- sr-users - kamailio.org

FreeSWITCH at FOSDEM, SIP and WebRTC Load Balancing and High Availability
by Giovanni Maruzzelli 03 Feb '17

03 Feb '17

See you all, in person or in the intertube, at: https://fosdem.org/2017/schedule/event/freeswitch/ sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT

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Websocket TLS Issue
by Jade SZ 03 Feb '17

03 Feb '17

Hi Guys, I am trying to setup the following flow: Browser >> WSS >> HA Proxy >>> WSS >> Kamailio But getting TLS errors in Kamailio logs: *[29634]: ERROR: <core> [tcp_read.c:1321]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f68ebe872b0 r: 0x7f68ebe87330* *[29631]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number* Browser <-----wss---->Kamailio works fine with same certs. Both HA Proxy and Kamilio are installed on separate servers, hosting on same port with different domain. Kamailio tls.conf has method = TLSv1 *@HA Proxy:* openssl s_client -connect HA-PROXY-DOMAIN:*10443* SSL-Session: Protocol : TLSv1.2 *@Kamailio :* openssl s_client -connect KAMAILIO-DOMAIN:*10443* SSL-Session: Protocol : TLSv1 So I made HA Proxy to be on TLSv1 "ssl-default-bind-options force-tlsv10" But still I get the same TLS error in Kamailio. *HA Proxy config looks like:* *frontend public* * bind *:10443 ssl crt /etc/haproxy/certs/cert.pem* * acl is_websocket hdr_end(host) -i m1.some-domain.com <http://m1.some-domain.com>* * use_backend wss if is_websocket* * default_backend wss* *backend wss* * timeout server 600s* * server ws1 k1.some-domain.com:10443 <http://k1.some-domain.com:10443>* * server ws1 k2.some-domain.com:10443 <http://k2.some-domain.com:10443>* Need some direction, thanks in advance. Regards, Jade

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id column i tables
by Kristian F. Høgh 02 Feb '17

02 Feb '17

Hi list, We provision our kamailio database from an external system. In the future we would like to provision from two different systems. To prevent the two systems from touching each others data, we would like to create views from the two tables. provision system 1 would write to table1 provision system 2 would write to table2 kamailio will read a view over the two tables. The tables we will create views over are: - address - aliases - dispatcher - uri - usr_preferences Now to my question :-) Does kamailio use the id field in any of the tables above? The id's are auto-incremented, and we might get conflicts between table1 and table2 *Kristian Høgh* Telefon: 4422 8822 support(a)uni-tel.dk[1] Gydevang 19 | 3450 Allerød www.uni-tel.dk [2] -------- [1] mailto:support@uni-tel.dk [2] http://www.uni-tel.dk

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Disabling DNS SRV records checks
by Ahmed Fouad 01 Feb '17

01 Feb '17

Hi, I'm running kamailio version 4.1 and having the following DNS related configuration use_dns_failover = off dns_try_naptr = no dns_udp_pref = -1 dns_try_ipv6 = off dns_retr_time=1 dns_retr_no=1 dns_servers_no=1 dns_use_search_list=no But still am seeing SRV record checks for my domains which I like to completely disable these checks and only use A records only. x.x.x.x.57828 > 8.8.8.8.53: 7417+ SRV? _sip._udp.subdomain.com 8.8.8.8.53 > x.x.x.x.57828: 7417 NXDomain 0/1/0 (129) I can't find a DNS option to disable SRV checks. Can you please advise ?

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could not make ul_rm_contact work
by Aqs Younas 01 Feb '17

01 Feb '17

Greetings list, I am trying to remove a specific contact from memory but always resulted in vain. root@debian:/usr/local/kamailio/sbin# ./kamctl ul show Domain:: location table=1024 records=1 max_slot=1 AOR:: 1040(a)192.168.0.101 Contact:: sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp Q= Expires:: 2416 Callid:: _3595KDc1PinhUGFpSf4Dg.. Cseq:: 2 User-agent:: Z 3.9.32144 r32121 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:192.168.0.101:5060 Methods:: 5087 Ruid:: uloc-58911db0-f45-1 Reg-Id:: 0 Last-Keepalive:: 1485905402 Last-Modified:: 1485905402 . /kamctl fifo ul_rm_contact location 1040(a)191.168.0.101 sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp 404 AOR not found Tried another command root@debian:/usr/local/kamailio/sbin# ./kamctl fifo ul_rm location 1040(a)192.168.0.101 root@debian:/usr/local/kamailio/sbin# ./kamctl ul show Domain:: location table=1024 records=1 max_slot=1 AOR:: 1040(a)192.168.0.101 Contact:: sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp Q= Expires:: deleted Callid:: _3595KDc1PinhUGFpSf4Dg.. Cseq:: 2 User-agent:: Z 3.9.32144 r32121 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:192.168.0.101:5060 Methods:: 5087 Ruid:: uloc-58911db0-f45-1 Reg-Id:: 0 Last-Keepalive:: 1485905402 Last-Modified:: 1485905402 And this. . /kamctl ul rm 1040(a)192.168.0.1 sip:1040@192.168.0.102:36619 ;rinstance=1dfaed088d873a18;transport=tcp 404 AOR not found Can someone help me know to remove a contact from memory? Any pointer is much appreciated. Best Regards.

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sdp_with_transport
by M. Salman 01 Feb '17

01 Feb '17

Hi Guys, Just wanted to clarify the following case: what should be result of sdp_with_transport("RTP/SAVPF") on line: m=audio 10231 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 I am having a weird behavior for two different versions of Kamailio. I hope I am doing something wrong. kamailio 4.3.2 => sdp_with_transport("RTP/SAVPF") = true kamailio 4.4.5 => sdp_with_transport("RTP/SAVPF") = false -- Regards M. Salman VoIP Professional

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Re: [SR-Users] sr-users Digest, Vol 140, Issue 30
by Andrew Miller 01 Feb '17

01 Feb '17

Is it too late to request a place for the meal at FOSDEM? I would love to come. Andy Miller On 30 Jan 2017 11:00, <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. Diversion Header - relay messed up (Roman Dissauer) > 2. Re: dlg_set_timeout bye to websockets client (Switch168) > 3. Presence Module do not load db_url (Vu. Minh Cao) > 4. Re: FOSDEM 2017 (Giacomo Vacca) > 5. Re: Replace old registration record with new one while > keeping single AOR per contact (Federico Cabiddu) > 6. Re: Replace old registration record with new one while > keeping single AOR per contact (Aqs Younas) > 7. Re: mhomed opened sockets (Diego Nadares) > 8. Re: [sr-dev] FOSDEM 2017 (Daniel-Constantin Mierla) > 9. Re: FOSDEM 2017 (Ivan Todorov) > 10. Re: [Kamailio-Business] [sr-dev] FOSDEM 2017 (Alexandr Dubovikov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 29 Jan 2017 18:42:23 +0100 > From: Roman Dissauer <roman(a)dissauer.net> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] Diversion Header - relay messed up > Message-ID: <DB3FC45E-ACCE-426E-B662-031909F6BA40(a)dissauer.net> > Content-Type: text/plain; charset=utf-8 > > When I get an INVITE with Diversion Header the Request is forwarded > without Diversion Header and the Request User is taken from Diversion User. > Problem is that on the Destination Host I cannot get original Request User > what is the intended destination! > Is this intended behaviour? How can I change this behaviour? > > I already tried to delete the Diversion Header on request_route but this > didn’t change the behaviour. > > Thanks in andvance for your help! > Roman > > Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as > Carrier IP > > Incoming INVITE: > > INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. > Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. > From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. > To: <sip:436761234567@x.x.x.x>. > Call-ID: 906143813_44460603(a)y.y.y.y. > CSeq: 28100 INVITE. > Max-Forwards: 19. > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, > NOTIFY,PRACK,UPDATE,OPTIONS. > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed. > Contact: <sip:+43987654321@y.y.y.y:5060>. > P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. > Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; > reason=unconditional; counter=1. > Supported: timer,100rel. > Session-Expires: 1800. > Min-SE: 90. > Content-Length: 260. > Content-Disposition: session; handling=required. > Content-Type: application/sdp. > ... > > > Relayed INVITE: > > INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. > Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. > Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. > Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582. > 4fc0216dbecafde29127db502993222c.0. > Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch= > z9hG4bK02B7871714d9345f843. > From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. > To: <sip:436761234567@x.x.x.x>. > Call-ID: 906143813_44460603(a)y.y.y.y. > CSeq: 28100 INVITE. > Max-Forwards: 18. > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, > NOTIFY,PRACK,UPDATE,OPTIONS. > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed. > Contact: <sip:+43987654321@y.y.y.y:5060>. > P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. > Supported: timer,100rel. > Session-Expires: 1800. > Min-SE: 90. > Content-Length: 274. > Content-Disposition: session; handling=required. > ... > > > > > > > > ------------------------------ > > Message: 2 > Date: Sun, 29 Jan 2017 18:21:37 +0000 > From: Switch168 <team(a)switch168.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org>, miconda(a)gmail.com > Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client > Message-ID: > <CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh > 7z+jRAEg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Daniel, > > > For anyone else who might run into this issue I found > https://github.com/kamailio/kamailio/issues/85 to be related to mine. And > its a good a starting point. > > Cheers > Andrew > > > > > On Fri, Jan 27, 2017 at 10:47 AM Switch168 <team(a)switch168.com> wrote: > > > HI Daniel, > > > > modparam("dialog", "dlg_flag", 4) modparam("dialog", "send_bye", 1) > > modparam("dialog", "timeout_noreset", 1) > > > > So regular bye's by useragent i can succesfully relay to this > > <sip:6gjlmali@ec2c66jsa0ei.invalid;transport=ws> > > style of uri by using this snippet below > > > > # Handle requests within SIP dialogs > > route[WITHINDLG] { > > if(is_method("BYE")) { > > xlog("DEBUG: Received BYE"); > > route(NATDETECT); > > loose_route(); > > dlg_manage(); > > t_check_trans(); > > handle_ruri_alias(); > > rtpengine_delete(); > > route(RELAY); > > exit; > > } > > ... > > > > But the byes that are sent out by dlg_set_timeout("$var(timer)") gives > out > > error cannot resolve > > the random.uri and I don't know how to fix the bye before it get sent > out. > > > > Thanks > > Andrew > > > > > > > > > > > > > > > > > > On Fri, Jan 27, 2017 at 12:15 AM, Daniel-Constantin Mierla < > > miconda(a)gmail.com> wrote: > > > > Hello, > > > > isn't the dialog module setting the right value there? What function are > > you using to update the contact? > > > > Cheers, > > Daniel > > > > On 27/01/2017 06:01, Andrew Tan wrote: > > > > Hello, > > > > Just wondering if it's possible to intercept the bye message that > > dlg_set_timeout sends out to do some nat_helper function to fix it so the > > bye can be sent to random.invalid uris. > > > > I know there is the edge proxy outbound module but wondering if I can do > > it without. > > > > Regular bye between 2 clients i was able intercept and fix with nat > helper > > but I dont know how to fix the byes that is sent out from dlg_set_timeout > > function. > > > > Thanks in advance! > > Andrew > > > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > listsr-users@lists.sip-router.orghttp://lists.sip-router. > org/cgi-bin/mailman/listinfo/sr-users > > > > > > -- > > Daniel-Constantin Mierlawww.twitter.com/miconda -- > www.linkedin.com/in/miconda > > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - > www.asipto.com > > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > >

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Re: [SR-Users] FOSDEM 2017
by Hugh Waite 01 Feb '17

01 Feb '17

Hi all, Please count me and one other in for the meal on Saturday. See you in Brussels! Hugh Waite On 30 Jan 2017 11:00 a.m., <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. Diversion Header - relay messed up (Roman Dissauer) 2. Re: dlg_set_timeout bye to websockets client (Switch168) 3. Presence Module do not load db_url (Vu. Minh Cao) 4. Re: FOSDEM 2017 (Giacomo Vacca) 5. Re: Replace old registration record with new one while keeping single AOR per contact (Federico Cabiddu) 6. Re: Replace old registration record with new one while keeping single AOR per contact (Aqs Younas) 7. Re: mhomed opened sockets (Diego Nadares) 8. Re: [sr-dev] FOSDEM 2017 (Daniel-Constantin Mierla) 9. Re: FOSDEM 2017 (Ivan Todorov) 10. Re: [Kamailio-Business] [sr-dev] FOSDEM 2017 (Alexandr Dubovikov) ---------------------------------------------------------------------- Message: 1 Date: Sun, 29 Jan 2017 18:42:23 +0100 From: Roman Dissauer <roman(a)dissauer.net> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Diversion Header - relay messed up Message-ID: <DB3FC45E-ACCE-426E-B662-031909F6BA40(a)dissauer.net> Content-Type: text/plain; charset=utf-8 When I get an INVITE with Diversion Header the Request is forwarded without Diversion Header and the Request User is taken from Diversion User. Problem is that on the Destination Host I cannot get original Request User what is the intended destination! Is this intended behaviour? How can I change this behaviour? I already tried to delete the Diversion Header on request_route but this didn’t change the behaviour. Thanks in andvance for your help! Roman Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as Carrier IP Incoming INVITE: INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 19. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; reason=unconditional; counter=1. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 260. Content-Disposition: session; handling=required. Content-Type: application/sdp. ... Relayed INVITE: INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582. 4fc0216dbecafde29127db502993222c.0. Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 18. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 274. Content-Disposition: session; handling=required. ... ------------------------------ Message: 2 Date: Sun, 29 Jan 2017 18:21:37 +0000 From: Switch168 <team(a)switch168.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>, miconda(a)gmail.com Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client Message-ID: <CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh7z+jRAEg(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi Daniel, For anyone else who might run into this issue I found https://github.com/kamailio/kamailio/issues/85 to be related to mine. And its a good a starting point. Cheers Andrew On Fri, Jan 27, 2017 at 10:47 AM Switch168 <team(a)switch168.com> wrote: > HI Daniel, > > modparam("dialog", "dlg_flag", 4) modparam("dialog", "send_bye", 1) > modparam("dialog", "timeout_noreset", 1) > > So regular bye's by useragent i can succesfully relay to this > <sip:6gjlmali@ec2c66jsa0ei.invalid;transport=ws> > style of uri by using this snippet below > > # Handle requests within SIP dialogs > route[WITHINDLG] { > if(is_method("BYE")) { > xlog("DEBUG: Received BYE"); > route(NATDETECT); > loose_route(); > dlg_manage(); > t_check_trans(); > handle_ruri_alias(); > rtpengine_delete(); > route(RELAY); > exit; > } > ... > > But the byes that are sent out by dlg_set_timeout("$var(timer)") gives out > error cannot resolve > the random.uri and I don't know how to fix the bye before it get sent out. > > Thanks > Andrew > > > > > > > > > On Fri, Jan 27, 2017 at 12:15 AM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > > Hello, > > isn't the dialog module setting the right value there? What function are > you using to update the contact? > > Cheers, > Daniel > > On 27/01/2017 06:01, Andrew Tan wrote: > > Hello, > > Just wondering if it's possible to intercept the bye message that > dlg_set_timeout sends out to do some nat_helper function to fix it so the > bye can be sent to random.invalid uris. > > I know there is the edge proxy outbound module but wondering if I can do > it without. > > Regular bye between 2 clients i was able intercept and fix with nat helper > but I dont know how to fix the byes that is sent out from dlg_set_timeout > function. > > Thanks in advance! > Andrew > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router. org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >

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any public databases or schemas for dial plans?
by Daniel Pocock 31 Jan '17

31 Jan '17

Hi, Are there any public databases of dial plan information? Are there any schemas (e.g. XML / XSD) useful for describing arbitrary dial plans from different carriers around the world? The type of things that I'm interested in: - being able to translate any local number to E.164 (if possible) - knowing the expected length (or range of lengths) for different numbers - being able to recognize numbers that can't be translated to E.164 (e.g. the emergency numbers) - being able to classify specific types of number that usually have a carrier-specific or country-specific meaning (e.g. classifying numbers as directory assistance, customer service, emergency) I'm familiar with Google's libphonenumber, although it is not a database, rather, it is a library that has dial plan logic in Java. It supports many of the things I want for numbers that can be translated to E.164 and country-level dial plans, but not local dial plans (e.g. dialing a London number without the 020 area code). Regards, Daniel

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Using DNS for Service Discovery with Kamailio
by Casco, Alessio 31 Jan '17

31 Jan '17

Hello Guys! I'd like to be able to use SRV records inside dispatcher groups, so I can automatise the discovery of new backends on my network using DNS I've tried with the following formats into the dispatcher.list file: $group _sip._udp.sip-voice_backends $group sip-voice_backends and none looks to be working, running the kamailio_ctl dispatcher.list won't show me the group set with SRV records. ( I also try NAPTR records, with no luck as well.) * Does the dispatcher module support SRV or NAPTR records? (maybe my format is not correct) * If not, do you have any other idea on what I could do to achieve what I'd like to do? Thanks Alessio

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RFC: Kamailio tutorials/cookbooks/docs - wiki or markdown
by Daniel-Constantin Mierla 31 Jan '17

31 Jan '17

Hello, I want to start a discussion about some of the kamailio tutorials from the wiki, mainly those related to core cookbook, variables and transformations. They are now in dokuwiki at: - https://www.kamailio.org/wiki/#cookbooks The wiki served us pretty well so far, however the contributions there are not that active. From time to time we still get spammers, even we added restriction that only registered and authenticated users can write. I am thinking that maybe people don't like creating yet another account just add some example or rephrase for clarity. On the other hand, there are over 400 registered users, many of them I expect to be either made because if misunderstanding the purpose or by spammers that guessed the (sip) captcha (plenty of random usernames). Besides that I think the wiki adds overhead when releasing a new version as each time we have to copy and paste old version content to a new set of pages and update the version number. With the above in mind, I wonder if won't be better to store those tutorials in markdown format and use mkdocs.org or gitbook (https://github.com/GitbookIO/gitbook) tools to generate the html and host it in kamailio.org. The wiki will still be used to index them. Among the benefits I see: * store in github.com (a new repo, like: kamailio-docs) and all devs can contribute directly, other users can make pull requests * create branches for each major version of kamailio and backport across them with git whenever is applicable * the edit can be done directly via github.com website (similar to the wiki right now) and even read it directly from github in html transformed from markdown by github * one can get them offline by cloning the git repo, use them or enhance when offline Many if the tutorials will stay in the wiki, but I think those that are related to documenting per kamailio version should be migrated to gihub storage + markdown. If there is a strong pro opinion for this change, I can try to convert one of the cookbooks very soon so people can feel better the difference. If we get to a consensus, then I think we can have those tutorials in the new format for v5.0. Should anyone have comments, other suggestions or improvements, let's put them on debate. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

2 7

serial forking calls using TSILO
by Tomas Zanet 30 Jan '17

30 Jan '17

Hello, I'm using Kamailio 4.4.4 with TSILO module in order to support Push Notifications used by our voip app on Apple ios10. So far, everything works fine: Kamailio can get an incoming call, suspend it, then send push notification with an external script, receive a new app registration and then call it after resuming the invite. To do so, thanks to Mr, Cabiddu, I used all the functions described here http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-… Right now, I would like to support serial forking calls using push notification because it was already supported with legacy voip. To do so, I used failure_route function where, after getting the call destinations and setting them into an avp, I set the new SIP request uri $ru. Here is my failure_route: failure_route[MANAGE_FAILURE] route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) t_reply("404","Not found"); exit; #!endif if (is_avp_set("$avp(group_members_db)")) $ru ="sip:"+$avp(group_members_db)+"@"+$fd; $avp(group_members_db) = $null; route(LOCATION); exit; I am troubleshooting this scenario and this is what I see: 1) Kamailio receives incoming call 2) Suspend it 3) Send Push notification to account1 4) Kamailio receives account1 registration -> INVRESUME route is now called 5) Then call account1 6) Nobody answers the call 7) failure_route[MANAGE_FAILURE] is now called -> set $ru with next sipaccount: account2 8) suspend invite 9) send push notification to account2 10) account2 sends its sip registration -> INVRESUME route is now called 11) Kamailio calls account1 instead of account2 -> this is my issue On step 10: despite account2 is registered, I checked it using "kamctl ul show", the second call is forwarded all'account1 and not to account2. I also checked the TSILO logs and everything seems ok. Here is the logs: First call: suspended transaction [53945:1648394094] asterisk => account1 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account1 53945:1648394094) second call: suspended transaction [53945:1648394094] asterisk => account2 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account2 53945:1648394094) In order to find out the issue I put some xlogs, printing $ru value: what I see is the $ru value is set correctly on failure_route but as soon as the t_continue is called, the ru overwritten back to account1 Can anyone address me to find out the solution? Thanks in advance T. Tomas Zanet Software Design Department tzanet(a)came.com

3 7

Re: [SR-Users] FOSDEM 2017
by DanB 30 Jan '17

30 Jan '17

Hey Daniel, if anything to be reserved, please count us also in. Will come back regarding number of seats but just to be aware that we are interested to join the dinner. Thanks! DanB

8 7

mhomed opened sockets
by Diego Nadares 30 Jan '17

30 Jan '17

Hi Guys, We have two network interfaces. One public and one private. We had enabled mhomed and works great BUT if we do a netstat -naup | grep 5060 we see that some sockets keeps opened also when the call is finished. The source port in netstat it's not 5060 so I suppose that kamailio use another one for something else. Is this the right behavior? How long does this sockets remain opened? Are we missing something in our cfg? 6 2017-01-25 15:52:04.986651 172.16.213.38 172.16.200.159 SIP/SDP 146 Request: INVITE sip:xxx@172.16.200.159:5060 34 2017-01-25 15:52:19.636895 172.16.213.38 172.16.200.159 SIP 716 Request: ACK sip:xxx@172.16.200.159:5060 | 37 2017-01-25 15:54:38.947831 172.16.208.111 172.16.213.38 SIP 736 Request: BYE sip:127.0.0.8:5060;line=sr-987jhlk* | 40 2017-01-25 15:54:38.950899 172.16.213.38 172.16.200.159 SIP 660 Request: BYE sip:172.16.208.111:5060 | [root@dwrfsd01 kamailio]# netstat -naup | grep 5060 udp 0 0 172.16.213.38:*56086* <http://172.16.213.38:56086> 172.16.200.159:5060 ESTABLISHED 17015/kamailio udp 0 0 172.16.213.38:5060 0.0.0.0:* 17006/kamailio udp 0 0 172.16.213.38:*41971* <http://172.16.213.38:41971> 172.16.200.159:5060 ESTABLISHED 17051/kamailio udp 0 0 172.16.213.38:*54651* <http://172.16.213.38:54651> 172.16.200.159:5060 ESTABLISHED 17008/kamailio Thanks in advance! Diego.

