int>5</int></value></member><member><name>used</name><value><int>4642360</int></value></member><member><name>free</name><value><int>467988</int></value></member><member><name>real_used</name><value><int>5823468</int></value></member></struct> <struct><member><name>entry</name><value><int>12</int></value></member><member><name>pid</name><value><int>5877</int></value></member><member><name>rank</name><value><int>6</int></value></member><member><name>used</name><value><int>4618200</int></value></member><member><name>free</name><value><int>493396</int></value></member><member><name>real_used</name><value><int>5798060</int></value></member></struct> <struct><member><name>entry</name><value><int>13</int></value></member><member><name>pid</name><value><int>5878</int></value></member><member><name>rank</name><value><int>7</int></value></member><member><name>used</name><value><int>4618200</int></value></member><member><name>free</name><value><int>493396</int></value></member><member><name>real_used</name><value><int>5798060</int></value></member></struct> <struct><member><name>entry</name><value><int>14</int></value></member><member><name>pid</name><value><int>5884</int></value></member><member><name>rank</name><value><int>4294967292</int></value></member><member><name>used</name><value><int>4035956</int></value></member><member><name>free</name><value><int>497228</int></value></member><member><name>real_used</name><value><int>5307880</int></value></member></struct> </value> </param> </params> </methodResponse>

2 3

INVITE proxy auth
by Marc Soda 04 Mar '14

04 Mar '14

Hey all, I have a pretty general SIP question that I'm hoping some of you can shed some light on. I hope this ok for the list. I am setting up a SIP proxy with Kamailio. The backend server (Asterisk in my case) requires authentication. Is it standard/best practice to require a proxy to authenticate to a backend server for INVITEs? I am already registering, with authentication, on behalf of the client, i.e, the client registers to the proxy, then the proxy registers to the backend server, all with authentication. I was hoping not to have to do this on INVITEs, but if I don't I'm left with the following: client proxy backend | -------INVITE-------> | | | <--------407--------- | | | ---------ACK--------> | | | ----INVITE(auth)----> | | | <--------100--------- | | | | -------INVITE-------> | | | <--------401--------- | | | ---------ACK--------> | | <--------401--------- | | | ---------ACK--------> | | | ----INVITE(auth)----> | | | <--------100--------- | | | | ----INVITE(auth)----> | | | <--------100--------- | | | <---------OK--------- | | <---------OK--------- | | | ---------ACK--------> | | | | ---------ACK--------> | It works, but, it's terrible... Before I try to make it work differently, what do you all think it should do? Marc

2 1

New official Debian and Ubuntu repository
by Victor Seva 04 Mar '14

04 Mar '14

The new build system for Debian and Ubuntu packages is now in place. This service is kindly sponsored by SipWise [0] thanks to Andreas Granig [1]. Sipwise is providing the hosting and man power to create and manage this new system. deb.kamailio.org is based on jenkins-debian-glue[2] project running on AWS EC2 environment thanks to Michael Prokop [3] and myself. All the needed files, scripts and info to reproduce this system is kept public at [4]. In [5] is described all the repositories available. * nightly builds are been built if a change is detected in the branch, once by night. - kamailiodev-nightly branch: - 'master' distributions: jessie, wheezy, squeeze, precise - kamailio41-nightly branch: - '4.1' distributions: jessie, wheezy, squeeze, precise - kamailio40-nightly branch: - '4.0' distributions: lenny, squeeze, wheezy, lucid, precise - kamailio33-nightly branch: - '3.3' distributions: lenny, squeeze, wheezy, lucid, precise * tags are been built if a new tag is detected once by night. - kamailio41 branch: - '*/tags/4.1*' distributions: jessie, wheezy, squeeze, precise - kamailio40 branch: - '*/tags/4.0*' distributions: lenny, squeeze, wheezy, lucid, precise - kamailio33 branch: - '*/tags/3.3*' distributions: lenny, squeeze, wheezy, lucid, precise Now there is no 4.1.1 package built so the default kamailio repository is pointing to a empty repository. As soon as we release 4.1.2 this will be fixed. Regards, Victor Seva ( kamailio admin hat on ) [0] http://www.sipwise.com/ [1] http://www.kamailio.org/w/andreas-granig/ [2] https://github.com/mika/jenkins-debian-glue [3] http://michael-prokop.at/ [4] https://github.com/sipwise/kamailio-deb-jenkins [5] http://deb.kamailio.org/index.html

4 3

Re: [SR-Users] Regd. Stored procedure support in Kamailio
by Alex Balashov 04 Mar '14

04 Mar '14

You can executed any SQL statements you want through 'sqlops', including invocation of stored procedures. Our Kamailio-based Class 4 routing/trunking product is built almost entirely on Postgres stored procedures. We have dozens of them. As to whether they are "really needed", I don't know how to answer that. :-) In our case, we decided they were really needed because they offer the opportunity to execute complex business logic beyond what Kamailio's relatively primitive route scripting language can offer, while eliminating unnecessary back-and-forth data interchange between Kamailio and the database. Stored procedures and triggers also execute in a transactional manner, which is additionally beneficial. Shankar <shankar.rk(a)plintron.com> wrote: >Hello All, > > > >Does Kamailio support invoking stored procedures instead of sending >plain DB >queries already? > > > >We have been asked to migrate to stored procedures. I am trying to >estimate >the effort involved in migrating Kamailio code to support stored >procedure. > > > >Other question that comes to my mind is whether stored procedures are >really >needed in Kamailio. Can someone please clarify? > > > >Regards, > >Shankar > > > > > >------------------------------------------------------------------------ > >_______________________________________________ >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 -- Sent from my mobile, and thus lacking in the refinement one might expect from a fully fledged keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com

1 0

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