2 3

Presence Module do not load db_url
by Vu. Minh Cao 29 Jan '17

29 Jan '17

Hi, I got error when loading presense module in Kamailio 4.4: [Configure] loadmodule "presence.so" modparam("presence", "db_url", "mysql://kamailio:kamailiorw@127.0.0.1/kamailio") [Error] ERROR: presence [presence.c:608]: fixup_presence(): Bad config - you can not call 'handle_publish' function (db_url not set) Which is really strange since according to the source code this error just occurs if “db_url” is not set. I already checked my database also, I can access mysql with about username/password. Yours sincerely, — Vu

1 0

rtpengine - Key file error
by Serhat Guler 29 Jan '17

29 Jan '17

Hello, I am trying to setup my Kamailio environment on a new Debian system. Everything went well, except I started facing a problem with the rtpengine. It did install it fine and I can view the program parameters via the help menu; however, when I run it with this command "rtpengine --interface 192.168.0.66 --listen-ng=127.0.0.1:22222 -m 30000 -M 35000 -L 2" or any other commands, it gives an error as "CRIT: Fatal error: Bad command line: Key file does not have group 'rtpengine'". Where is this key file it is complaining about ? I couldn't find anything neither in the doc nor anywhere else. Cheers, Serhat

2 2

missing memory after tls cleanup
by Andy 28 Jan '17

28 Jan '17

Hi, the tls total memory before and after a tls connection is cleaned up does not match, during test there were 10 tls connection created for which I see tls_h_close and tls_h_tcpconn_clean were called exactly 10 times for the same connection ids, bit before and after the tls shm memory does not match, using command: kamcmd mod.stats tls shm could there be a leak? please suggest thanks! -- View this message in context: http://sip-router.1086192.n5.nabble.com/missing-memory-after-tls-cleanup-tp… Sent from the Users mailing list archive at Nabble.com.

2 4

Ansible Kamailio role
by Alberto Llamas 27 Jan '17

27 Jan '17

Hello Kamailians, I just wanted to share an Ansible role for the installation and compilation of a Kamailio server. It may required improvements and I will be more than happy to heard them. *https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/ <https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/>* Cheers, -- Alberto Llamas Telecommunications Engineer dCAA|dCAP|KPAC|SSCA

4 4

Re: [SR-Users] FOSDEM 2017
by DanB 26 Jan '17

26 Jan '17

Hey Daniel, Would be there 2 persons for sure, so please count us also in. Thanks, DanB

2 1

parallel media forking during ringing phase
by Tomas Zanet 26 Jan '17

26 Jan '17

Hi all, I'm writing to the community because I need a hint from Kamailio and voip gurus: the goal is adding parallel forking call with early-media to our PABX where there's a Kamailio instance running. I mean media streams should be sent to all call destinations before answering the call order, for instance, to see the incoming video. As far as I know, from signaling side should not be a problem because, using Kamailio, there’s append_branch function which forks many branches to all users. The problem is related to media streams because there’s no way to fork media to all users after 180 + SDP or 183, right? Obviously all users must negotiate the same media because media transcoding is basically impossible, especially for video streams. I think there are two ways to do that: 1) using Kamailio and Asterisk. Asterisk has Queue function with ringall strategy, but I’m not sure if Asterisk supports multi domain scenarios. Have ever used Asterisk with Kamailio in multi domain scenarios? 2) enhancing rttpproxy-ng. I don't really know the effort to achieve that. In your opinion, what’s the easiest way to achieve that? Thanks, Best Regards. Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

2 2

How does Kamailio decide which protocol to use when fwding to another proxy?
by Pranathi Venkatayogi 25 Jan '17

25 Jan '17

Hi, I have two instances of Kamailio acting as edge proxies. One on the customer side and one on the agent side. Like: customer -> proxy1 -> proxy2 -> agent. Both customer and agent are registered to proxy1/proxy2 via TLS. However when proxy1 forwards to proxy2, it is using UDP. How can I force it to use TLS? Attached is the result of nslookup on the domain: translation.sms-test.cyracom.com. Thanks Pranathi [cid:image001.jpg@01D27631.42773770] From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Ryan Wagoner Sent: Tuesday, January 24, 2017 8:26 AM To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Asterisk Proxy Multiple Devices / BLF Issues I'm following the latest Kamailio and Asterisk Realtime guide to offload registrations from my FreePBX / Asterisk setup and possibly load balance down the road. I'm running Kamailio 4.4.5 and Asterisk 11.6-cert15. I realize FreePBX isn't realtime and will work around that with a database view, etc. I was excited to see Kamailio will handle multiple devices registering to the same device/extension and placing / receiving calls works. I did run into an issue when any device unregisters Kamailio always forwards the register with expires 0 to Asterisk. To workaround this I modified the route[REGFWD] and added the if($hdr(Expires)==$null) chunk of code. I wanted to use caller->count, but ran into stale contact records with expires set to deleted. I then tried enumerating the contacts, but don't understand why ulc(caller->expires) is 10 when kamctl ul show shows expires deleted. The code below works, but I was hoping for an explanation of the expires = 10 or if there was a better way to handle this scenario. Additionally I enabled presence (WITH_PRESENCE) but Kamailio responds 489 bad event for subscribe requests from devices registered to it. I was hoping it would proxy these to Asterisk for BLF support. If somebody could point me in the right direction it would be appreciated. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($hdr(Expires)==$null) { reg_fetch_contacts("location", "$sel(contact.uri)", "caller"); $var(i) = 0; $var(j) = 0; while($var(i) < $(ulc(caller=>count))) { if($(ulc(caller=>expires)[$var(i)])!=10) { $var(j) = $var(j) + 1; } $var(i) = $var(i) + 1; } if($var(j)>=1) { 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); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } Thanks, Ryan

2 3

kamctl cr reload got 500 failed to re-built tree
by Igor Potjevlesch 25 Jan '17

25 Jan '17

Hello, With more than 173k entriers into carrierroute, now I'm doing "kamctl cr reload", I got the following reply: 500 failed to re-built tree, see log What could going wrong? Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 4

Issue with From address being modified in ACK when UAC module used version 4.1
by Jonathan Hunter 25 Jan '17

25 Jan '17

Hi Guys, running kamailio 4.1.4 and using uac_replace_from, I am seeing a strange issue with the proxying of an ACK message back from a carrier to freeswitch on the ingress path into a network. So its just a normal call inbound, where on outbound leg we modify the From address, on the inbound leg all remains the same. Now after the ingress side receives the 200ok, it sends an ACK as below; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+44792498881474@192.168.24.8:5080;user=phone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Route: <sip:192.168.24.8;lr=on;ftag=carrier.peering.telecom.im+3+863d20a3+1b9801d2;vsf=AAAAAAAAAAYNBgAPAAgLAgZ3UTMACgBXFUsVFwwcDkAKTwMZGh0YAA8KDkEEQ1NYC0NDXgdOFRpSHg1vbmU-> Content-Length: 0 Max-Forwards: 68 However kamailio changes the From address; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 109.73.69.165:5060;branch=z9hG4bKc1ce.47974fc3da2b669a78f2dcc9a057a127.0 Via: SIP/2.0/UDP 4.4.4.4:5060;rport=5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+441444680332@es132y$}-9>.8n?~9,*%(;zyk393;7e&C^NRone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Content-Length: 0 Max-Forwards: 67 Causing FreeSWITCH to not recognise the request, and therefore not send an ACK. There are no rules set against the ACK processing. Has anyone seen this before? We dont know when it started happening which doesnt help, I will look to setup debug on test environment but just wondered if this is an issue thats been seen before? Many thanks in advance. Jon

2 5

Event when acc is written?
by Tobias 25 Jan '17

25 Jan '17

Hi! I'm wondering if there is a way for the acc-module to notify/send an event to Kamailio (so I can pick it up from the config, similar to event_route) once the acc-module has performed a write? As an example I'm using acc and db-mysql to store acc data in MySQL, but instead of having a service polling the MySQL for new acc's written I'd like to make Kamailio send an event once an acc-row has been written. I realize this could most likely be done by using MySQL triggers "ON INSERT", but I'm wondering if there is another (reliable) way to do this directly from within Kamailio? I haven't tested it, but I would guess that if I tried to use for example event_route[dialog:end] there could be times where the event would hit Kamailio before the acc-module have written to missed_calls? Thanks! /Tobias

4 11

Executing perl script with command line parameters.
by Venkatesh Macha 25 Jan '17

25 Jan '17

Hi all, I am trying to execute external Perl script from the Kamailio config file. My script accepts few command line parameters but if I try to pass command line parameters like below, It is not working. exec_msg("/root/test.pl 10000 '$var(env)' '$fU'"); # i tried with exec_avp also How to execute the external Perl scripts with few arguments. Thank you in Advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Executing-perl-script-with-command-… Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] How does Kamailio decide which protocol to use when fwding to another proxy?
by Pranathi Venkatayogi 25 Jan '17

25 Jan '17

In the case below though TLS session is established between proxy1 and proxy2, invite is fwded on UDP. 1. How to force Kamailio to use existing tls session to communicate? (proxy1) 2. What is the command to look at if existing tls sessions are present from proxy1 a. kamcmd tls.list does it, but I need one which works from within config. From: Pranathi Venkatayogi Sent: Tuesday, January 24, 2017 11:02 AM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Subject: How does Kamailio decide which protocol to use when fwding to another proxy? Hi, I have two instances of Kamailio acting as edge proxies. One on the customer side and one on the agent side. Like: customer -> proxy1 -> proxy2 -> agent. Both customer and agent are registered to proxy1/proxy2 via TLS. However when proxy1 forwards to proxy2, it is using UDP. How can I force it to use TLS? Attached is the result of nslookup on the domain: translation.sms-test.cyracom.com. Thanks Pranathi [cid:image002.jpg@01D27654.0A627200]

2 1

kamdbctl error.
by Venkatesh Macha 25 Jan '17

25 Jan '17

Hi all, I am using debian jessie, I just installed kamailio 4.4 from source and trying to create kamailio database using kamdbctl but when i try to run the kamdbctl create i am getting following charset error. /usr/local/sbin/kamdbctl create MySQL password for root: INFO: creating database kamailio ... ERROR 1115 (42000) at line 1: Unknown character set: 'ISO' Mysql Version : 5.5.52 Thank you in advance -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamdbctl-error-tp155288.html Sent from the Users mailing list archive at Nabble.com.

2 5

Asterisk Proxy Multiple Devices / BLF Issues
by Ryan Wagoner 24 Jan '17

24 Jan '17

I'm following the latest Kamailio and Asterisk Realtime guide to offload registrations from my FreePBX / Asterisk setup and possibly load balance down the road. I'm running Kamailio 4.4.5 and Asterisk 11.6-cert15. I realize FreePBX isn't realtime and will work around that with a database view, etc. I was excited to see Kamailio will handle multiple devices registering to the same device/extension and placing / receiving calls works. I did run into an issue when any device unregisters Kamailio always forwards the register with expires 0 to Asterisk. To workaround this I modified the route[REGFWD] and added the if($hdr(Expires)==$null) chunk of code. I wanted to use caller->count, but ran into stale contact records with expires set to deleted. I then tried enumerating the contacts, but don't understand why ulc(caller->expires) is 10 when kamctl ul show shows expires deleted. The code below works, but I was hoping for an explanation of the expires = 10 or if there was a better way to handle this scenario. Additionally I enabled presence (WITH_PRESENCE) but Kamailio responds 489 bad event for subscribe requests from devices registered to it. I was hoping it would proxy these to Asterisk for BLF support. If somebody could point me in the right direction it would be appreciated. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($hdr(Expires)==$null) { reg_fetch_contacts("location", "$sel(contact.uri)", "caller"); $var(i) = 0; $var(j) = 0; while($var(i) < $(ulc(caller=>count))) { if($(ulc(caller=>expires)[$var(i)])!=10) { $var(j) = $var(j) + 1; } $var(i) = $var(i) + 1; } if($var(j)>=1) { 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); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } Thanks, Ryan

1 0

(no subject)
by jay binks 24 Jan '17

24 Jan '17

Hi Guys, Im using a keepalived solution between 2 kamailio boxes, with net.ipv4.ip_nonlocal_bind =1 . On the "non master" I pretty much constantly get these logs in syslog. What Im looking for is a way to suppress these because they are expected in my configuration. is there any way to tell kamailio to not log this... ( or at least, not at CRIT and ERROR ) *Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: ERROR: tm [../../forward.h:208]: msg_send(): udp_send failed* *Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: ERROR: <core> [udp_server.c:576]: udp_send(): sendto(sock,0x7fc0732d5ab0,380,0,xxx.xxx.xxx.xxx:5060,16): Invalid argument(22)* *Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: CRITICAL: <core> [udp_server.c:581]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net* -- Sincerely Jay

4 9

How to determine correct port to set in record-route header
by Pranathi Venkatayogi 23 Jan '17

23 Jan '17

Hi, I am using Kamailio behind NAT, unable to figure how to make it put “public ip” in Record-route header, I am manually inserting the hard-coded header myself as below. However now I am having trouble choosing the right port number in all scenarios. $var(dstPort) = 5061; if (dst_port==5060) { $var(dstPort) = 5060; } insert_hf("Record-Route: <sip:MY_PUBLICIP_ADDR:$var(dstPort);transport=tls;lr;nat=yes>\r\n", "Record-Route"); In one scenario I see the conflicting port numbers in “via” header vs “Record route”. The ACK is being set to port 5060 based on Record route header and is not being received by the callee. 2017-01-23 14:52:49.970233 [blink.exe 2780]: SENDING: Packet 11, +0:00:36.411867 10.0.27.108:58217 -(SIP over TLS)-> 172.31.211.31:5061 SIP/2.0 180 Ringing Via: SIP/2.0/TLS 63.149.103.72:5061;received=172.31.211.31;branch=z9hG4bK4799.7067bd2a48063748a4c353fa408eaefa.0 Via: SIP/2.0/UDP 10.0.16.52;rport=5060;branch=z9hG4bK4799.935a27c6ef5ad112225964cbe7c1be44.0;i=2 Via: SIP/2.0/TLS 10.11.200.12:51793;rport=51793;received=10.11.200.12;branch=z9hG4bKPj49f6aeca3f9b4155ab4c5304b544aa4d;alias Record-Route: <sip:63.149.103.72:5060;transport=tls;lr;nat=yes> Record-Route: <sip:10.0.16.52:5061;transport=tls;lr;nat=yes> Call-ID: 09ce10efa6a946bf9445ccc21857393e From: "cust1" <sip:cust1@devtranslation.sms-test.cyracom.com>;tag=95082caabecc42548c2fec5ccd29e5de To: <sip:spanish@translation.sms-test.cyracom.com>;tag=ec365cc8489b48a7bc5725d21b7d97a1 CSeq: 17422 INVITE Server: Blink 3.0.0 (Windows) Contact: <sip:20745891@10.0.27.108:58216;transport=tls> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Content-Length: 0 Questions: 1. How can I let in-built record-route function automatically choose “public ip” so I can get rid of my manual insertion altogether. 2. If no option, what is the right way to choose the port on which packet is being received, so it is same as what is on “VIA”. 3. Any other pointers to improve the logic here?

1 0

NAPTR/SRV lookups fail when use_dns_failover is enabled.
by Jeremy Gadd 23 Jan '17

23 Jan '17

Hello, everyone! I'm trying to enable dns_srv_lb but I'm stuck just before I can get there. If I don't enable use_dns_failover, lookups work and calls route but always to the first host. We need to load balance, so that doesn't work very well. If I enable use_dns_failover, the lookups stop working completely and we get a dns lookup failure like so: ERROR: tm [ut.h:296]: uri2dst2(): ERROR: uri2dst: failed to resolve " redacted.hostname.here.com" :unresolvable A or AAAA request (-7) This works: use_dns_cache=on dns_use_search_list=no dns_try_naptr=on # use_dns_failover=on This breaks it: use_dns_cache=on dns_use_search_list=no dns_try_naptr=on use_dns_failover=on I have checked to see that use_dns_failover is compiled in: version: kamailio 4.4.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2

2 4

"Named Telephone Event" (I think) problem
by Pete Kelly 23 Jan '17

23 Jan '17

Hi I am running Kamailio 4.4 and have come across a strange SBC which is sending DTMF within the SDP body of a NOTIFY with a Content-Type audio/telephone-event. Has anybody come across this before? The content itself comes through on a tcpdump/ngrep trace as four dots "...." and Kamailio puts an empty string into the logs if I try to log the contents of $rb, however Wireshark manages to decode it as something readable "\005\000\a\320" and usable. (essentially the digit 5) Is there any way I can get access to this decoded version within Kamailio at all? Even if I have to pass it out to an external program? Many thanks for any help anyone can offer! Pete

2 1

Best way to log incoming/outgoing sip messages?
by Pranathi Venkatayogi 23 Jan '17

23 Jan '17

Hi, What is the best way to get a complete dump of incoming/outgoing SIP messages of Kamailio server? I encountered “SIPTrace” module, but it writes to database. Is there a way we can write to syslog instead? Thanks Pranathi

5 4

Kamailio IMS Secure Websocket closes connection
by Serhat Guler 22 Jan '17

22 Jan '17

Dear all, I am trying to make a call between 2 sipml5 client over WSS. Everything works fine and as expected when using WS. While using WSS, the clients are registered successfully, but when I make a call, the called party is disconnected from the network and the caller gets "User offline" message. There a wireshark capture here: https://drive.google.com/file/d/0B6LMw8kmoAMYMGhrYXNUVXJySEk/view?usp=shari… The call is made from Alice to Bob, the port Nrs: 4060: PCSCF, 5060:ICSCF, and 6060 SCSCF. WWS is running on 4443. The error message that is shown in Firefox console: "The connection to wss://192.168.0.11:4443/ was interrupted while the page was loading." Thanks in advance, Serhat

1 0

Get ip from contact on redirect
by Diego Nadares 20 Jan '17

20 Jan '17

Hi Guys, Anybody knows how to get the ip of the new contact of the redirect? I need it to update one xavp variable that contains the INVITE destination. 6 2017-01-20 12:44:55.138473 172.16.213.38 *172.16.208.121* SIP/SDP 85 Request: INVITE sip:01152780776@172.16.208.121:5060 | 9 2017-01-20 12:44:55.142296 172.16.208.121 172.16.213.38 SIP 661 Status: 302 Moved temporarily | 10 2017-01-20 12:44:55.156268 172.16.213.38 172.16.208.121 SIP 423 Request: ACK sip:01152780776@172.16.208.121:5060 | 16 2017-01-20 12:44:55.293349 172.16.213.38 *172.16.208.111* SIP/SDP 95 Request: INVITE sip:1152780776@172.16.208.111:5060;user=phone | My cfg: # Manage failure routing cases failure_route[MANAGE_FAILURE] { .... if (t_check_status("302")) { xlog("Got a 302 - redirecting"); get_redirects("*:1"); * $xavp(ip_b[$avp(current_route)]=>name) = "172.16.208.111";#Hardcoded for testing* xlog("Got a 302 - redirecting $hdr(Contact)"); route(RELAY); #t_relay(); } Thanks in advance. Diego.

2 3

Siremis Multi domain
by Annus Fictus 20 Jan '17

20 Jan '17

Hi, I think Siremis not work correctly in a multi domain scenario if the extension number are the same for two domains. I explain a little more. I have two extension: 1 200(a)sip1.domain.org 2 200(a)sip2.domain.org On the Siremis I create the user 200 with the SIPuser role. When I login with this user, and go to subscriber data i can see bots extensions. I think is relate to this select on kamailio database: .....SELECT T0.id, T0.username, T0.domain, T0.password, T0.email_address, T0.ha1, T0.ha1b, T0.rpid FROM `subscriber` T0 WHERE T0.username='200' LIMIT 10 Any hint? Regards

2 7

UAC authentication
by anfecora 20 Jan '17

20 Jan '17

can someone please guide me where to find the usage of auc module, thanks in advance. I am registering the uac in one end then i want to forward that registration to an external registrar, if i use $du some domain, some of the ipv6 info is going to be forwarded, i need to make it look more like an a b2bua than a outbound proxy therefore i need to use the auc module(also i need to cover all ipv6 info since i am proxying from ipv4 to ipv6). i am able to register the client to kamailio, then kamailio send the registration to the external registrar, but when the registrar sends back the challenge uac does not reply, any ideas how can achieve this. uac(linphone etc) --registration--->kamailio(registrar module)------register(uac)--->SBC sbc----invite---->kamailio(lookup)---------invite---->uac(linphone etc) this is the code i am using. # Handle SIP registrations route[REGISTRAR] { xlog("REGISTERING "); if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if(af==INET6) setbflag(FLB_IPV6); xlog("IT IS IPV6"); if (!save("location")) sl_reply_error(); route(REGSBCFWD); exit; } } route[REGSBCFWD]{ xlog("CREATING OTHER LEG TO THE SBC"); if(!is_method("REGISTER")) { return; } xlog("CREATING PACKAGE $si -- $fU-- $au--$du-- $tU --$ru"); $var(rip) = "sip.somedomain.com"; $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + "5060"; $uac_req(furi)="sip:" + $tU + "@" + $var(rip); $uac_req(turi)="sip:" + $tU + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $tU + "@" + "10.0.0.222" + ":" "5060" + ">\r\n"; if($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(); } REGISTER sip:sip.somedomain.com:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.222:5060 ;branch=z9hG4bK285f.13a7f0c0000000000000000000000000.0. To: <sip:14052423@sip.somedomain.com>. From: <sip:14052423@sip.somedomain.com >;tag=a82eaf71666096790d1397845fbd3254-ebaa. CSeq: 10 REGISTER. Call-ID: 357e21ae51d8416d-26605(a)10.0.0.222. Max-Forwards: 70. Content-Length: 0. User-Agent: kamailio (4.4.4 (x86_64/linux)). Contact: <sip:140524@10.0.0.222:5060>. Expires: 3600. . SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.0.0.222:5060 ;branch=z9hG4bK4b53.ad842757000000000000000000000000.0. To: <sip:14052423@sip.somedomain.com>. From: <sip:14052423@sip.somedomain.com >;tag=a82eaf71666096790d1397845fbd3254-37da. Call-ID: 357e21ae51d8416d-26603(a)10.0.0.222. CSeq: 10 REGISTER. WWW-Authenticate: Digest realm="sip.somedomain.com", nonce="f5831435-3e69-4b78-ab6d-dcb0897e477b", algorithm=MD5, qop="auth". User-Agent: somedomain.com Host:some proxy. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Content-Length: 0. .

2 3

Return body/string from Perl library
by David Cunningham 20 Jan '17

20 Jan '17

Hello, We are using Kamailio's perl module and want our custom library to set the body on a SIP message. Is it possible to set a SIP message body from within Perl, or can we return a string to the Kamailio configuration to then use with set_body()? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019

2 3

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 19 Jan '17

19 Jan '17

root@homer02:~# netstat -altp Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 0 0 homer02.xxxxx.xxx:sip *:* LISTEN 16018/kamailio tcp 0 0 homer02.xxxxx.xxx:5066 *:* LISTEN 16018/kamailio Thanks. JR > If you use netstat, what is in the recv queue for tcp packets on sip ports? > > netstat -altp > > Cheers, > Daniel > > > On 18/01/2017 16:43, JR Richardson wrote: >> Yes, this is a sipcapture node. I'm listening on a switch port that is >> set to mirror my VoIP traffic. I see all SIP UDP/TCP packets on the >> mirror port and the Ethernet port of the host node. Just don't see any >> TCP packets process in kamailio, debug 3. UDP packets are processed as >> expected. >> >> My config is using port mirror for the capture parameters: >> >> modparam("sipcapture", "capture_on", 1) >> modparam("sipcapture", "hep_capture_on", 0) >> modparam("sipcapture", "raw_ipip_capture_on", 0) >> modparam("sipcapture", "raw_moni_capture_on", 1) >> >> modparam("sipcapture", "raw_sock_children", 4) >> modparam("sipcapture", "raw_interface", "eth1") >> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") >> modparam("sipcapture", "promiscious_on", 1) >> modparam("sipcapture", "raw_moni_bpf_on", 1) >> >> Is there a method I could diagnose if the SIP TCP Packets are getting >> from the kernel network process and the kamailio process? >> >> # ngrep -d eth1 -W byline host x.x.x.x | /var/run/kamailio/kamailio.pid >> >> Or pipe to kamailio local unix socket? >> >> I don't know, I'm just guessing. >> >> Thanks. >> >> JR >> >> >>> Somehow is not clear for me how you have the configuration there ... >>> before commenting further, this needs to be clarified. >>> >>> The node you presented the config is a sipcapture instance, right? What >>> is sending traffic to it? Is another kamailio with siptrace module? Or >>> the sipcature agent? Or you have a port mirroring in the router? >>> >>> Cheers, >>> Daniel >>> >>> >>> On 17/01/2017 16:37, JR Richardson wrote: >>>>> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >>>>>> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >>>>>> using raw socket capture, I think this may be a sipcapture issue, >>>>>> debuging kamailio shows normal startup and processing of UDP SIP >>>>>> packets, but does not show any activity with TCP packets. >>>>> I never used HOMER sofar but when I saw your first message my thoughts >>>>> was that this can't work in a simple way since for TCP you need to >>>>> complete a 4 way handshake before you can start to send data. >>>>> >>>> Interesting. Are you referring to handshaking on the network stack or >>>> SIP TCP TLS handshaking? I guess I can see it two ways. >>>> >>>> 1) if your talking about TCP/IP handshake, even though the SIP packet >>>> comes into the mirror port on the host node, the kernel processing the >>>> TCP packet is not establishing a valid connection due to no TCP >>>> handshake because its only a monitor port, no transmit back, then the >>>> kernel network stack does not pass the SIP TCP packet to the kamailio >>>> process for capture because it drops the packet due to no valid >>>> handshake? >>>> >>>> 2) the kernel network stack is passing the SIP TCP packet to the >>>> kamailio process, but since kamailio cannot handshake back it drops >>>> the packet and does not process through the sipcapture module. This >>>> kinda breaks the whole capture ability for homer with SIP TCP. Using >>>> ngrep, I see all SIP TCP packets, invite -->, trying <--, session >>>> progress <--, request timeout <--, ack -->, etc... >>>> >>>> So how would I diagnose if the network stack is the culprit? Debugging >>>> kamailio is pretty straight forward, setup and listening for SIP TCP, >>>> but never see any processing of any TCP packets. >>>> >>>> Thanks. >>>> >>>> JR >

2 1

Phone number formats per country
by Daniel-Constantin Mierla 19 Jan '17

19 Jan '17

Hello, slightly off-topic, but related to voip -- does anyone have a link that can quickly share to a site with up to date details about phone numbering plans for as many country as possible? Ideally to include the toll and premium number prefixes, the split between mobile and fixed lines if it is the case. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

4 6

ds_is_from_list and NAPTR / SRV
by Björn Bylander 19 Jan '17

19 Jan '17

Hello, I’m trying to resolve an issue with the ds_is_from_list function from the dispatcher module. I’m calling it as follows: ds_is_from_list(”42”, "3", "sip:sip.foo.bar”) This makes Kamailio log an error: ERROR: dispatcher [dispatch.c:2327]: ds_is_addr_from_list(): could not resolve sip.foo.bar sip.foo.bar doesn’t have any A records but there are NAPTR and SRV records: $ host -t naptr sip.foo.bar sip.foo.bar has NAPTR record 20 0 "s" "SIP+D2T" "" _sip._tcp.sip.foo.bar. sip.foo.bar has NAPTR record 10 0 "s" "SIP+D2U" "" _sip._udp.sip.foo.bar. $ host -t srv _sip._tcp.sip.foo.bar _sip._tcp.sip.foo.bar has SRV record 10 0 5060 sip1.foo.bar. _sip._tcp.sip.foo.bar has SRV record 10 0 5060 sip2.foo.bar. $ host -t srv _sip._udp.sip.foo.bar _sip._udp.sip.foo.bar has SRV record 10 0 5060 sip2.foo.bar. _sip._udp.sip.foo.bar has SRV record 10 0 5060 sip1.foo.bar. $ host sip1.foo.bar. sip1.foo.bar has address 10.47.11.1 $ host sip2.foo.bar. sip2.foo.bar has address 10.47.11.2 The following DNS options are set: dns_srv_lb = yes dns_try_naptr = yes dns_naptr_ignore_rfc = no The Kamailio version is 4.2.8 (installed from the Debian binary package) on Debian Wheezy. Shouldn’t ds_is_from_list be able to resolve the NAPTR and SRV records? Can I change anything to be able to use ds_is_from_list with the specified domain? Thanks, Björn Bylander

2 1

Upgrade from 4.3 to 4.4
by Laura 18 Jan '17

18 Jan '17

Dear list, I'm planning the upgrade fromn 4.3.6 tree to 4.4.x version on Debian distribution. Can i run the apt-get update and upgrade without problem.. or is possible that i will found some error during upgrade ? Is the db schema upgraded or not from 4.3 up to 4.4 ? Regards Laura

2 1

Please explain : t_relay() vs t_relay_to_udp()
by Abdoul Osséni 18 Jan '17

18 Jan '17

hello all, I want to use kamailio like as a proxy SIP (outbound proxy) to transcode tls to udp like us: call flow is: Softphone --> tls --> Kamailio server (kamailio listen on 443 port) --> udp --> Asterisk server (asterisk listen on 5060 port) I know Kamailio can convert tls to udp. According tm documentation http://kamailio.org/docs/modules/devel/modules/tm.html, I think I must use t_relay_to_udp(). Do you confirm I must replace t_relay() function by t_relay_to_udp() function? Thank for your help. Regards Abdoul.

2 2

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 18 Jan '17

18 Jan '17

Yes, this is a sipcapture node. I'm listening on a switch port that is set to mirror my VoIP traffic. I see all SIP UDP/TCP packets on the mirror port and the Ethernet port of the host node. Just don't see any TCP packets process in kamailio, debug 3. UDP packets are processed as expected. My config is using port mirror for the capture parameters: modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 0) modparam("sipcapture", "raw_ipip_capture_on", 0) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "raw_interface", "eth1") modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") modparam("sipcapture", "promiscious_on", 1) modparam("sipcapture", "raw_moni_bpf_on", 1) Is there a method I could diagnose if the SIP TCP Packets are getting from the kernel network process and the kamailio process? # ngrep -d eth1 -W byline host x.x.x.x | /var/run/kamailio/kamailio.pid Or pipe to kamailio local unix socket? I don't know, I'm just guessing. Thanks. JR > Somehow is not clear for me how you have the configuration there ... > before commenting further, this needs to be clarified. > > The node you presented the config is a sipcapture instance, right? What > is sending traffic to it? Is another kamailio with siptrace module? Or > the sipcature agent? Or you have a port mirroring in the router? > > Cheers, > Daniel > > > On 17/01/2017 16:37, JR Richardson wrote: >>> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >>>> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >>>> using raw socket capture, I think this may be a sipcapture issue, >>>> debuging kamailio shows normal startup and processing of UDP SIP >>>> packets, but does not show any activity with TCP packets. >>> I never used HOMER sofar but when I saw your first message my thoughts >>> was that this can't work in a simple way since for TCP you need to >>> complete a 4 way handshake before you can start to send data. >>> >> Interesting. Are you referring to handshaking on the network stack or >> SIP TCP TLS handshaking? I guess I can see it two ways. >> >> 1) if your talking about TCP/IP handshake, even though the SIP packet >> comes into the mirror port on the host node, the kernel processing the >> TCP packet is not establishing a valid connection due to no TCP >> handshake because its only a monitor port, no transmit back, then the >> kernel network stack does not pass the SIP TCP packet to the kamailio >> process for capture because it drops the packet due to no valid >> handshake? >> >> 2) the kernel network stack is passing the SIP TCP packet to the >> kamailio process, but since kamailio cannot handshake back it drops >> the packet and does not process through the sipcapture module. This >> kinda breaks the whole capture ability for homer with SIP TCP. Using >> ngrep, I see all SIP TCP packets, invite -->, trying <--, session >> progress <--, request timeout <--, ack -->, etc... >> >> So how would I diagnose if the network stack is the culprit? Debugging >> kamailio is pretty straight forward, setup and listening for SIP TCP, >> but never see any processing of any TCP packets. >> >> Thanks. >> >> JR > -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

Kamailio v4.4.5 Released
by Daniel-Constantin Mierla 18 Jan '17

18 Jan '17

Hello, Kamailio SIP Server v4.4.5 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.5. For more details about version 4.4.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/01/kamailio-v4-4-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Planning the release of v4.4.5
by Daniel-Constantin Mierla 18 Jan '17

18 Jan '17

Hello, I am considering to release v4.4.5 with latest fixes from branch 4.4 later this week, on Wednesday or Thursday (Jan 19 or 20). As usual, if anyone has fixes that were not backported yet, do it if you are a developer or write to sr-dev to be sure it is not missed. Sumbit also the new issues you are ware of and they are not on the bug tracker. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Record-Route Header Question
by Andres 17 Jan '17

17 Jan '17

Hi, I am troubleshooting some strange call setup problems using an Edgewater Networks SBC. A couple of Polycom phones sit behind the SBC and they connect to a SIP Server in the cloud. The SBC is in passthrough mode. What I see that I am not sure about is: INVITE from SIP Server to SBC IP (no Record-Route header...so far no surprise here) INVITE from SBC to Phone (Record-Route header inserted by SBC but with the IP of the SIP Server) I do not think the SBC is supposed to add a Record-Route header with the IP of the remote SIP Server. I think It should only be allowed to add Record-Route headers with its own IP. Does that sound right? The RFC lists the usage as: "20.30 Record-Route The Record-Route header field is inserted by proxies in a request to force future requests in the dialog to be routed through the proxy." It says nothing about being able to arbitrarily force future requests to go through a remote IP. Please advise. Thanks, -- Technical Support

2 2

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 17 Jan '17

17 Jan '17

> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >> using raw socket capture, I think this may be a sipcapture issue, >> debuging kamailio shows normal startup and processing of UDP SIP >> packets, but does not show any activity with TCP packets. > > I never used HOMER sofar but when I saw your first message my thoughts > was that this can't work in a simple way since for TCP you need to > complete a 4 way handshake before you can start to send data. > Interesting. Are you referring to handshaking on the network stack or SIP TCP TLS handshaking? I guess I can see it two ways. 1) if your talking about TCP/IP handshake, even though the SIP packet comes into the mirror port on the host node, the kernel processing the TCP packet is not establishing a valid connection due to no TCP handshake because its only a monitor port, no transmit back, then the kernel network stack does not pass the SIP TCP packet to the kamailio process for capture because it drops the packet due to no valid handshake? 2) the kernel network stack is passing the SIP TCP packet to the kamailio process, but since kamailio cannot handshake back it drops the packet and does not process through the sipcapture module. This kinda breaks the whole capture ability for homer with SIP TCP. Using ngrep, I see all SIP TCP packets, invite -->, trying <--, session progress <--, request timeout <--, ack -->, etc... So how would I diagnose if the network stack is the culprit? Debugging kamailio is pretty straight forward, setup and listening for SIP TCP, but never see any processing of any TCP packets. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

Using the auth_ephemeral module
by Hemanshu Vadehra 17 Jan '17

17 Jan '17

I'm involved in setting up a Kamailio instance and was hoping to make use of the auth_ephemeral module for authentication. But the module documentation doesn't quite make clear how exactly the module is to be employed or the web service set up. Does anyone have a working example? Regards, Hemanshu Vadehra hemanshu.v(a)directi.com

5 8

usrloc module with dbmode=2 does not insert records to database
by Vik Killa 17 Jan '17

17 Jan '17

Hello, we've noticed that the usrloc module does not "sync" all the records from memory into the database. I use a bash script to generate in-memory AoRs ( http://paste.debian.net/plain/908521) then i perform kamcmd ul.flush and no records are inserted. We have tried various usrloc parameters but none seem to work Here is our basic setup # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "timer_interval", 120) modparam("usrloc", "timer_procs", 4) We are using postgresql. are we missing something? Thanks /V

2 11

Info: github template for reporting issues
by Daniel-Constantin Mierla 17 Jan '17

17 Jan '17

Hello, yesterday I added a template for opening new issues on github.com bug tracker. You can see it in action at: - https://github.com/kamailio/kamailio/issues/new Its bare text can be seen at: - https://raw.githubusercontent.com/kamailio/kamailio/master/.github/ISSUE_TE… More about the issue templates for github can be found at: - https://help.github.com/articles/creating-an-issue-template-for-your-reposi… Beside informing about this addition, I would like to get also feedback about it. Do you find it useful or you think it's going to be a hassle? If it should be kept, is anything that you think it should be added or removed? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

LCR module dont_strip_or_tag_flag
by Annus Fictus 17 Jan '17

17 Jan '17

Hello, I'm trying LCR module and dont_strip_or_tag_flag parameter. If I use this line in the module configuration: modparam("lcr", "dont_strip_or_tag_flag", 10) and then restart kamailio, i receive this error: ERROR: <core> [modparam.c:141]: set_mod_param_regex(): parameter <dont_strip_or_tag_flag> of type <2> not found in module <lcr> I'm using 4.4.4 version Any hint? Thank you Regards

3 3

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 16 Jan '17

16 Jan '17

Hey Daniel, Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, using raw socket capture, I think this may be a sipcapture issue, debuging kamailio shows normal startup and processing of UDP SIP packets, but does not show any activity with TCP packets. I have enabled/disabled the different capture methods and no TCP SIP packets are processed. Also, I didn't mention before I'm testing on 2 different instances, one is a Homer 3.5 (Kamailio 4.1.2) and another is Homer 5.0 (Kamailio 4.4.0). I'll post to the Homer list and see what they say. Thanks. JR > is it this about a sipcature node? If yes, then it receives the packets > over HEP, meaning that is not going to the normal processing path. > Kamailio detects it is not a SIP packet, executes some callbacks for > non-sip packets registered by modules (in this case sipcapture). The > module detecting it's a packet for it, can execute the request_route > once it builds a local sip packet. > > To troubleshoot further, run with debug=3 in kamailio.cfg and see what > you get in the logs. > > Cheers, > Daniel > > On 13/01/2017 16:33, JR Richardson wrote: >> Iptables is not blocking, but it was worth a check. >> >> Thanks. >> >> JR >> >> >> I assume you have ruled out firewall? It's something that can nab even >> experienced people: >> >> # iptables -Ln >> >> -- Alex >> >> On Thu, Jan 12, 2017 at 03:25:27PM -0600, JR Richardson wrote: >> >>> Hi All, >>> >>> Just enabled SIP TCP on a homer capture server, I can see the SIP TCP >>> Sessions on the server with ngrep, just like all the UDP traffic. I >>> have Kamailio listening on TCP ports but its not capturing any TCP >>> traffic. >>> >>> kamailio.cfg: >>> >>> #disable_tcp=yes >>> listen=tcp:10.99.99.99:5060 #monitor port >>> listen=udp:10.99.99.99:5060 #monitor port >>> >>> loadmodule "pv.so" >>> loadmodule "db_mysql.so" >>> loadmodule "sipcapture.so" >>> loadmodule "textops.so" >>> loadmodule "rtimer.so" >>> loadmodule "xlog.so" >>> loadmodule "sqlops.so" >>> loadmodule "htable.so" >>> loadmodule "sl.so" >>> loadmodule "siputils.so" >>> >>> >>> modparam("sipcapture", "capture_on", 1) >>> modparam("sipcapture", "hep_capture_on", 0) >>> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") >>> modparam("sipcapture", "raw_interface", "eth1") >>> modparam("sipcapture", "raw_ipip_capture_on", 0) >>> modparam("sipcapture", "table_name", "sip_capture") >>> modparam("sipcapture", "raw_sock_children", 4) >>> modparam("sipcapture", "db_insert_mode", 0) >>> modparam("sipcapture", "raw_moni_capture_on", 1) >>> modparam("sipcapture", "promiscious_on", 1) >>> modparam("sipcapture", "raw_moni_bpf_on", 1) >>> modparam("sipcapture", "capture_node", "homer02") >>> modparam("sipcapture", "authorization_column", "authorization") >>> >>> >>> ## logging all INVITES top of the [route] block >>> if (is_method("INVITE|REGISTER")) { >>> xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\" >>> from \"$si\"\n"); >>> >>> Logging reports all SIP UDP traffic to logs fine, but no TCP traffic. >>> >>> root@homer02:~# netstat -al >>> Active Internet connections (servers and established) >>> Proto Recv-Q Send-Q Local Address Foreign Address State >>> tcp 0 0 homer02.me.com:sip *:* LISTEN >>> >>> >>> I don't think this is a homer issue because logging invites is prior >>> to any homer processing. I'm thinking this is something simple I'm >>> overlooking, any help is much appreciated. >>>

2 1

Remove/Append headers in 200 canceling and 200 ok to BYE
by Diego Nadares 16 Jan '17

16 Jan '17

Hi List, I'm trying to add and remove some headers in 200 canceling and 200 ok to BYEs with no success. I don't know in which route to add my remove_hf and append_hf. It seems that I can't catch it on reply either on request route. I tried adding it in event_route[tm:local-response] { xlog("L_ERR", "TM LOCAL REPONSE!"); route(MANAGE_HEADERS); } This is my route to manage hdrs: route[MANAGE_HEADERS]{ xlog("L_ERR", "MANAGE HDRS !"); keep_hf("TH"); # for topoh append("AHDR1:asdf\r\n"); return; } I suppose SERVER HDR should be removed but: U 2017/01/12 12:45:41.566733 172.16.213.38:5060 -> 172.16.50.41:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 172.16.50.41:5060;branch=z9hG4bKgg8ekd3080braugkm700.1. CSeq: 102 CANCEL. From: <sip:xx@172.16.50.41>;tag=SD7b10b01-as5ba5e269. To: <sip:xxx@172.16.213.38>;tag=4a86e96addfb27a943bc86e1872032b6-0a8b. Call-ID: SD7b10b01-6a7d1089793bda79481f25cf88c023ed-c540dl1. Server: kamailio (4.4.4 (x86_64/linux)). Content-Length: 0. Any help will be very appreciated. Thanks in advance! Diego

3 9

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 16 Jan '17

16 Jan '17

Iptables is not blocking, but it was worth a check. Thanks. JR I assume you have ruled out firewall? It's something that can nab even experienced people: # iptables -Ln -- Alex On Thu, Jan 12, 2017 at 03:25:27PM -0600, JR Richardson wrote: > Hi All, > > Just enabled SIP TCP on a homer capture server, I can see the SIP TCP > Sessions on the server with ngrep, just like all the UDP traffic. I > have Kamailio listening on TCP ports but its not capturing any TCP > traffic. > > kamailio.cfg: > > #disable_tcp=yes > listen=tcp:10.99.99.99:5060 #monitor port > listen=udp:10.99.99.99:5060 #monitor port > > loadmodule "pv.so" > loadmodule "db_mysql.so" > loadmodule "sipcapture.so" > loadmodule "textops.so" > loadmodule "rtimer.so" > loadmodule "xlog.so" > loadmodule "sqlops.so" > loadmodule "htable.so" > loadmodule "sl.so" > loadmodule "siputils.so" > > > modparam("sipcapture", "capture_on", 1) > modparam("sipcapture", "hep_capture_on", 0) > modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") > modparam("sipcapture", "raw_interface", "eth1") > modparam("sipcapture", "raw_ipip_capture_on", 0) > modparam("sipcapture", "table_name", "sip_capture") > modparam("sipcapture", "raw_sock_children", 4) > modparam("sipcapture", "db_insert_mode", 0) > modparam("sipcapture", "raw_moni_capture_on", 1) > modparam("sipcapture", "promiscious_on", 1) > modparam("sipcapture", "raw_moni_bpf_on", 1) > modparam("sipcapture", "capture_node", "homer02") > modparam("sipcapture", "authorization_column", "authorization") > > > ## logging all INVITES top of the [route] block > if (is_method("INVITE|REGISTER")) { > xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\" > from \"$si\"\n"); > > Logging reports all SIP UDP traffic to logs fine, but no TCP traffic. > > root@homer02:~# netstat -al > Active Internet connections (servers and established) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp 0 0 homer02.me.com:sip *:* LISTEN > > > I don't think this is a homer issue because logging invites is prior > to any homer processing. I'm thinking this is something simple I'm > overlooking, any help is much appreciated. > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses > Chasing the Azeotrope > > _______________________________________________ > 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

3 2

Kamailio segfault on TM module
by José Seabra 16 Jan '17

16 Jan '17

Hi there, I have noticed a segfault on my kamailio server but as i have coredumps disabled and i don't know how i can reproduce it, i can't provide that kind of information. Anyway i don't know if anyone of you can help with the following information: segfault at 600 ip 00007f5cb97af1ea sp 00007ffed96b8d60 error 6 in tm.so[7f5cb9718000+11d000] ERROR: tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 404 reply when a final 200 was sent out ERROR: sl [sl.c:269]: send_reply(): failed to reply stateful (tm) WARNING: db_postgres [km_dbase.c:281]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] ALERT: <core> [main.c:739]: handle_sigs(): child process 2048 exited by a signal 11 CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 14 ALERT: <core> [main.c:742]: handle_sigs(): core was not generated INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received Database server was down during this segfault, but the segfault was on TM module not on database module. Thank you Regards José Seabra

2 1

dbtext module issue
by Satish Patel 16 Jan '17

16 Jan '17

I am trying use dbtext module instead of mysql and following is my configuration. We are using old SER version because of custom written module. Do you think following config looks good. # -- permissions params -- modparam("permissions", "db_url", "dbtext:///etc/ser/dbtext") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") In function if ( !allow_trusted() && !search ("^User-Agent:.*Some Perl client.*") ) { xlog("L_WARN", "Drop if User-Agent isn't Perl client"); drop; break; }; My dbtest contents $ cat /etc/ser/dbtext/trusted src_ip(str) proto(str) from_pattern(str) 10.0.2.20:any:.* When i run ser in fork=no mode i am seeing following ser -f ser.cfg Content of result table_version(int) 2 Content of [trusted] src_ip(str) proto(str) from_pattern(str) 10.0.2.20:any:.* Content of result proto(str) from_pattern(str) "any" ".* " Content of result proto(str) from_pattern(str) "any" ".* " it look like it read dbtext file and load data but when i use sipp to generate some call then i am seeing following result only, that means its not loading IP address from table, why its extracting two field but not src_ip ? Content of result proto(str) from_pattern(str) "any" ".* "

2 2

comma as param.count delimiter produces parse error
by Juha Heinanen 15 Jan '17

15 Jan '17

Parameter list transformation param.count produces parse error when delimiter is comma (,) character. >From document: {param.value,name[, delimiter]} 'delimiter' allows you to specify a single character to use as the parameter delimiter. For example, when parsing HTTP URL query strings use '&'. When I have in config statement: $var(count) = $(var(servers){param.count,,}); I get to syslog: Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: pv [pv_trans.c:2645]: tr_parse_paramlist(): invalid separator in transformation: count,,} Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: <core> [core/pvapi.c:1629]: tr_lookup(): error parsing [{param.count,,}] Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: <core> [core/pvapi.c:1010]: pv_parse_spec2(): bad tr in pvar name "var" Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: <core> [core/pvapi.c:1036]: pv_parse_spec2(): invalid parsing in [$(var(servers){param.count,,})] at (4) Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) : <core> [core/cfg.y:3407]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 6426, column 19-48: Can't get from cache: $(var(servers){param.count,,}) Jan 13 07:48:15 lohi sip-proxy[18000]: ERROR: bad config file (1 errors) -- Juha

2 2

Kamailio not processing SIP TCP
by JR Richardson 12 Jan '17

12 Jan '17

Hi All, Just enabled SIP TCP on a homer capture server, I can see the SIP TCP Sessions on the server with ngrep, just like all the UDP traffic. I have Kamailio listening on TCP ports but its not capturing any TCP traffic. kamailio.cfg: #disable_tcp=yes listen=tcp:10.99.99.99:5060 #monitor port listen=udp:10.99.99.99:5060 #monitor port loadmodule "pv.so" loadmodule "db_mysql.so" loadmodule "sipcapture.so" loadmodule "textops.so" loadmodule "rtimer.so" loadmodule "xlog.so" loadmodule "sqlops.so" loadmodule "htable.so" loadmodule "sl.so" loadmodule "siputils.so" modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 0) modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") modparam("sipcapture", "raw_interface", "eth1") modparam("sipcapture", "raw_ipip_capture_on", 0) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "db_insert_mode", 0) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "promiscious_on", 1) modparam("sipcapture", "raw_moni_bpf_on", 1) modparam("sipcapture", "capture_node", "homer02") modparam("sipcapture", "authorization_column", "authorization") ## logging all INVITES top of the [route] block if (is_method("INVITE|REGISTER")) { xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\" from \"$si\"\n"); Logging reports all SIP UDP traffic to logs fine, but no TCP traffic. root@homer02:~# netstat -al Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State tcp 0 0 homer02.me.com:sip *:* LISTEN I don't think this is a homer issue because logging invites is prior to any homer processing. I'm thinking this is something simple I'm overlooking, any help is much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

C++ module
by Patrick 12 Jan '17

12 Jan '17

I added a C++ module to kamalio at some point, and now I am trying to integrate that module with another C++ internal library, but I'm getting all sorts of headaches when trying to compile. The latest problem is during the "make install" part of the build - it seems like the objects from the module are "infecting" the link line of the internal library it's dependent on. Here is the Makefile for the module. At the time, the only way I figured to link in the C++ object was adding them to the LIBS. Now, i think this interferes with the 'make install' dependencies. Any thoughs on how to do this properly would be great! Thanks *Module Makefile* # # db_cassandra module makefile # # # WARNING: do not run this directly, it should be run by the master Makefile include ../../Makefile.defs CXX=g++ LD=g++ auto_gen= NAME=db_arl.so ifeq ($(CROSS_COMPILE),) CURL_BUILDER=$(shell \ if pkg-config --exists libcurl; then \ echo 'pkg-config libcurl'; \ else \ which curl-config; \ fi) endif DEFS+=-DKAMAILIO_MOD_INTERFACE -DRTC_COMMON_TRACE *LIBS += db_arl_base.o jsoncpp.o* -lcurl -pthread CXXFLAGS=$(CFLAGS:-Wno-deprecated option=) CXXFLAGS+= -Wno-write-strings -Wno-deprecated -Wno-unused-function -Wno-sign-compare -Wno-strict-aliasing CXXFLAGS+= -I ${COL_LOCAL_BUILDS_ROOT}/ExternalLibs/boost/V1.60.0_27/ CXXFLAGS+= -I ./jsoncpp-dist/ SERLIBPATH=../../lib SER_LIBS+=$(SERLIBPATH)/srdb2/srdb2 SER_LIBS+=$(SERLIBPATH)/srdb1/srdb1 *SER_LIBS+=$(SERLIBPATH)/rtctrace/rtctrace* include ../../Makefile.modules jsoncpp.o: jsoncpp-dist/jsoncpp.cpp @echo COL_LOCAL_BUILDS_ROOT=${COL_LOCAL_BUILDS_ROOT} @echo "Compiling $<" $(CXX) $(CXXFLAGS) $(CFLAGS) $(C_DEFS) $(DEFS) -c $< -o $@ db_arl_base.o: db_arl_base.cpp db_arl_base.h @echo COL_LOCAL_BUILDS_ROOT=${COL_LOCAL_BUILDS_ROOT} @echo "Compiling $<" $(CXX) $(CXXFLAGS) $(CFLAGS) $(C_DEFS) $(DEFS) -c $< -o $@ db_arl.so: db_arl_base.o jsoncpp.o During 'make install' the objects in *red* have no business in the link line for the internal library /opt/rh/devtoolset-4/root/usr/bin/g++ -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -mtune=opteron -Wall -I/home/pshea/_localbuilds/RTC/Common/2.0.0_247//Applications_rhel6_x64_dts4_pic/include -DNAME='"kamailio"' -DVERSION='"4.2.4"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.4.7"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=4002004 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DDBG_F_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -std=c++0x -I/home/pshea/_localbuilds/RTC/Common/2.0.0_247//Applications_rhel6_x64_dts4_pic/include -c rtctrace.c -o rtctrace.o -MMD -MP /opt/rh/devtoolset-4/root/usr/bin/g++ -shared -m64 -Wl,-O2 -Wl,-E -Wl,-soname,librtctrace.so.1 rtctrace.o * db_arl_base.o jsoncpp.o* -lcurl -pthread -L/home/pshea/_localbuilds/RTC/Common/2.0.0_247//Applications_rhel6_x64_dts4_pic/lib -lrtccommon -L/home/pshea/_localbuilds/ExternalLibs/boost/V1.60.0_22/Server_Linux26_x64_dts4_pic/lib -lboost_system -lboost_thread -lpthread -o librtctrace.so.1.0 *g++: error: db_arl_base.o: No such file or directory* *g++: error: jsoncpp.o: No such file or directory*

2 7

Difference between HTTP clients (utils.so vs http_client.so)
by Joel Serrano | VOZELIA 12 Jan '17

12 Jan '17

Hi all, Currently I see there are 2 http clients: utils.so --> http_query() http_client.so --> http_client_query() Any advice on which to use? Any pros/cons of using one over the other? Thanks, Joel. Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado.Les informamos que los datos contenidos en el presente mail, han sido facilitados por usted o recogidos de Fuentes Accesibles al Público. Dicha información personal, será incluida en un fichero bajo la responsabilidad de VOZELIA TELECOM S.L.Todos los ficheros que trata nuestra empresa se encuentran debidamente inscritos ante la Agencia Española de Protección de Datos.Vd. Como propio interesado podrá ejercitar sus derechos de Acceso, Cancelación, Rectificación u Oposición, en el mail: administracion(a)vozelia.com o en la dirección Avenida de Manoteras 8 escalera 3 2ºB, 28050 Madrid.Tal y como establece la Ley Orgánica de Protección de Datos 15/1999 de 13 de Diciembre.Para ejercer sus derechos desde Panama podrá hacerlo en la dirección Vía Argentina, Edificio Mediteranean Loft. Panama.En caso de no ser el destinatario y haya recibido este mensaje por error, agradeceremos que nos lo comunique inmediatamente al remitente sin difundir, almacenar o copiar su contenido.La divulgación o el suministro, en todo o en parte, a cualquier tercero, no podrá ser realizada sin el previo, expreso y escrito consentimiento de VOZELIA TELECOM S.L. .

3 4

RPC exact command to change voicemail IP or Port
by Annus Fictus 12 Jan '17

12 Jan '17

Hello list, I can't find the exact RPC command to change IP or port of voicemail server (module CFG_RPC): the default Kamailio configuration is: */#!ifdef WITH_VOICEMAIL/**/ /**/# VoiceMail Routing on offline, busy or no answer/**/ /**/#/**/ /**/# - by default Voicemail server IP is empty to avoid misrouting/**/ /**/voicemail.srv_ip = "" desc "VoiceMail IP Address"/**/ /**/voicemail.srv_port = "5060" desc "VoiceMail Port"/**/ /**/#!endif/* Thank you Regards

2 8

how to immediately close a tls connection
by Andy 12 Jan '17

12 Jan '17

Hi, trying to close a tls connection immediately, have ref to the connection (tcpconn) pointer please suggest, thanks! -- View this message in context: http://sip-router.1086192.n5.nabble.com/how-to-immediately-close-a-tls-conn… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio and default paramenters to a specific gatewy
by Laura 11 Jan '17

11 Jan '17

Dear all, I'm using kamailio 4.3.6 and LCR modules.. i was wondering if it was possibile to add some special paramenter to all call sent to a specific gateway. I was try to use the " params" field on lcr_gw table but it's not working like i was think.. The issue is simple.. i want to add to field "Privacy: id" to any call i send over a specific gateway.. What is the best and easy way to do that ? Any idea is appreciate. Regards Laura

3 9

Diameter AVP handling from kamaiio.cfg
by Zoltán Miricz 10 Jan '17

10 Jan '17

Dear All, I'm wondering is it possible to catch Diameter Messages and their AVP values from kamailio.cfg just like for SIP messages? (Using pseudo variables) I looked into this function from the attached S-CSCF example. route[REG_MAR_REPLY] { #this is async so to know status we have to check the reply avp xlog("L_DBG","maa_return code is *$avp(s:maa_return_code)*\n"); Sadly it gives back only 1, -1, -2. (which is based on cxdx_mar.c the rc) What I mainly looking for is the EXPERIMENTAL_RESULT (experimental_rc) and some different AVP values besides. Thank you very much in advance! *Cheers,* *Zoltan*

2 5

Kamailio on Cluecon Weekly – Jan 11, 2017
by Daniel-Constantin Mierla 10 Jan '17

10 Jan '17

Hello, for those interested in participating to a (video) chat tomorrow, Wed, Jan 11, I will be joining the ClueCon Weekly at 12:00CT (USA) to discuss about latest Kamailio, what's new and the plans for the near future. Integration with FreeSwitch is going to be among the topics. You can join the video conference call with a webrtc capable browser. You can also join only for audio using a SIP phone or dialing from PSTN or mobile networks. More specific details are available at: - https://www.kamailio.org/w/2017/01/kamailio-on-cluecon-weekly-jan-11-2017/ Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Daniel-Constantin Mierla 10 Jan '17

10 Jan '17

Hello, are you using topoh module? A pcap with all sip messages part of the dialog, from the initial INVITE to the last BYE transaction would be important to have in order to investigate how the headers are changed. Cheers, Daniel On 28/12/2016 13:11, Nihar Ranjan Deb wrote: > > Hi All, > > > > I am a new user of kamailio. We have setup a kamilio server with RTP > proxy. Most of the features are working fine except BYE. > > > > > > Say User A and B are registered in kamailio server. A (caller) > connected to B(called), call established. > > > > 1. If Caller (A) disconnects the call, everything goes fine. B gets > disconnected and call releases. > > > > 2. But if Called party (B) disconnects the call, A never gets > disconnected. I checked trace as well, in this case caller (A) gets a > BYE with B's callid. So A rejects call with "481 call leg transaction > doesn't exists". > > Basically A party and B party has separate dialog established with > Kamilio. So they have two separate callids . All messages are > exchanged correctly. Only when B party disconnects call, kamailio > sends BYE to A party with B party callid. > > > > Please help me to resolve this issue. Do let me know if any more > information required. > > > > > > ------ INVITE-------------------------- > > > > INVITE sip:966268@*** SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport > > Max-Forwards: 70 > > Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> > > To: <sip:966268@***> > > From: "334031"<sip:334031@****>;tag=4406552c > > Call-ID: *82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc* > > CSeq: 2 INVITE > > Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, > OPTIONS, MESSAGE > > Content-Type: application/sdp > > Proxy-Authorization: Digest > username="334031",realm="*****",nonce="****",uri="sip:966268@***",response="***",algorithm=MD5 > > Supported: replaces > > User-Agent: X-Lite release 4.9.6 stamp 82158 > > Content-Length: 334 > > > > v=0 > > o=- 13127393523589215 1 IN IP4 192.168.1.36 > > s=X-Lite release 4.9.6 stamp 82158 > > c=IN IP4 192.168.1.36 > > t=0 0 > > m=audio 57116 RTP/AVP 9 8 120 0 84 101 > > a=rtpmap:120 opus/48000/2 > > a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 > > a=rtpmap:84 speex/16000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=sendrecv > > > > > > > > -------------------------BYE-------------------------------- > > > > > > BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 > > Via: SIP/2.0/UDP > ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 > > Via: SIP/2.0/UDP > 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* > > From: <sip:966268@***>;tag=1665200125 > > To: "334031" <sip:334031@***>;tag=4406552c > > Call-ID: > *!!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRweMH4N6RemHYMI9Y0tV*** > > CSeq: 3 BYE > > Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> > > Max-Forwards: 69 > > User-Agent: Yealink SIP-T20P 9.72.0.80 > > Content-Length: 0 > > > > > > -------------------------- 481--------------------- > > > > SIP/2.0 481 Call/Transaction Does Not Exist > > Via: SIP/2.0/UDP > ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.164.228.12 > > Via: SIP/2.0/UDP > 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* > > To: "334031" <sip:334031@***>;tag=4406552c > > From: <sip:966268@***>;tag=1665200125 > > Call-ID: > !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRweMH4N6RemHYMI9Y0tV** > > CSeq: 3 BYE > > Accept-Language: en > > Content-Length: 0 > > > > > > > > > > Regards, > > Nihar > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

[Error] Install Kamailio 4.4 in Mac 10.12
by Luis De Marchi 10 Jan '17

10 Jan '17

Hi, Sorry, my english, I'm using the translator. I'm having trouble installing Kamailio for Mac 10.12 Terminal: Luis:kamailio luisdemarchi$ sudo make all CC (gcc) [M db_mysql.so] km_db_mysql.o km_db_mysql.c:41:10: fatal error: 'mysql.h' file not found #include <mysql.h> ^ 1 error generated. make[1]: *** [km_db_mysql.o] Error 1 make: *** [modules] Error 1 Luís De Marchi Diretor Administrativo +55 (41) 99815-6800 +55 (41) 3027-7669 www.snowmanlabs.com <http://www.snowmanlabs.com/>

4 4

t_precheck_trans
by Annus Fictus 10 Jan '17

10 Jan '17

Hi, where can I find documentation for this function present on the default script? In TM module not exist (Kamailio 4.4.4) Thank you Regards

3 3

use case
by Thufir Hawat 09 Jan '17

09 Jan '17

I'm ok with Linux and am dipping my toe into Asterisk by running an AWS EC2 instance. What is the most frequent usage for kamailio? Would it mainly be for SIP to SIP? So that the URI would be for a specific domain? thanks, Thufir

3 2

using kamailio to forward ipv6 to ipv4
by anfecora 09 Jan '17

09 Jan '17

Hi guys hope you have a great new year. I would appreciate if anyone can point me in to the right direction . I need to build a proxy to translate from ipv6 to ipv4, but kamailio should not process the registers or invites, only translate from ipv6 to ipv4 and forward all registrations over. is that possible? thanks.

4 5

why a register error?
by Olvera, Victor 09 Jan '17

09 Jan '17

As I attempt to use X-lite I get a 504 error. On the CSCFs I get the following: pcscf 0(2066) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2066) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.1:55880) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) icscf 11(2150) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 0(2138) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2138) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.76:4060) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) 0(2138) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 0 scscf 11(2047) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 192.168.56.78 port 3868 11(2047) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 9(2045) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket Great regards, Victor Olvera victor.olvera(a)verizonwireless.com<mailto:victor.olvera@verizonwireless.com> mobile # 001-214-923-9405

2 1

Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Nihar Ranjan Deb 09 Jan '17

09 Jan '17

Anyone any help on this. Regards, Nihar From: Nihar Ranjan Deb [mailto:nihar.ranjan@microtalkgroup.com] Sent: Wednesday, December 28, 2016 5:42 PM To: 'sr-users(a)lists.sip-router.org' Subject: BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists". Hi All, I am a new user of kamailio. We have setup a kamilio server with RTP proxy. Most of the features are working fine except BYE. Say User A and B are registered in kamailio server. A (caller) connected to B(called), call established. 1. If Caller (A) disconnects the call, everything goes fine. B gets disconnected and call releases. 2. But if Called party (B) disconnects the call, A never gets disconnected. I checked trace as well, in this case caller (A) gets a BYE with B's callid. So A rejects call with "481 call leg transaction doesn't exists". Basically A party and B party has separate dialog established with Kamilio. So they have two separate callids . All messages are exchanged correctly. Only when B party disconnects call, kamailio sends BYE to A party with B party callid. Please help me to resolve this issue. Do let me know if any more information required. ------ INVITE-------------------------- INVITE sip:966268@*** SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport Max-Forwards: 70 Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> To: <sip:966268@***> From: "334031"<sip:334031@****>;tag=4406552c Call-ID: 82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc CSeq: 2 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="334031",realm="*****",nonce="****",uri="sip:966268@***",response=" ***",algorithm=MD5 Supported: replaces User-Agent: X-Lite release 4.9.6 stamp 82158 Content-Length: 334 v=0 o=- 13127393523589215 1 IN IP4 192.168.1.36 s=X-Lite release 4.9.6 stamp 82158 c=IN IP4 192.168.1.36 t=0 0 m=audio 57116 RTP/AVP 9 8 120 0 84 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -------------------------BYE-------------------------------- BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* From: <sip:966268@***>;tag=1665200125 To: "334031" <sip:334031@***>;tag=4406552c Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> Max-Forwards: 69 User-Agent: Yealink SIP-T20P 9.72.0.80 Content-Length: 0 -------------------------- 481--------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.1 64.228.12 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* To: "334031" <sip:334031@***>;tag=4406552c From: <sip:966268@***>;tag=1665200125 Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Accept-Language: en Content-Length: 0 Regards, Nihar

1 0

VERY IMPORTANT: deciding when to remove the MI code
by Daniel-Constantin Mierla 06 Jan '17

06 Jan '17

Hello, we started discussing about removing MI (so called management interface) for very long time, more or less since 2008. The RPC should remain the control interface, given its better structure for commands, parameters, etc ... MI is custom protocol using a line-oriented communication via fifo or socket file with kamailio (e.g., implemented mi_fifo and mi_datagram modules). RPC is the alternative, a more standardized concept, with better structured format. I think it's time to set a clear roadmap for doing the removal. Overall, it will be easier to maintain the code, right now being duplicated code for doing the same operation over MI or RPC, and MI shows its limitations (or complexity to deal with) for advanced needs (see the discussions about how to provide multi-line value parameters over MI). So, I want to know if there are many relying on MI directly and they still want to keep it, what would be the expected duration they need for upgrading their tools to work with RPC interface, other relevant aspects people have in favour of mi vs rpc. I am even willing to do the removal in time befire freezing the 5.0 branch. We will ensure a clean start of 5.x series. The main concern from my point of view is kamctl -- but I think we can preserve the compatibility for kamctl commands and parameters (so command line execution of kamctl will be the same), but the output might be different. That's because it should be easy to updated it to communicate with jsonrpc-s module, but then it will get json-formatted results. To summarize, two big questions to answer: a) Are you ok to remove the MI code/commands? b) If yes to a), are you ok to be done for v5.0? Not providing feedback will be considered as 'yes' for both questions, so **speak up if you want MI to be kept or delay it removal**. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

6 30

presentity tabla, faster update when drop TCP/TLS connection
by João Resende 05 Jan '17

05 Jan '17

I'm testing kamailio with a SIP client with presence support, when I get a connection drop from a client (i.e when the client lost the access to the Internet connection) I do not receive the change in the status to offline immediately, I just received when the timeout from the presentity database finishes. Exists a way to define this? or it is a bug? or should be a new feature? I know that I'm able to change the expiry parameter from the presentity table, but that increases my battery consumption and the data exchange, the same happens if I reduce the publish interval on the client side.

3 9

Re: [SR-Users] using kamailio to forward ipv6 to ipv4
by anfecora 05 Jan '17

05 Jan '17

Yes i am planning on proxing media. The case is that client will only talk ipv6 but the registrar only have ipv4 connectivity. Therefore need to create a proxy to translate this dialogs. Thi thnking on using kamailio to receive the registration modify the pckets and forward registrations in ipv4. I was thinking in using the asterisk kamailio example but instead of asterisk behind will be an sbc. Therefore kamailio will listen in ipv6 and translate over to ipv4 including media of course via rtpproxy. On Jan 5, 2017 9:43 AM, "Frank Carmickle" <frank(a)carmickle.com> wrote: > On Jan 5, 2017, at 12:24 PM, anfecora <anfecora(a)gmail.com> wrote: > > Hi guys hope you have a great new year. > I would appreciate if anyone can point me in to the right direction . > > I need to build a proxy to translate from ipv6 to ipv4, but kamailio should not process the registers or invites, only translate from ipv6 to ipv4 and forward all registrations over. A bit more information about the situation would be helpful. What exactly are you trying to do? > is that possible? 6 to 4 translation is possible, but until we understand the situation I’m not sure we can be that helpful. Are these phones? If so are they dual stack? Is the underlying network dual stack? There’s a lot of work to do to get this right in a Kamailio config. You’ll need to run rtpengine to proxy the media if the endpoints are not dual stack. Who keeps the mapping of the v6 to v4 and v4 to v6 translations? How is that mapping determined? I’d recommend ds-lite if you are in a position to alter the network and the phones/systems allow for it. HTH —FC > thanks. > > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

CDR - acc_cdrs
by Annus Fictus 05 Jan '17

05 Jan '17

Hello, I'm trying to save Kamailio CDR on acc_cdrs table without success. My relevant configuration look like: ----- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 ----- #!define WITH_ACCDB ---- loadmodule "dialog.so" loadmodule "acc.so" modparam("dialog", "dlg_flag", 0) ---- # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_log_enable", 1) modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$td") #!endif Any hint is really appreciate. Regards

2 4

unexpected char - status 6
by AL S 05 Jan '17

05 Jan '17

Hi, Kamailio is throwing error message from "parse_to.c" with status=6. The source code shows status 6 as URI_ENCLOSED. However, tcpdump of the call does not show any enclosed angle bracket is missing. (tcpdump pcap file is attached.) Kamailio log Jan 4 10:18:21 orl-ar-openser01 /usr/sbin/kamailio[2374]: ERROR: <core> [parser/parse_to.c:626]: ERROR: parse_to : unexpected char [#015] in status 6: <<<sip:2057030997@72.31.12.161;tag=gK04001cd3>> . FROM and TO in INVITE received by Kamailio From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161>;tag=gK04001cd3 FROM and TO in INVITE forward by Kamailio for the same call From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161\000\000>;tag=gK04001cd3 PS: this error message is happening on all 4 kamailio servers (3.3.0) PS: call failure only happens in re INVITE (call stays active longer than 15 min) Your help would be appreciated. Thanks, AS

2 2

RTPProxy with VPN
by Rodrigo Moreira 05 Jan '17

05 Jan '17

Hello, I am having problems with RTPPROXY. I need it to work as a bridge. On one side will have a VPN tunnel called a tun0, on the other an eth0 interface. Is it possible for RTPPROXY to run correctly in this scenario? How can I test if RTPPROXY is running correctly? Is there any command? What would the configuration look like inside /etc/default/rtpproxy Please help me. Best regards -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Re: [SR-Users] Regarding usage of Redis Module in Kamailio core
by Vicente Hernando 05 Jan '17

05 Jan '17

Hello Hemanth, this minimal kamailio cfg file works for me to test ndb_redis module. #!KAMAILIO debug=4 log_stderror=yes children=4 fork=yes #disable_tcp=yes mpath="modules/" loadmodule "ndb_redis.so" # An already configured redis server is needed here modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379;db=0") request_route { redis_cmd("srv1", "SET foo bar", "r"); exit; } Changing the parameters to suit your configuration, you can use it to debug your system. Regards, Vicente. On 01/05/2017 07:55 AM, hemanth(a)advaitamtech.com wrote: > > Hello Vicente, > > Thanks for your response, > > In response to your query, > > Yes i have Built ndb_redis module and i can see ndb_redis.so in /usr/local/lib64/kamailio/modules folder, I have installed hiredis depenency also, > > I have also modified the kamailio.cfg file loadmodule and modparam list, > > I feel like the way i am using the redis_cmd is wrong, I think i need to add it in route section of kamailio.cfg file, Can you please guide me regarding how to use redis db in order to update user information to DB on registration of each user. > > Thanks & Regards > Hemanth > > > Message: 8 > Date: Wed, 4 Jan 2017 14:41:42 +0530 (IST) > From: "hemanth(a)advaitamtech.com" <hemanth(a)advaitamtech.com> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] Regarding usage of Redis Module in Kamailio core > Message-ID: <1483521102.18025940(a)apps.rackspace.com> > Content-Type: text/plain;charset=UTF-8 > > Hello, > > I am using Kamailio as a SIP proxy, > > I want to store Registered user information in Redis DB, > > I am able to configure REDIS DB, But i am not able to run the redis command from > > Kamailio core using the function > > "redis_cmd("srvN","SET key 1", "r")" > > I am getting undefined reference error as below > > "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" > > can anyone please help me to fix this issue > > Thanks & Regards > Hemanth > > > > > ------------------------------ > > Message: 9 > Date: Wed, 4 Jan 2017 10:50:05 +0100 > From: Vicente Hernando <vhernando(a)systemonenoc.com> > To: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Regarding usage of Redis Module in Kamailio > core > Message-ID: <b003d2aa-ddd0-574d-b9e3-9b91e44976f1(a)systemonenoc.com> > Content-Type: text/plain; charset=utf-8; format=flowed > > Hello Hemanth, > > just a quick response, I would check if you compiled ndb_redis module > correctly (also if you have hiredis library dependency). > > Also I would check if you loaded ndb_redis module in your configuration > file. > > > Regards, > Vicente. > > On 01/04/2017 10:11 AM, hemanth(a)advaitamtech.com wrote: >> Hello, >> >> I am using Kamailio as a SIP proxy, >> >> I want to store Registered user information in Redis DB, >> >> I am able to configure REDIS DB, But i am not able to run the redis command from >> >> Kamailio core using the function >> >> "redis_cmd("srvN","SET key 1", "r")" >> >> I am getting undefined reference error as below >> >> "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" >> >> can anyone please help me to fix this issue >> >> Thanks & Regards >> Hemanth >> >> >> _______________________________________________ >> 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 > > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > ------------------------------ > > End of sr-users Digest, Vol 140, Issue 3 > **************************************** > > > >

1 0

Kamailio cancel branch only after receiving replies
by Aqs Younas 05 Jan '17

05 Jan '17

Greetings list, I am forking a call to multiple destinations and want to keep the only branch which sends quicker first 180/183 reply and cancel the remaining branches. Below is my related cfg snippet. modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("tm", "cancel_b_method", 2) ... route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); append_branch(); append_branch(); route(RELAY); exit; } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { xlog("L_INFO","Received $rs (IP:$si:$sp)\n"); if(status=~"18[03]"){ t_cancel_branches("others"); xlog("L_INFO","cancelled all other branches\n"); } route(NATMANAGE); } } ... But I see kamailio does not instantly send CANCEL to other branches after it has received 180/180 from any branch. Kamailio is sending CANCEL to other branches after they start sending 180/183 one by one. How can I cancel all other branches instantly? I am sure there is something wrongly configured in my configuration. Any suggestion is much appreciated. Best Regards.

3 3

location
by Slava Bendersky 05 Jan '17

05 Jan '17

Hello Everyone, Why save location will ignore on pv var when trying save on reply 200 OK. if(is_method("REGISTER") && fnmatch("$rs", "200")) { xlog("L_INFO", "OnReply: Registration is $rs OK saving --> $avp(rcvcontact) in db\n"); $var(oar) = "sip:" + $tU + "@" + $(avp(rcvcontact){uri.host}) + ":" + $(avp(rcvcontact){uri.port}); if(!save("location", "0x04", "$var(oar)")) { xlog("L_INFO", "Error saving the location\n"); } Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=619 a=26 n=xlog Dec 26 11:43:20 cavprx00 kamailio: 4(5546) INFO: <script>: OnReply: Registration is 200 OK saving --> sip:4300@client_pub_ip:31376;transport=tls in db Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=620 a=63 n=assign Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=624 a=16 n=if Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=621 a=27 n=save Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-1] (1 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-2] (2 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-2 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1004]: db_update_ucontact_ruid(): ruid:uloc-5860b463-10c5-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1133]: db_update_ucontact_ruid(): contact:sip:4300@10.18.130.27:5060;transport=udp ----> LAN side contact Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 10 chars, out: 10 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 21 chars, out: 21 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 33 chars, out: 33 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 40 chars, out: 40 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 20 chars, out: 20 chars Slava.

2 7

Kamailio v5.0 - development frozen
by Daniel-Constantin Mierla 05 Jan '17

05 Jan '17

Hello, development (master) branch of Kamailio enters now in pre-release phase for version 5.0.0. Therefore, no new feature should be pushed to master until we create a dedicated branch for 5.0 (expected to be in about 4 weeks or so). If in doubt to push or not a commit to master, push it first on a personal branch (or attach to an email) and discuss it on sr-dev. The new modules can be a bit more dynamic if there is need to get them to the right shape (e.g., like decision to rename functions, parameters or adjust database structure). We hope to get many people involved in testing, to reach a stable state before releasing 5.0.0. If you want to get involved and need assistance, don't hesitate to write to mailing lists. Besides the new features, there were two major changes for 5.0: 1) source code tree restructuring - this should not affect the stability of the code, only installation scripts or packaging may still need tuning 2) mi (management/control interface) code has been removed. SIP routing code should not be affected by this change that much, but testing of RPC commands needs a special care. There are few RPC commands not ported yet from the MI code, they can be done during the testing period Moreover, help with updating the wiki page for migration from 4.4 to 5.0 as well as what is new in 5.0 is very appreciated. I will try post updates about them very soon. Many thanks to everyone involved in development of 5.0 and the early testers that played with master branch so far. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

unexpected char [#015]
by AL S 04 Jan '17

04 Jan '17

Hi, Kamailio is complaining about unexpected char [#015] in TO header of INVITE: Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: <core> [parser/parse_to.c:626]: ERROR: parse_to : unexpected char [#015] in status 6: <<<sip:2057030997@72.31.12.161;tag=gK006cb08b>> . Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: <core> [parser/msg_parser.c:180]: ERROR: get_hdr_field: bad to header Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:2057030997@] Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: tm [t_lookup.c:1118]: ERROR: reply cannot be parsed Here is the TO header for the same call in both text and hex format. I am unable to see any ^M char in this TO header. To: <sip:2057030997@72.31.12.161;tag=gK006cb08b 546f3a203c7369703a323035373033303939374037322e33312e31322e3136313e3b7461673d674b30303663623038620d0a Any help would be appreciated. AS,

2 2

IMPORTANT: MI code removed (or disabled)
by Daniel-Constantin Mierla 04 Jan '17

04 Jan '17

Hello, as of now, no active MI reladed code should be in master branch -- the internal library kmi and the mi_datagram, mi_fifo, mi_xmlrpc, mi_rpc and pua_mi modules were removed. Couple of modules still have some mi code disabled with ifdefs, they are pending the port to RPC commands. These are: - carrierroute - ims_dialog - mohqueue - p_usrloc - pdb - rtpengine - userblacklist - utils They should be updated in the next few days. With MI code removed, the code is slimmer and the development effort is reduced -- in the recent past there were various issue rising due to usage of the old line-based MI commands. >From now on, the RPC interface has to be used for interacting with kamailio at runtime. The tools: - kamctl - was already migrated to use RPC via jsonrpcs module. - kamcmd - was developed to use only RPC interface via ctl module. - kamcli - is also working with RPC via jsonrpcs module Helping with testing the RPC commands is very appreciated -- open an issue on bug tracker whenever you discover a problem. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Re: [SR-Users] sr-users Digest, Vol 140, Issue 3
by hemanth＠advaitamtech.com 04 Jan '17

04 Jan '17

Hello Vicente, Thanks for your response, In response to your query, Yes i have Built ndb_redis module and i can see ndb_redis.so in /usr/local/lib64/kamailio/modules folder, I have installed hiredis depenency also, I have also modified the kamailio.cfg file loadmodule and modparam list, I feel like the way i am using the redis_cmd is wrong, I think i need to add it in route section of kamailio.cfg file, Can you please guide me regarding how to use redis db in order to update user information to DB on registration of each user. Thanks & Regards Hemanth Message: 8 Date: Wed, 4 Jan 2017 14:41:42 +0530 (IST) From: "hemanth(a)advaitamtech.com" <hemanth(a)advaitamtech.com> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Regarding usage of Redis Module in Kamailio core Message-ID: <1483521102.18025940(a)apps.rackspace.com> Content-Type: text/plain;charset=UTF-8 Hello, I am using Kamailio as a SIP proxy, I want to store Registered user information in Redis DB, I am able to configure REDIS DB, But i am not able to run the redis command from Kamailio core using the function "redis_cmd("srvN","SET key 1", "r")" I am getting undefined reference error as below "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" can anyone please help me to fix this issue Thanks & Regards Hemanth ------------------------------ Message: 9 Date: Wed, 4 Jan 2017 10:50:05 +0100 From: Vicente Hernando <vhernando(a)systemonenoc.com> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Regarding usage of Redis Module in Kamailio core Message-ID: <b003d2aa-ddd0-574d-b9e3-9b91e44976f1(a)systemonenoc.com> Content-Type: text/plain; charset=utf-8; format=flowed Hello Hemanth, just a quick response, I would check if you compiled ndb_redis module correctly (also if you have hiredis library dependency). Also I would check if you loaded ndb_redis module in your configuration file. Regards, Vicente. On 01/04/2017 10:11 AM, hemanth(a)advaitamtech.com wrote: > Hello, > > I am using Kamailio as a SIP proxy, > > I want to store Registered user information in Redis DB, > > I am able to configure REDIS DB, But i am not able to run the redis command from > > Kamailio core using the function > > "redis_cmd("srvN","SET key 1", "r")" > > I am getting undefined reference error as below > > "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" > > can anyone please help me to fix this issue > > Thanks & Regards > Hemanth > > > _______________________________________________ > 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 ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ End of sr-users Digest, Vol 140, Issue 3 ****************************************

1 0

Regarding usage of Redis Module in Kamailio core
by hemanth＠advaitamtech.com 04 Jan '17

04 Jan '17

Hello, I am using Kamailio as a SIP proxy, I want to store Registered user information in Redis DB, I am able to configure REDIS DB, But i am not able to run the redis command from Kamailio core using the function "redis_cmd("srvN","SET key 1", "r")" I am getting undefined reference error as below "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" can anyone please help me to fix this issue Thanks & Regards Hemanth

2 1

Update: removing MI code
by Daniel-Constantin Mierla 04 Jan '17

04 Jan '17

Hello, based on lib linking in Makefiles, the following modules still use MI, categorized by their state from my point of view. Hopefully some other people will help with those listed in 2) to 6). 1) to be removed - mi_datagram - mi_fifo - mi_rpc - mi_xmlrpc - pua_mi 2) rpc commands implemented - dialog - others modules depend on some mi callback from it (qos and sst) - qos - this looks ready to wipe out the mi code, but didn't get the time not analyze properly as I don't use it usually 3) some rpc commands implemented and the rest should not be very complex - carrierroute - I implemented the dump rpc command, but it was reported that it has issues, so it needs follow up. 4) rpc commands have to be implemented, expecting not to be very complex - cplc - imc 5) expecting some degree of complexity, but they are important modules - rtpengine - rtpproxy 6) not familiar with the mi commands in these modules, so not able to assert what to expect - ims_dialog - some rpc commands are implemented, not sure if the rest of MI are used/usefull - mohqueue - it doesn't seem complex to implement rpc commands at first sight, but the indentation style didn't allowed a fast analyze on a quick look - p_usrloc - several mi commands - sst - uses some callback for MI from dialog module. qos has something similar with rpc alternative already implemented - userblacklist - several mi commands - utils - there are few mi commands related to some forwarding rules: https://www.kamailio.org/docs/modules/devel/modules/utils.html#idp21741924 The 1) to 5) should be done in a way or another, before of after freeze of 5.0.0. But 6) would require the devs of the modules (or the MI parts) to help if they want those commands via RPC. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Kamailio as loadbalancer + NAT pinning
by Vladyslav Zakhozhai 03 Jan '17

03 Jan '17

Hi, I want to use Kamailio as SIP loadbalancer and NAT helper in front of freeswitch boxes. Main tasks here are: 1. load balancing SIP requests to freeswitch boxes 2. fixing contact information during registrations and invites for nat'ed clients (kamailio is not registrar) 3. fixing sdp (or proxying rtp via rtpproxy) for nat'ed clients 4. security (check SIP messages, pike, etc) 5. outbound proxy for freeswitch boxes The main question here is about nat pinning. Acctording to module doc ( http://kamailio.org/docs/modules/4.4.x/modules/nathelper.html) I need nathelper module and usrloc module. So I can proxy REGISTER requests to freeswitch boxes and in onreply_route check for REGISTER statuses: 1. if no success we use pike to limit bruteforce attacks 2. if success I need to save contact to usrloc db table But there is one issue. Multiple registration are allowed on freeswitch boxes and in replies (freeswitch -> kamailio) I see more than one Contact header. Is it an issue or kamailio will handle it without problems? Or I need to save Contact header from request to hash table and retreive it in onreply_route? But I do not know how to save it in usrloc db table then. I thin this is more sophisticated approach. I appreciate your advice. Thank you in advance. -- С уважением, Владислав Захожай

2 4

Help Asterisk with Kamailio unable to register with remote VOIP providers
by Manoj Gupta 03 Jan '17

03 Jan '17

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1: Issue is REGISTER messages sent by asterisk are not responded by kamailio asterisk is constantly doing a re-transmit on REGISTER messages. Retransmitting #1 (NAT) to 10x.3x.6x.236:5060: REGISTER sip:ims.airtel.in SIP/2.0 Via: SIP/2.0/UDP 10x.3x.6x.236:5080;branch=z9hG4bK40ac1ed8;rport Max-Forwards: 70 From: <sip:+91224400@ims.airtel.in>;tag=as565ef860 To: <sip:+91224400@ims.airtel.in> Call-ID: 7dd361ec25cb07344ea9df526e89ca8a(a)10x.3x.6x.236 CSeq: 182 REGISTER User-Agent: Asterisk PBX 11.13.0 Expires: 120 Contact: <sip:+91224400@10x.3x.6x.236:5080> Content-Length: 0 Issue 2: We are not able to see any log messages in /var/log/kamailio.log. We have done all kind of changes suggested on online blogs Why does kamailio not have a pre-set configuration as it is mostly used as sip proxy? Manoj K. Gupta

2 11

Rewrite From domain
by Gonzalo Gasca Meza 03 Jan '17

03 Jan '17

Hi all, I'm using Kamailio to forward calls between 2 Service Providers and I need to rewrite the From header "domain" URI. Example: From: "+18888888888" <sip:+18888888888@sip.sp1.com> to From: "+18888888888" <sip:+18888888888@*sip.sp2.com <http://sip.sp2.com>*> *Call flow:* Phone A --- > SP1 ---> sip ----> (kamailio) SP2 --(LOCATION)-> Phone B When Phone A calls SP2 PhoneB, it contains original sip domain from sp1. ( sip.sp1.com) hence user in SP2 can see call comes from SP1. I would like to rewrite the From domain field in this conditions: a) Calls comes from "sip.sp1.com" AND b) Call is being routed to PhoneB. Right now Im using the following code to find user and send call to B. #!ifdef WITH_ALIASDB # search in DB-based aliases xlog("L_INFO","alias_db_lookup: Call received. $rU\n"); if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif I found this in documentation: $fd - From URI domain if($hdr(From)=~"sip.sp1\.com") { ...} But not sure where is the best place to overwrite the From URI domain header. Thanks

3 7

Help Asterisk with Kamailio unable to register with remote VOIP providers
by Manoj Gupta 02 Jan '17

02 Jan '17

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1: Issue is REGISTER messages sent by asterisk are not responded by kamailio asterisk is constantly doing a re-transmit on REGISTER messages. Retransmitting #1 (NAT) to 10x.3x.6x.236:5060: REGISTER sip:ims.airtel.in SIP/2.0 Via: SIP/2.0/UDP 10x.3x.6x.236:5080;branch=z9hG4bK40ac1ed8;rport Max-Forwards: 70 From: <sip:+91224400@ims.airtel.in>;tag=as565ef860 To: <sip:+91224400@ims.airtel.in> Call-ID: 7dd361ec25cb07344ea9df526e89ca8a(a)10x.3x.6x.236 CSeq: 182 REGISTER User-Agent: Asterisk PBX 11.13.0 Expires: 120 Contact: <sip:+91224400@10x.3x.6x.236:5080> Content-Length: 0 Issue 2: We are not able to see any log messages in /var/log/kamailio.log. We have done all kind of changes suggested on online blogs Why does kamailio not have a pre-set configuration as it is mostly used as sip proxy? Manoj K. Gupta

1 0

Freezing for release v5.0.0
by Daniel-Constantin Mierla 02 Jan '17

02 Jan '17

Hello, I propose to freeze the development of v5.0.0 on January 4, 2017. Hopefully it is just in time for getting the frozen version in next Debian stable. Meanwhile, I hope we can get rid of mi code to have a proper fresh start with cleaned up code in 5.0.0. The big code tree restructuring was already done, config files in embedded languages is there as well (it needs more exports, but that's only about available time to fill some structures, not new feature development). If anyone wants to make other proposals, reply to mailing lists and let's discuss. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Happy 2017!
by Daniel-Constantin Mierla 01 Jan '17

01 Jan '17

A fresh year ahead, let it be a great one for all Kamailio friends! Wishing a prosperous and safe 2017 everywhere! Happy New Year! -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

TOPOS contact issue on RE-INVITE
by Daren FERREIRA 30 Dec '16

30 Dec '16

Hello, I made lots of tests in order to make Kamailio act as a topology hiding proxy, but i always got problems when using topos to store needed Route and Record-Route informations. I thought that was because of possible error(s) playing with : - socket binding - destination tweaking that may have lead to transaction breaking, so i decided to use NAT mode (even if that only can be temporary). My problem is the same. I have not found any way to influence TOPOS on the domain it use on the « Contact » header. Everything is getting well, iPs on Contact Header are toggled between external and internal sides, until RE-INVITE. From there, Contact domain are no more toggled, and so: - Public IP appear on Private side : i can manage my divide to force it to reply to the sender of the request instead of the contact - Private IP appear on Public side : i can’t do anything to make the external side ignore contact’s domain I wonder if that is a bug, related to RE-INVITE, or if there is any variable i can change to say topos which domain to be advertised on Contact headers? PUBLIC NETWORK SIDE RFC6598 NETWORK SIDE ————————— | | PROVIDER Y.Y.Y.Y———————X.X.X.X—| KAMAILIO ]—100.64.10.1———————— 100.64.10.2 ASTERISK (or anything else) | | ————————— 20:58:46.215911 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 1004 INVITE sip:+0598@100.64.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 246 v=0 o=root 79038203 79038203 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=audio 13564 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv 20:58:46.216642 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 375 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport=5060;received=100.64.10.2 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 0 20:58:46.222391 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 1040 INVITE sip:+0598@Y.Y.Y.Y:5060 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 Max-Forwards: 69 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 264 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=root 79038203 79038203 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=audio 57810 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=nortpproxy:yes 20:58:46.234592 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 290 SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE 20:58:46.548771 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 718 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:46.554903 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 719 SIP/2.0 183 Session Progress From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:47.735919 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:47.741229 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:48.050762 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:48.056175 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.811066 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 774 SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:52.818381 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 775 SIP/2.0 200 OK From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.819315 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 446 ACK sip:atpsh-58656af1-769f-2@100.64.10.1 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK18c31028;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 20:58:52.825894 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 491 ACK sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.d149220aa711a5230da51bfea457d8bc.0 Max-Forwards: 69 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> 20:59:00.388481 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 910 INVITE sip:btpsh-58656af1-769f-1@X.X.X.X <sip:btpsh-58656af1-769f-1@X.X.X.X> SIP/2.0 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 273 Max-Forwards: 69 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103905 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=image 17818 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.390679 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 420 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1;rport=5060 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 0 20:59:00.395723 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 890 INVITE sip:+9997@100.64.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 289 Max-Forwards: 68 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=- 1510281183 168103905 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=image 62130 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.397160 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 580 SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Length: 0 20:59:00.397344 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.405004 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 SIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.896864 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.904049 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 TSIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes And so on until timeout and BYE

3 4

Astersik kamailio integration.
by Deep Pratap Singh Tomar 29 Dec '16

29 Dec '16

Hi I have modified the file so that if i send register request to kamailio it sud register on asterisk also. i have shared you the code but i cant see anyting happening in the asterisk server. please help me ASAP. L&T Technology Services Ltd www.LntTechservices.com<http://www.lnttechservices.com/> This Email may contain confidential or privileged information for the intended recipient (s). If you are not the intended recipient, please do not use or disseminate the information, notify the sender and delete it from your system.

2 1

Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 29 Dec '16

29 Dec '16

I implemented full NAT logic as per the sample config. Still unable to resolve the issue. How do I let Kamailio change record_route header to use public ip address? Please help!!! (attached are latest scripts) From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 12:39 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Subject: Kamailio behind NAT, ACK to private IP not advertised public IP. Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

3 4

String split function
by Grant Bagdasarian 29 Dec '16

29 Dec '16

Hi, Is there a string split function built into Kamailio for use in the cfg script? I'd like to split a string of key/value pairs inside the cfg script. For example: Key1=Value1;Key2=Value2; First split on ; Then on = I think I can use the String Transformation {s.select,index,separator}, but I was hoping there is a built in function. {param.valueat,index[, delimiter]} may also be an option. Regards, Grant

2 2

Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 29 Dec '16

29 Dec '16

I am attaching latest config along with kamailio log file and client trace. Please let me know how I can get Kamailio to advertise public ip in record route. I followed all the steps mentioned here - http://www.fredposner.com/1457/kamailio-behind-nat/ Thanks Pranathi From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 5:09 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Cc: Mick McGrellis <MMcGrellis(a)cyracom.com>; 'miconda(a)gmail.com' <miconda(a)gmail.com> Subject: RE: Kamailio behind NAT, ACK to private IP not advertised public IP. I implemented full NAT logic as per the sample config. Still unable to resolve the issue. How do I let Kamailio change record_route header to use public ip address? Please help!!! (attached are latest scripts) From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 12:39 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Subject: Kamailio behind NAT, ACK to private IP not advertised public IP. Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 28 Dec '16

28 Dec '16

Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Nihar Ranjan Deb 28 Dec '16

28 Dec '16

Hi All, I am a new user of kamailio. We have setup a kamilio server with RTP proxy. Most of the features are working fine except BYE. Say User A and B are registered in kamailio server. A (caller) connected to B(called), call established. 1. If Caller (A) disconnects the call, everything goes fine. B gets disconnected and call releases. 2. But if Called party (B) disconnects the call, A never gets disconnected. I checked trace as well, in this case caller (A) gets a BYE with B's callid. So A rejects call with "481 call leg transaction doesn't exists". Basically A party and B party has separate dialog established with Kamilio. So they have two separate callids . All messages are exchanged correctly. Only when B party disconnects call, kamailio sends BYE to A party with B party callid. Please help me to resolve this issue. Do let me know if any more information required. ------ INVITE-------------------------- INVITE sip:966268@*** SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport Max-Forwards: 70 Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> To: <sip:966268@***> From: "334031"<sip:334031@****>;tag=4406552c Call-ID: 82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc CSeq: 2 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="334031",realm="*****",nonce="****",uri="sip:966268@***",response=" ***",algorithm=MD5 Supported: replaces User-Agent: X-Lite release 4.9.6 stamp 82158 Content-Length: 334 v=0 o=- 13127393523589215 1 IN IP4 192.168.1.36 s=X-Lite release 4.9.6 stamp 82158 c=IN IP4 192.168.1.36 t=0 0 m=audio 57116 RTP/AVP 9 8 120 0 84 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -------------------------BYE-------------------------------- BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* From: <sip:966268@***>;tag=1665200125 To: "334031" <sip:334031@***>;tag=4406552c Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> Max-Forwards: 69 User-Agent: Yealink SIP-T20P 9.72.0.80 Content-Length: 0 -------------------------- 481--------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.1 64.228.12 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* To: "334031" <sip:334031@***>;tag=4406552c From: <sip:966268@***>;tag=1665200125 Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Accept-Language: en Content-Length: 0 Regards, Nihar

1 0

rtpengine destroys voice branch on multiple branch call
by Yuriy Gorlichenko 27 Dec '16

27 Dec '16

Hi. I have 2 endpoints 1001 and 1002 1001 registered from 1 device 1002 registered from 2 devices at the same time When I called to 1002 kamailio makes 2 branches rtpengine_manage command called from branch rout for handling every branch directly (it can be different endoint types (ws/tls/udp) for each branch) When i picking up at the 1002 on one device server sends CANCEL (answered elswhere) to another devices of 1002 rtpengine_manage reacts at the CANCEL and deletes this brach after 30 seconds My question is: What is solution for use rtpengine_manage and call branches together?

1 0

set_body_multipart and append_body_part
by Diego Nadares 26 Dec '16

26 Dec '16

Hi guys, I'm trying to append a content type to a message that has only one content type. I tried using just set_body_multipart and now I'm using set_body multipart and append_body_part with no success too. The result is a message with multipart but has "two" sdp and no gtd. It has the old sdp with no ip changed for rtpproxy and the new sdp with ip changed BUT with no content type. And the message has not the end boundary (--unique-boundary-1.) Kamailio log has an info and a warning Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: INFO: <core> [msg_translator.c:1693]: get_boundary(): Content-Type hdr has no params <application/sdp> Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: WARNING: <core> [msg_translator.c:1959]: build_req_buf_from_sip_req(): check_boundaries error kamailio.cfg ...... route[RELAY]{ $avp(ch) = 1 .... if ($avp(ch)){ set_body_multipart(); msg_apply_changes(); $var(gtd) ="IAM,\r\nGCI,asdafsdfasd\r\n\r\n"; append_body_part("$var(gtd)", "application/gtd", "signal;handling=optional"); } .... } Ngrep U 2016/12/26 11:36:22.496634 172.16.213.21:5060 -> 172.16.213.38:5060 ... Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. U 2016/12/26 11:36:22.688458 172.16.213.38:5060 -> 172.16.208.11:5060 .... Content-Type: multipart/mixed;boundary="unique-boundary-1". Mime-Version: 1.0. Remote-Party-ID: <sip:1152565064@172.16.213.38:5060 >;party=calling;privacy=off;screen=no. . --unique-boundary-1. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. . --unique-boundary-1. v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 58724 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=sendrecv. a=rtcp:58725. a=ice-ufrag:hhXDwrco. a=ice-pwd:5temIXP8veOGuD8Dsllm5HXkWU. Any help will be very very appreciated! Thanks in advance. Diego

2 4

help configure dispatcher in Amazon ec2
by huan nguyen duy 26 Dec '16

26 Dec '16

Hi, I’m very appreciated if someone can figure out this problem for me. My aim is to setup HA Kamailio using Dispatcher module in Amazon EC2, system like this: soft phone 1(Linphone) —————> | Load balancer —————> | Sip Server huancomputer | private IP: 172.31.11.35 | 172.31.13.113 soft phone 2(Linphone) —————> | public IP: 35.166.179.244 | 35.166.152.104 huannguyen | | The dispatcher.lst config file in /usr/local/etc/kamailio as below: # gatewayss 1 sip:172.31.13.113:5060;transport=udp <sip:172.31.13.113:5060;transport=udp> 0 0 abc=xyz User records subscriber table: ——————————————————————————————————————————————————— id | username | domain ——————————————————————————————————————————————————— 70 | nguyenhuan | nguyenhuan(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> 71 | huancomputer | huancomputer(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> Location records from location table: ——————————————————————————————————————————————————— username | domain | contact ——————————————————————————————————————————————————— huancomputer | NULL | sip: huancomputer(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:64166;transport=udp nguyenhuan | NULL | sip: nguyenhuan(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:33267;transport=udp But soft phone 1 cannot make a call to soft phone 2, below is data is extracted from wireshark: Load balancer: -------------------------------- No. Time Source Destination Protocol Length Info 177 4.463129 14.161.41.70 172.31.11.35 SIP/SDP 1433 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 179 4.463367 172.31.11.35 14.161.41.70 SIP 452 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 180 4.463421 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1215 32.552942 14.161.41.70 172.31.11.35 SIP 1105 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1216 32.553170 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1219 32.555588 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1220 32.555706 172.31.11.35 14.161.41.70 SIP 593 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1609 42.603150 14.161.41.70 172.31.11.35 SIP 1113 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1610 42.603364 172.31.11.35 172.31.13.113 SIP 1269 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1613 42.605800 172.31.13.113 172.31.11.35 SIP 749 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1614 42.605932 172.31.11.35 14.161.41.70 SIP 599 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1688 44.425237 172.31.11.35 14.161.41.70 SIP 471 Status: 408 Request Timeout | No. Time Source Destination Protocol Length Info 1689 44.425285 172.31.11.35 172.31.13.113 SIP 491 Request: CANCEL sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1691 44.426018 172.31.13.113 172.31.11.35 SIP 519 Status: 200 canceling | Data from Sip Server: ------------------------------- No. Time Source Destination Protocol Length Info 237 6.242928 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 243 6.244409 172.31.13.113 172.31.11.35 SIP 602 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 244 6.244541 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 288 6.710414 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 328 7.710454 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 396 9.710411 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 547 13.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 688 17.710406 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 835 21.710381 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 985 25.710385 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1130 29.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1279 33.710400 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1303 34.332395 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1312 34.334415 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1459 37.710402 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1609 41.710395 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | Huan Nguyen, --------------------------------------- Mobile app developer Phan Thi Company Sai Gon, VietNam Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

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Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 24 Dec '16

24 Dec '16

While waiting for Santa, it's a good time to also express my thanks and greetings to all the friends, developers and community members that made 2016 another amazing year for Kamailio project. Enjoy the winter holidays together with your beloved ones! Merry Christmas! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Merry Christmas to all Kamailians
by Olle E. Johansson 24 Dec '16

24 Dec '16

Friends, Trying to calm down and prepare for Christmas with the family. It’s been another great year for Kamailio and I’m proud to be part of the Kamailio development community. We’ve made great releases, had a great conference and overall done good stuff :-) For all of you that celebrate Christmas - a Merry Christmas! To the rest of you: Happy Holidays! A big Thank You to all developers and a special giga-Thank You to Daniel for all the time you spend working with the code, helping out in discussions on the lists and in the bug tracker. And to all of you in the community - Thank You for being part of this! Greetings from a cold Sweden, currently without snow in my area just north of Stockholm. We celebrate on Christmas Eve (tomorrow) so there’s a lot of food being prepared right now and the last minute shopping is just about to start... Keep your Kamailio running! /Olle PS. If you are not currently following the project on Twitter - it’s time! Find us as @kamailioproject

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baresip version 0.5.0 released
by Alfred E. Heggestad 23 Dec '16

23 Dec '16

Hi, my apologies for the off-topic posting but I think this could be interesting for the SIP Community. Baresip version 0.5.0 is now released, you can find the announcement email here: http://lists.creytiv.com/pipermail/re-devel/2016-December/001154.html Merry Xmas from Norway! /alfred

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Re: [SR-Users] rtjson + mhomed and record route (corrected)
by Diego Nadares 23 Dec '16

23 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The call is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header? ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } .... Thanks in advance. Diego 2016-12-22 12:08 GMT-03:00 Diego Nadares <dnadares(a)gmail.com>: > Hi list, > > We have two eth, one with a private ip and the other with a public ip. We > have MHOMED configured. The call comes from private network and then is > routed to a public network via rtjson routing. The is routed but the thing > is that the record route is being set with the private one (It's the same > when calls come from public and is routed to a private network, record > route has the public and not the private). > > With mhomed configured shouldn't Record Route be updated as VIA header > > ".... When activated, sip-router will select a socket that can reach the > destination (to be able to connect to the remote address). (sip-router > opens a UDP socket to the destination, then it retrieves the local IP which > was assigned by the operating system to the new UDP socket. Then this > socket will be closed and the retrieved IP address will be used as IP > address in the Via/Record-Route headers)..." > > This is part of ngrep > > U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 > INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. > To: <sip:1111111111@172.16.213.38:5060>. > Call-ID: 1-13024(a)172.16.213.21. > CSeq: 1 INVITE. > Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. > .... > > U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 > SIP/2.0 100 trying -- your call is important to us. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > .... > > U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 > INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. > *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* > Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f > Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 > > This is my cfg > > mhomed=1 > > request_route { > > > # per request initial checks > route(REQINIT); > > # NAT detection > #route(NATDETECT); > > # CANCEL processing > if (is_method("CANCEL")) { > if (t_check_trans()) { > route(RELAY); > } > exit; > } > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # handle retransmissions > if(t_precheck_trans()) { > t_check_trans(); > exit; > } > t_check_trans(); > > # authentication > route(AUTH); > > # record routing for dialog forming requests (in case they are > routed) > # - remove preloaded route headers > remove_hf("Route"); > if (is_method("INVITE|SUBSCRIBE")) { > record_route(); > } > > > Thanks in advance. > > Diego. > >

1 1

kamailio dispatcher call flow example
by huan nguyen duy 23 Dec '16

23 Dec '16

Hi, I want to load balance multiple kamailio server using dispatcher module but not success at configuration. I’m not clearly how it work. Someone can show me an example call flow like below i’m very appreciated. Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| | | | 407 F2 | | | |<---------------| | | | ACK F3 | | | |--------------->| | | | INVITE F4 | | | |--------------->| INVITE F5 | | | 100 F6 |--------------->| INVITE F7 | |<---------------| 100 F8 |--------------->| | |<---------------| | | | | 180 F9 | | | 180 F10 |<---------------| | 180 F11 |<---------------| | |<---------------| | 200 F12 | | | 200 F13 |<---------------| | 200 F14 |<---------------| | |<---------------| | | | ACK F15 | | | |--------------->| ACK F16 | | | |--------------->| ACK F17 | | | |--------------->| | Both Way RTP Media | |<================================================>| | | | BYE F18 | | | BYE F19 |<---------------| | BYE F20 |<---------------| | |<---------------| | | | 200 F21 | | | |--------------->| 200 F22 | | | |--------------->| 200 F23 | | | |--------------->| | | | | Huan Nguyen, ——————————————————— Mobile app developer Phan Thi Company Sai Gon, VietNam

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rtpproxy/rtpengine. Help to understand.
by Igor Olhovskiy 22 Dec '16

22 Dec '16

Hi! Issue I can’t figure out. Or all working ok and that’s just me who not understands. I have situation softPhone (111.222.3.2) -> Kamailio w. rtpproxy (111.222.3.3) -> PBX (111.222.3.4) All addresses are on same public network. rtpproxy is running with -l 111.222.3.3 -A 111.222.3.3 I want all media goes through rtpproxy. So, in INVITE and ACK I call rtpproxy_manage(). But according to sip packets on PBX (10.0.0.4) I got media address 10.0.0.2 in SDP, but not 10.0.0.3 If I use scheme, where client in some other NATted network (like 192.168.1.100), all is replaced correctly (means rtpproxy_manage() is working) So, I can’t understand, rtpproxy_manage() changes only rfc1918 addresses or I’m missing something? If any - can provide pcap’s with both cases. Regards, Igor

3 5

rtjson + mhomed and record route
by Diego Nadares 22 Dec '16

22 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } Thanks in advance. Diego.

1 0

Socket in rtjson module
by Diego Nadares 22 Dec '16

22 Dec '16

Hi Guys, Anybody knows if each 'socket' specified in routes struct should be use to send the messages? The socket is configured in kamailio. In kamailio.cfg listen=udp:172.16.213.38:5060 json struct "routes": [ { "uri": "sip:127.0.0.1:5080", "dst_uri": "sip:127.0.0.1:5082", "path": "<sip:127.0.0.1:5084>, <sip:127.0.0.1:5086>", ----> "socket": "udp:172.16.213.38:5060",<---- "headers": { "from": { "display": "Alice", "uri": "sip:alice@127.. In our kamailio has two eth and I want to select which one to use but I can't make it work through rtjson. In the code of the module I can't find where is being assigned the socket. Thanks in advance! Diego

2 3

kamailio dispatcher can not forward call in Amazon EC2
by huan nguyen duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information. Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

1 0

kamailio dispatcher can not forward call in Amazon EC2
by Huân Nguyễn Duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information.

1 0

Memory leak in Kamailio 4.4.4 TLS Module
by raja raja 22 Dec '16

22 Dec '16

Hi, shared memory not freed once TLS connection close means shm available memory decreasing for new connection that leads to the following error *tls: ssl bug #1491 workaround: not enough memory for safe operation* *TestCase:* 1. start kamailio 2. use sample program to establish tls connection and close imediately. 3. Kamailio Log is For first TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63604096 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63545192 For second TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63545192 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63489352 *Memory Leak:* for tls connection connection 63604096 - 63545192 = 58904 bytes not freed. for tls connection connection 63545192 - 63489352 = 55840 bytes not freed. I tested same scenario on kamailio - 4.2.4, 4.3.4 versions result is same. Could you please provide the fix? Thanks & Regards, Raja Chegondi.

2 1

register_timer cb function does not have access to shared memoery var
by Andy 22 Dec '16

22 Dec '16

in mod_init I have something like list = (struct list_t*) shm_malloc ..... this list is available from any functions in the module, now there is a timer registered, in the same module register_timer(callback_f, 0 , 5) the callback_f function when tries to use list, always sees as empty -- View this message in context: http://sip-router.1086192.n5.nabble.com/register-timer-cb-function-does-not… Sent from the Users mailing list archive at Nabble.com.

2 2

new URI [] shorter than old URI - error
by Nelson Migliaro 21 Dec '16

21 Dec '16

Hello, My SIP vendor request me to replace the from in order to authenticate calls. I use: uac_replace_from("111111111", "sip:111111111@vendor-domain.com"); And when finishing calls I have this error: ERROR: uac [replace.c:607]: restore_uri(): new URI [] shorter than old URI [sip:34912345678@vendor-public-ip] Information ------------------------------------------------------------------ 34912345678 -> original CLI 111111111 -> my vendor contract vendor-domain.com -> domain of the vendor ------------------------------------------------------------------ I use the function uac_replace_from in the request_route My Kamailio version is: 4.2.8-1.1 Thank you Nelson.-

1 0

kamailio 4.4 segmentation fault when dp_translate match dialplan with ftmatch operator and empty match expression
by Julia Boudniatsky 21 Dec '16

21 Dec '16

version: kamailio 4.4.4 id:852c47 ENTRY: { PRIO: 16765956 MATCHOP: 2 *MATCHEXP: * MATCHLEN: 0 SUBSTEXP: REPLEXP: ATTRS: 1 } Please see generated core file in extension. Thank you, Julia.

2 7

nathelper - variable natping_interval per registration
by Phil Lavin 20 Dec '16

20 Dec '16

Hi all, Is it possible to set the natping_interval to a different value, per registration, at the time the user registers? For example, user A has 10 seconds and user B has 60 seconds. Thanks

3 3

auth result -3
by Jon Bonilla (Manwe) 19 Dec '16

19 Dec '16

Hi all I'm getting an auth challenge result of -3 and I don't know what it means as it's not listed in the auth module doc. Here's syslog' output: db_postgres [km_dbase.c:230]: db_postgres_submit_query(): sending query ok: 0x7f36d1f75f00 (2) - [select secret from sip_buddies where username='48481628780'] db_postgres [km_res.c:108]: db_postgres_get_columns(): 1 columns returned from the query db_postgres [km_res.c:133]: db_postgres_get_columns(): RES_NAMES(0x7f36d1f92238)[0]=[secret] db_postgres [km_res.c:166]: db_postgres_get_columns(): use DB1_STRING result type db_postgres [km_res.c:211]: db_postgres_convert_rows(): no rows returned from the query auth_db [authorize.c:199]: get_ha1(): no result for user '48481628780@' INFO: <script>: Ha fallado la auth con resultado -3 cheers Jon

3 4

INFO: jsonrpcs used instead of mi_fifo in the default kamailio.cfg and kamctl
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, with MI code planned to be removed, I updated the kamctl and default kamailio.cfg to use jsonrpcs (jsonrpc server) module instead of mi_fifo to interact with kamailio at runtime. The default fifo file for rpc command is /var/run/kamailio/kamailio_rpc.fifo and expects JSONRPC commands as described at: - http://kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idp31525708 Switching back to mi_fifo is still possible by loading the module and editing kamctlrc, but the module might not make it to 5.0, so for new installations and testing is better not to use it. Few rpc commands are missing in carrierroute, otherwise the kamctl commands should work just as before, but be aware that the output is now in json format. Testing of the master branch with the latest kamctl and providing feedback will be really appreciated -- I could do tests only for the modules I typically use. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Forking with 404 replies
by José Seabra 19 Dec '16

19 Dec '16

Hello there, I have a Kamailio server acting as registrar server and when the subscriber has more than 1 AOR, kamailio sends an INVITES to each AOR, if one of them replies 404 what should be the kamailio behavior: 1. should Kamailio reply with an ACK and not pass the 404 to UAC? 2. should Kamailio forward the 404 to UAC? I'm asking this because i'm facing the second behavior, that is, kamailio is forwarding the 404 to UAC and the call is terminated. -- Regards José Seabra

2 3

RFC: renaming the modules with - (dash) in name
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, given the work for 5.0 which will bring major changes on source tree structure, I am considering to remove the - (dash) from the modules that have it in the name. There are 4 of them now: cpl-c janssonrpc-c jsonrpc-c jsonrpc-s One of the reasons is that this kind of name doesn't work for the feature of defining MOD_modname for each loaded module, because - is a separator for ids in config language grammar, so for example, MOD_jsonrpc-c cannot be used in an #!ifdef. More over, for embedded interpreters, most of exported functions there use kamailio module name as package name, and again this will break. Of course, a different package name can be defined, but won't be coherent anymore. Anyone against this renaming? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Re: [SR-Users] Question on multiple 200OK received after forking
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, well, the specs say that all 200ok must be forwarded to the caller, because a middle point cannot decide for the user making the call. Even sending an ACK for the 200ok from a middle point is not enough because the callee of that branch will think it is in an active call. The caller can take different actions, the usual is to ACK the 2nd 200ok and then immediately send BYE. But it can also decide to end previous branch and switch to this one (e.g., uses better codecs, or has video, etc...). Cheers, Daniel On 14/12/2016 15:05, Marino Mileti wrote: > > Hi guys, > > > > I've a Kamailio instance listening on both eth0 and wlan0. > > Kamailio forks call to two branches, so both clients rings together. > > > > In case (very very very rare J) of simultaneous answers is there any > way to kill/hangup/destroy/cancel the "slowest" leg? I've tried to > arrange a semaphore with lock/unlock capabilities of cfgutils, I've > used the drop command but client still sending the 200OK because it's > waiting for an ACK L > > > > Any help will be very appreciated J > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it/ <mailto:marino.mileti@alice.it>// > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

(Correction) unusual characters added to reINVITE message
by AL S 19 Dec '16

19 Dec '16

Hi, An INVITE with following FROM and TO headers received by Kamailio: From: <sip:+11111111111@10.10.10.10;user=phone>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> Kamailio forward it to next node with following FROM and TO headers: From: <sip:+11111111111@10.10.10.10>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> After 15 min, reINVITE for the same call received by Kamailio with following FROM and TO headers: From: <sip:12222222222@20.20.20.20;user=phone>;tag=gK04c77151 To: <sip:11111111111@10.10.10.10>;tag=gK006cb08b Kamailio forward reINVITE to next node with following unusual characters in TO headers From: <sip:12222222222@20.20.20.20;user=phone>;tag=gK04c77151 To: <sip:11111111111@10.10.10.10\000\000;user=phone>;tag=gK006cb08b I was wondering if there is a bug in Kamailio to insert these unusual characters. Thanks, AS

2 1

Limiting calls based on CLI and dst number
by Cibin Paul 19 Dec '16

19 Dec '16

Hi, Is there a way to limit the number of active calls per cli or dst number basis? Yesterday my system had a ddos attack in which 123456 (CLI) was hitting 2345 (DID) at a rate of 100-200 calls per minute. I would like to restrict to 2-4 simultaneous calls per CLI/DST. Any help would be appreciated. Thanks, Cibin

4 9

http_async_client - queries
by Ramachandran, Agalya (Contractor) 17 Dec '16

17 Dec '16

Hi team, We are trying to use kamailio for our project purpose, which will be used as a proxy server. Am trying to use "http_async_client" module to send my rest queries asynchronously. I have few queries regarding the usage of http_async_query() and using PUT method. http_append_header("Authorization: Bearer $var(auth)"); http_append_header("Content-Type: application/json"); http_set_method("PUT"); http_async_query("http://sdk-evm-wcdc-c-001.rtc.sys.comcast.net/events/createxmpprooteventfor…", "{'from_tn':'$fU','to_tn':'$tU','callid':'ci'}", "HTTP_REPLY"); Here am passing from, to, callid in post data field. Passing as 'key':'value' is mandatory or I can pass only value- ? In the file, async_http.c - in async_send_query(), if I modify the post data to the format I need, I see the value is truncated up to three fields only. How can I achieve that my payload format whatever I form in async_send_query(),doesn't get truncated and will be passed to the server-? Can you please explain how this works-? Also, I observed that kamailio is restricting the content-length of HTTP request to 256. Is this something modifiable-? If so please guide me how this can be achieved. Regards, Agalya

2 2

sipp dead calls
by Diego Nadares 16 Dec '16

16 Dec '16

Hi guys, We are testing kamailio with sipp. We are running it with 20cps and some calls do the following. When Kamailio is processing the 'ringing' a '200ok' arrives in the middle. First, kamailio forwards the 200 ok and then the ringing. 'ACK' arrives and, I suppose, the call is established. The thing is than when the 'BYE' arrives kamailio responds with a 404. This is the summary of the call Id Time Source Destination Protocol Len Info 6483 2016-12-14 11:26:06.264697 SIPP-A KAMAILIO SIP/SDP 692 Request: INVITE sip:11111111@172.16.213.38:5060 | 6484 2016-12-14 11:26:06.264937 KAMAILIO SIPP-A SIP 367 Status: 100 trying -- your call is important to us | 6485 2016-12-14 11:26:06.266327 KAMAILIO SIPP-B SIP/SDP 1179 Request: INVITE sip:22222222@172.16.213.31:5060 | *6486 2016-12-14 11:26:06.267217 SIPP-B KAMAILIO SIP 566 Status: 180 Ringing | * 6487 2016-12-14 11:26:06.267268 SIPP-B KAMAILIO SIP/SDP 733 Status: 200 OK | 6488 2016-12-14 11:26:06.267758 KAMAILIO SIPP-A SIP/SDP 788 Status: 200 OK | *6489 2016-12-14 11:26:06.267833 *KAMAILIO SIPP-A* SIP 442 Status: 180 Ringing | * 6490 2016-12-14 11:26:06.268868 SIPP-A KAMAILIO SIP 493 Request: ACK sip:127.0.0.8;line=sr-N6IAzBFwMJZfWJZLM.M7MlF-W.y6Mx14NEt7Nw05NhPQKjaP | 6491 2016-12-14 11:26:06.269162 KAMAILIO SIPP-B SIP 609 Request: ACK sip:172.16.213.31:5060;transport=UDP | 6492 2016-12-14 11:26:06.269614 SIPP-A KAMAILIO SIP 404 Request: BYE sip:11111111@172.16.213.38:5060 | 6493 2016-12-14 11:26:06.269782 KAMAILIO SIPP-A SIP 348 Status: *404 Not here* | We are using modules rtjson, evapi, uac, topoh, rtpproxy for all calls. My debug level is -1. With higher levels this behavior increase. Kamailio is running in a virtual machine with centos7 with 8 cores and 8gb of ram. Do you need any further information? I can send you a pcap or ngrep file. Best regards, Diego.

2 3

unusual characters added to reINVITE message by Kamailio
by AL S 16 Dec '16

16 Dec '16

Hi, An INVITE with following FROM and TO headers received by Kamailio: From: <sip:+11111111111@10.10.10.10;user=phone>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> Kamailio forward it to next node with following FROM and TO headers: From: <sip:+11111111111@10.10.10.10>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> After 15 min, reINVITE for the same call received by Kamailio with following FROM and TO headers: From: <sip:7272697443@71.44.13.84;user=phone>;tag=gK04c77151 To: <sip:2057030997@72.31.12.161>;tag=gK006cb08b Kamailio forward reINVITE to next node with following unusual characters in TO headers From: <sip:7272697443@71.44.13.84;user=phone>;tag=gK04c77151 To: <sip:2057030997@72.31.12.161\000\000;user=phone>;tag=gK006cb08b I was wondering if there is a bug in Kamailio to insert these unusual characters. Thanks, AS

1 0

ACK not routed to destination
by Wim Van Cauwenbergh 16 Dec '16

16 Dec '16

Hi, I have setup kamailio with the dispatcher module and have a problem with the ACK in response to 200OK not being forwarded by kamailio: Scenario is very simple: sipp(.219) --> kamailio (.220) --> sipp(.239) SIP trace: root@DNS-enum-8Gb-3:/var/log# ngrep -W byline -d eth0 port 5060 interface: eth0 (10.57.26.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # U 10.57.26.219:5060 -> 10.57.26.220:5060 INVITE sip:service@10.57.26.220:5060 SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.219. s=-. c=IN IP4 10.57.26.219. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.239:5060 INVITE sip:service@10.57.26.220:5060 SIP/2.0. Record-Route: <sip:10.57.26.220;lr>. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.219. s=-. c=IN IP4 10.57.26.219. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.219:5060 -> 10.57.26.220:5060 ACK sip:10.57.26.239:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-5. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 ACK. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. XContactinOK: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # *U 10.57.26.219:5060 <http://10.57.26.219:5060> -> 10.57.26.220:5060 <http://10.57.26.220:5060>* *ACK sip:10.57.26.239:5060;transport=UDP SIP/2.0.* *Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-5.* *From: sipp <sip:sipp@10.57.26.219:5060 <http://sip:sipp@10.57.26.219:5060>>;tag=15698SIPpTag021.* *To: sut <sip:service@10.57.26.220:5060 <http://sip:service@10.57.26.220:5060>>;tag=14783SIPpTag018.* *Call-ID: 1-15698(a)10.57.26.219 <1-15698(a)10.57.26.219>.* *CSeq: 1 ACK.* *Contact: sip:sipp@10.57.26.219:5060 <http://sip:sipp@10.57.26.219:5060>.* *Max-Forwards: 70.* *Subject: Performance Test.* *XContactinOK: <sip:10.57.26.239:5060;transport=UDP>.* *Content-Length: 0.* . # U 10.57.26.219:5060 -> 10.57.26.220:5060 BYE sip:10.57.26.239:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-7. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 2 BYE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-7. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 2 BYE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.219:5060 -> 10.57.26.220:5060 BYE sip:service@10.57.26.220:5060 SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1--1. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@10.57.26.219:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1--1. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 3 BYE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. *kamailio.conf:* # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- #modparam("dispatcher", "db_url", # "mysql://openser:openserro@localhost/openser") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") ####### Routing Logic ######## # main request routing logic route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); exit; } # 1= only check the header RURI, 1126 checks some other general stuff if(!sanity_check("1126", "1")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing xlog("#######WITHINDLG:hastotag $si:$sp\n"); if (loose_route()) { xlog("#######WITHINDLG:looseroute $si:$sp\n"); if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("#######WITHINDLG:ACK transaction found $si:$sp\n"); # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. xlog("#######WITHINDLG:ACK no transaction found so discard $si:$sp\n"); exit; } } sl_send_reply("404","Not here"); } exit; } } Debug: Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <ACK> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:10.57.26.239:5060 ;transport=UDP> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-15698-1-5>; state=16 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts... Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: sl [sl_funcs.c:412]: sl_filter_ACK(): DEBUG : sl_filter_ACK: to late to be a local ACK! Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=14783SIPpTag018 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [57]; uri=[sip:service@10.57.26.220:5060] Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [sut < sip:service@10.57.26.220:5060>] Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> <ACK> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: siputils [checks.c:106]: has_totag(): totag found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) ERROR: <script>: #######WITHINDLG:hastotag 10.57.26.219:5060 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: rr [loose.c:113]: find_first_route(): No Route headers found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: rr [loose.c:929]: loose_route(): There is no Route HF Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=9 global id=8 T start=(nil) Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=53919, isACK=1 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=9 global id=9 T end=(nil) Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) ERROR: <script>: #######WITHINDLG:ACK no transaction found so discard 10.57.26.219:5060 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)

3 3

Siremis installation with postgresql.
by José Seabra 16 Dec '16

16 Dec '16

Hi there, I'm facing some issues trying to install siremis with postgresql. The file Config.xml has the correct configuration for postgresql driver but It seems that during installation phase siremis can't create database structure for postgresql. Anyone have installed siremis with postgresql, if yes can you share the steps needed for this installation. -- Cumprimentos José Seabra

4 6

kamcli - transferring the github repository to kamailio organization
by Daniel-Constantin Mierla 16 Dec '16

16 Dec '16

Hello, kamcli (kamailio command line interface tool written in python) got a consistent set of features trying to become in pair with kamctl. It is currently hosted by asipto organization on github, but I am considering to transfer (move) it to kamailio organization in order to manage easier the access for kamailio devs as well as increase its exposure to community. It interacts with kamailio using rpc commands via jsonrpcs module. You can read more about what it can do at: - https://github.com/asipto/kamcli The commands are implemented in a plugin-like fashion -- see the ones done so far: - https://github.com/asipto/kamcli/tree/master/kamcli/commands Given that it can handle better the jsonrpc responses, provides more options to format the output and the help is embedded in each command, I think it is a good candidate to replace the old shell(bash)-based kamctl. Probably we can adjust kamctl for rpc when mi code will be removed, but the options to build and extend in long term a good cli for kamailio are quite limited in shell. For the sake of an example, here is the implementation of 'ps' command, handling the json rpc to display a very compact list of kamailio processes: - https://github.com/asipto/kamcli/blob/master/kamcli/commands/cmd_ps.py Dealing with database records is also rather trivial: - https://github.com/asipto/kamcli/blob/master/kamcli/commands/cmd_group.py Anyone with comments/suggestions/alternatives? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Forwarding Emergency calls based on caller IP address
by João Vitor Torres 15 Dec '16

15 Dec '16

Hello all, I am looking for a solution to forward emergency calls inside an enterprise network. Is there any way to forward calls based on the IP address of caller device? The requirement is that calls to a unique emergency number must be forwarded to specific local numbers in each building, and the IP address of the caller is the criteria for choosing the destination. A table maps IP nets to specific local numbers. The SIP contact header has the caller IP address, but I am new to Kamailio and I am not sure where to start. Any suggestions? Thanks in advance. João Vitor.

2 1

Re: [SR-Users] htable lock - unlock
by Daniel-Constantin Mierla 15 Dec '16

15 Dec '16

Hello, the log message you refer to is inside the if block: if ($sht(pht=>$ci)!=$null) { ... } but you don't set the item in memory, no assignment like: $sht(pht=>$ci) = ...; The sht_lock(pht=>$ci) is just locking the slot where the value of $sht(pht=>$ci) will be stored, but don't store any valye for that variable. Cheers, Daniel On 15/12/2016 11:46, Marino Mileti wrote: > > Hi guys, > > > > I'm going out of mind J > > I've this piece of script inside an "on_reply_route" that captures > only 200OK of INVITE (i would like to save in ht the contact of who > has answered a call) > > > > sht_lock("pht=>$ci"); > > if ($sht(pht=>$ci)!=$null){ > > xlog("L_INFO","Before assignment.."); > > $sht(pht=>$ci)=$var(myvar); > > xlog("L_INFO","Assigned done..."); > > sht_unlock("pht=>$ci"); > > } > > else{ > > xlog("L_INFO","Skipping.."); > > } > > sht_unlock("pht=>$(hdr(Call-ID))"); > > > > The situation is that I never see the "Assigned done" message in log > file... > > > > Is this the right use of sht_lock/sht_unlock method? > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

2 1

Updating to SIREMIS 4.3
by Kevin Greene 15 Dec '16

15 Dec '16

Hello fellow SIP people: I upgraded from SIREMIS 4.2 to 4.3 for my Kamailio system and I now get the error message shown below. Any ideas on how to solve it? I am assuming there was a change on how SIREMIS did a date query and my version of Linux (6.8) or my PHP does not support it as expected. Any help would be useful. Thanks. Kevin [{"target":"ERROR","content":"\n[2016-12-14 22:59:55 (GMT)] An exception occurred while executing this script: \nError message: #2, date(): It is not safe to rely on the system's timezone settings. You are *required* to use the date.timezone setting or the date_default_timezone_set() function. In case you used any of those methods and you are still getting this warning, you most likely misspelled the timezone identifier. We selected 'UTC' for 'UTC\/0.0\/no DST' instead<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php:160<\/font> \n *function:<\/b> errorHandler ( 2, \"date(): It is not safe to rely on the system's timezone settings...\", \"\/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php\", 160, Array(8) ) @ \/var\/www\/siremis-4.3.0\/openbiz\/bin\/sysheader.inc 117\n *function:<\/b> userErrorHandler ( 2, \"date(): It is not safe to rely on the system's timezone settings...\", \"\/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php\", 160, Array(8) ) @ \n *function:<\/b> date ( \"m\/d\/Y\" ) @ \/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php 160\n The "date" command returns: $ date Thu Dec 15 01:13:41 UTC 2016 The /etc/localtime entry is: $ more /etc/localtime TZif2 UTC $ which php /usr/bin/php $ php -version PHP 5.3.3 (cli) (built: Aug 11 2016 13:38:15) Copyright (c) 1997-2010 The PHP Group Zend Engine v2.3.0, Copyright (c) 1998-2010 Zend Technologies PHP I am using is the latest patched version I have. ***

3 2

htable lock - unlock
by Marino Mileti 15 Dec '16

15 Dec '16

Hi guys, I'm going out of mind J I've this piece of script inside an "on_reply_route" that captures only 200OK of INVITE (i would like to save in ht the contact of who has answered a call) sht_lock("pht=>$ci"); if ($sht(pht=>$ci)!=$null){ xlog("L_INFO","Before assignment.."); $sht(pht=>$ci)=$var(myvar); xlog("L_INFO","Assigned done..."); sht_unlock("pht=>$ci"); } else{ xlog("L_INFO","Skipping.."); } sht_unlock("pht=>$(hdr(Call-ID))"); The situation is that I never see the "Assigned done" message in log file... Is this the right use of sht_lock/sht_unlock method? Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

Kamailio 4.3 Rewriting From and db_alias values
by Gonzalo Gasca Meza 15 Dec '16

15 Dec '16

Hi forum, This is the scenario: VM <- Kamailio -> SP1 ----- SP2 ---- PhoneA (john) PhoneB PhoneA is configured in *dbaliases* table: alias_username, alias_domain, username, domain 11111111, sp1.com, john, sp2.com When PhoneB dials 1111111, Kamailio converts 111111(a)sp1.com to john(a)sp2.com Now I need to provide VM services to PhoneA. (Opposite direction) PhoneA calls voicemail but the calling number is john(a)sp2.com I need to have SP1 use the 11111111(a)sp1.com format How can I overwrite the From? Pseudo algorithm ------------------------ If destination is VM and from domain ($fU) is sp2.com: Find user URI: ( In this case john extracted from john(a)sp2.com in dbaliases) If user is found: rewrite From URI: use alias_username@alias_domain I'm using default kamailio.cfg and calls successfully are forwarded to VM Thank you

2 1

Question on multiple 200OK received after forking
by Marino Mileti 14 Dec '16

14 Dec '16

Hi guys, I've a Kamailio instance listening on both eth0 and wlan0. Kamailio forks call to two branches, so both clients rings together. In case (very very very rare J) of simultaneous answers is there any way to kill/hangup/destroy/cancel the "slowest" leg? I've tried to arrange a semaphore with lock/unlock capabilities of cfgutils, I've used the drop command but client still sending the 200OK because it's waiting for an ACK L Any help will be very appreciated J Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

question on call cancel
by Kelvin Chua 14 Dec '16

14 Dec '16

I have a peculiar scenario, maybe you guys have an easy solution: 1. a call comes in through an SBC (which is also kamailio) 2. SBC passes call to core kamailio server 3. call reaches 180 ringing state 4. client suddenly changes from wifi to 4G (using 4G IP, VIA headers are all updated) 5. client sends cancel and receives cancelling msg from SBC 6. SBC sends cancel to kamailio core 7. kamailio core sends back Request Terminated after termination the issue is, this message was sent to the wifi IP not the 4G when checking the network dumps, looks like this was because the INVITE was received on kamailio core from the old wifi IP. Is there maybe a way to update the transaction perhaps? Kelvin Chua

2 2

Multi-Domain dispatcher
by Satish Patel 14 Dec '16

14 Dec '16

We have currently dispatcher running for single domain, but in future we have more domain coming so i want to do multi domain dispatching for example dispatcher redirect request for foo.com to foo.registar and bar.com to bar.registar SIP server currently we have following code in dispatcher logic. what should we do to handle multi-domain if(!ds_select_dst("1", "2")) { # if we are here that means no destination is available. We notify the user by 404 and exit the script. xlog("L_NOTICE", "No destination available!"); send_reply("404", "No destination"); exit; }

2 1

regular expression question
by Satish Patel 14 Dec '16

14 Dec '16

I am trying to block SIP scanner so i am trying to use following logic only allow "My-UserAgent" and block rest but its throwing error, but if i use =~ regular expression which works! why negative match doesn't work? if($ua !~ "(My-UserAgent") { xlog("L_INFO","On more scriptkiddie, coming from $si, blocking"); exit; } what could be wrong here?

3 3

event for new tcp connection from client
by Andy 14 Dec '16

14 Dec '16

similar to tcp closed, can this be achieved when a client is connect to tcp -- View this message in context: http://sip-router.1086192.n5.nabble.com/event-for-new-tcp-connection-from-c… Sent from the Users mailing list archive at Nabble.com.

3 6

Kamailio TM - t_set_fr
by José Seabra 13 Dec '16

13 Dec '16

Hello there, I'm using t_set_fr(0,2000) on calls to registered phones but by any reason this isn't working, the call only enters on the failure route after 30 seconds in case of no response from SBC/Phone, anyone has any idea what can be the reason for that? if(lookup("location")) { xlog("L_INFO", "routing to registered phone ruid=$ruid R=$ru - ID=$ci \n"); t_set_fr(0, 2000); t_on_failure("ASBC_FAILURE"); t_on_branch("ASBC_BRANCH"); route(RELAY); exit; } My RELAY route: route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } version: kamailio 4.4.2 (x86_64/linux) Thanks -- Cumprimentos José Seabra

2 11

Accessing SDP changes
by Pete Kelly 13 Dec '16

13 Dec '16

Hi I am using Kamailio 4.4. and calling the function sdp_remove_transport in order to remove some SRTP media lines. The function works (of course!) however I would like to get access to the modified SDP within the cfg. I thought this would do the trick: sdp_remove_transport("RTP/SAVP"); sdp_get("$avp(foo)"); xlog("$avp(foo)"); However $avp(foo) does not contain the newly modified SDP. Is there a way for me to access it? I have been looking at things like t_save_lumps() and msg_apply_changes(). Any pointers or confirmation that this is or isn't possible would be very welcome! Thanks Pete

3 5

Tsilo iOS Push Notifications with Kamailio/Asterisk Integration
by Benjamin Fitzgerald 12 Dec '16

12 Dec '16

Hello, my goal is to address iOS 10's removal of periodic TCP connections in iOS apps with their replacement PushKit, a push-based alternative all VoIP apps should implement. In order to have iOS apps receive SIP calls in the background, I believe I need to use tsilo to suspend the INVITE transaction send a push notification to the device. I have been following the presentation here ( http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-…) however I am not sure how to integrate it with Asterisk ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ). My questions can be summarized below: (1) What changes need to be made from Federico's presentation to use with the Kamailio/Asterisk integration? (2) How can I modify the config to send push notifications for EVERY invite? This is to address the scenario where a device goes offline before the SIP registration expires and a call is received within that window. iOS is very aggressive about killing background applications so it's likely as soon as user leaves the VoIP app, the device will become unreachable. Thanks for any help. Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us

2 3

DTMF Using INFO
by Nahum Nir 12 Dec '16

12 Dec '16

Hello, I tried Googling but didn'tfind anything. Is it possible to support DTMF using INFO? Thanks, Nir

4 8

msilo timestamp
by Slava Bendersky 12 Dec '16

12 Dec '16

Hello Everyone, What format stored inc_time in msilo module ? I am trying convert it to human readable format, but time looks UTC SELECT FROM_UNIXTIME($(xavp(dbquery[$var(i)]=>inc_time)), '%M %D %Y %h:%i:%s') Slava.

2 1

Error using Siremis 4.3.0 with Kamailio 4.4 and PHP 7
by Diogenes Aviles 09 Dec '16

09 Dec '16

I'm using Siremis 4.3.0 and PHP7. Installation was OK, but running siremis I have some problems in page "Administration-User Management". When I select an user there is the following error: [2016-12-06 23:18:47 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/siremis-4.3.0/ openbiz/bin/Resource.php:283 *function:* errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/siremis-4.3.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ *function:* trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/siremis-4.3.0/openbiz/bin/Resource.php 283 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/openbiz/bin/BizSystem.php 544 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/siremis/modules/service/userEmailService.php 108 *function:* SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/siremis-4.3.0/siremis/modules/common/form/ErrorForm.php 34 *function:* Report ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 110 I verified and the file "/var/www/siremis-4.3.0/openbiz/bin/Resource.php" exist. But I don't have any idea where the error is. Some help will be appreciate. Regards -- Diogenes

2 2

Asterisk Security Advisory (AST-2016-009)
by Matthew Jordan 09 Dec '16

09 Dec '16

Hey all - The Asterisk project just released a security advisory for a security vulnerability in which Asterisk using chan_sip with a proxy can allow for unauthenticated calls. This affects all supported versions of Asterisk (11, 13, 14). Since that may be relevant to those on this mailing list who are not also on the asterisk-users mailing list, I thought it prudent to mention it here as well. A description of the vulnerability follows: Description The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue does not affect you. The announcement can be seen here: http://lists.digium.com/pipermail/asterisk-announce/2016-December/000662.ht… Thanks again to Walter Doekes for reporting the vulnerability and providing the patch to fix it. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org

3 2

IP-based security + registrations
by JLL 09 Dec '16

09 Dec '16

Hello, I would like to know if anyone can help me accomplish the following: I have a Kamailio server (4.2.3 on CentOS 7 x64) and 5 Asterisk servers (1.6.2.13 . I know, it's old but we have a lot of custom modules written for it). I need for certain IP's to be "trusted" (able to send INVITE's to K from the internet - these would be our origination providers), but any other IP's that come along would need to register before sending an INVITE (the REGISTER would be sent to any of the Asterisk servers for validation, if this is possible). Can this be done? Thank you!

2 1

Siremis manual to learn
by Saranga Nagesh 08 Dec '16

08 Dec '16

Hi, I have installed kamailio and Siremis in centos but I don't know how to use Siremis. Can you forward me manual or link to learn about Siremis. Thanks & Best Regards, Nagesh Saranga

2 1

Action: restructuring the source code tree
by Daniel-Constantin Mierla 07 Dec '16

07 Dec '16

Hello, today I am going to do the restructuring of the source code tree as discussed previously. - http://lists.sip-router.org/pipermail/sr-dev/2016-December/037715.html If anyone has patches to push using existing structure, do it before 12:00UTC today. If you need more time, write to sr-dev and let's see how we can find an easy solution. Once I finish with the restructuring, I will send another notification, from that moment the commits can go on as usual. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Send SIP Info within a dialog using $uac_req(method)="INFO"
by Jonathan Hunter 07 Dec '16

07 Dec '16

Hi Guys, Is it still the case that when using uac_req_send, you cant send withing a specific dialog?I can modify call-id, but I presume tags may be more of a problem?See old post below from 2015; > I am familiar with uac_req_send. but how do I send it with in a > specific dialog and with data in the INFO req ? sending a new request inside a dialog is not possible with uac_req_send(). It is not easy over all because you change the sequence order (CSeq value). Practically, you need to track how many requests you sent from the middle to update (and restore in reply) when caller or callee sends a new request. dialog module can track changes in CSeq for requests sent to callee, being used now for authentication of INVITE to another provider, when Kamailio adds the credentials. But for more you would need to extend the dialog module. I just need to send a SIP info within an established dialog to stop some function up stream, so wondered if this is still a blocker? Many thanks Jon

2 40

Error using Siremis 4.3.0
by Diogenes Aviles 07 Dec '16

07 Dec '16

I'm using Siremis 4.3.0 and PHP7. Installation was OK, but running siremis I have some problems in Administration-User Management. When I select an user there is the following error: [2016-12-06 23:18:47 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/siremis-4.3.0/openbiz/bin/Resource.php:283 *function:* errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/siremis-4.3.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ *function:* trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/siremis-4.3.0/openbiz/bin/Resource.php 283 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/openbiz/bin/BizSystem.php 544 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/siremis/modules/service/userEmailService.php 108 *function:* SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/siremis-4.3.0/siremis/modules/common/form/ErrorForm.php 34 *function:* Report ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 110 Regards -- Diogenes

2 1

Getting destination ip
by David Villasmil 06 Dec '16

06 Dec '16

Hello Guys, I'm wondering how do I get destination IP on a reply? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 7

Re: [SR-Users] Local messages are malformed in Kamailio 4.4.X
by g.Helio 06 Dec '16

06 Dec '16

2 1

Limiting Registration Contacts
by Colin Morelli 05 Dec '16

05 Dec '16

Hey all, I know Kamailio's registrar module has a max_contacts parameter that will limit the number of active contacts for an AOR. However, is there any mechanism to control what happens when that number is hit? When using a web client where every page refresh registers a new contact with Kamailio, it would be useful to automatically roll off the oldest registrations and replace to make room for the new ones. I know this can be manually accomplished with sqlops, but I was hoping something like this already existed. Best, Colin

3 5

INFO: modules purple and iptrtpproxy moved to obsolete folder
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, be aware that the modules purple and iptrtpproxy were moved to obsolete folder. They were not compiling for several years and nobody showed interest in updating them. If anyone wast to rescue them, she/he is welcome to get them back and upgrade the code to work with current versions of their dependencies. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

INFO: kcore internal lib removed
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, mainly for those having private modules or other components for kamailio, take note that the internal library kcore has been removed. It was supposed to be a termporary place where to keep some kamailio specific functions at the time of merging with ser code back in 2008. The main code was related to internal statistics, which was re-implemented as fast counters several years ago, in libkcore being only some wrappers. You have to include now directly the counters.h from core. There were several files with some str operations helpers, not they are exported via strutils.h from core. The sst parser was used only by sst module and moved to it, same being done with some stats helper functions used only by snmpstats module. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

kamailio bridge internal and external network
by zhang90501 05 Dec '16

05 Dec '16

My kamailio listen at internal ip and external ip(advertise parameter), it send message to freeswitch from internal ip and to client from external ip. How can I configure my kamailio to auto generate correct record route header?

3 4

Re: [SR-Users] Kamailio Processes Stop Responding
by Peter DiSanto 05 Dec '16

05 Dec '16

OK thank you Alex. I will look into your suggestions next time the problem arises. Each call does generate one SQL LCR lookup, one CDR INSERT, and also a timer route that runs 3 queries every few minutes. Once Kamailio stops responding it will continue that way indefinitely until a process restart. Will report back. Thanks. Pete D

3 2

Restructuring source code tree for v5.0
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, I already started playing with various scripts to do the restructuring of the source code tree for v5.0 as we discussed in the past months and I am considering to do it quite soon, in the next few days. The result will look like: /Makefile (only-forwarding make commands to various subdirs) /README|INSTAL|LICENSE|... /src/Makefile* (build makefiles) /src/main.c (main c file to generate the binary) /src/core/*.{c,h} (c code for core, including subfolders such as parser, mem, atomic, ...) /src/lib/* (subfolders with the internal libraries) /src/modules/* (subfolders with the modules) Other subfolders from root directory may be relocated, but to be decided along the way. At the time of doing the restructuring, I will try to merge all the pull requests that look ok. Therefore it would be good that anyone interested in open pull requests to do the review (and merge) meanwhile. Also, I think it would be good to disable travis and nightly master builds during few days when the restructuring happens in order to avoid flooding sr-dev with alerts. If anyone has suggestions to add, do not hesitate to write on mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Re: [SR-Users] msilo
by Slava Bendersky 05 Dec '16

05 Dec '16

Hello Daniel, I updated OFFLINE MESSAGE store based on username. https://paste.fedoraproject.org/499377/14809030/ Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 10:59:45 Subject: Re: [SR-Users] msilo Hello, the order you use to store and update the record is not safe, other messages from different users can be stored by another kamailio process. Use callid and username to select the last record inserted matching them, eventually protected with lock/unlock from cfgutils. Cheers, Daniel On 24/11/2016 14:34, Slava Bendersky wrote: Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: BQ_BEGIN Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

1 0

Re: [SR-Users] RTP Pyload
by amar Smart Telecom 04 Dec '16

04 Dec '16

Dear all, We are seeing that some RTP Payload are with 14 bytes and some are with 15 bytes. 1. 14 byte is correct or 15 byte is correct? 2. Is TOC present in all payloads i.e. in all RTP messages? Following example with 14 byte payload. Regards, Amar

3 4

Re: [SR-Users] msilo
by Slava Bendersky 04 Dec '16

04 Dec '16

Hello Everyone, I need help with while loop. Can't find why it try go over only first value from db search and not each value. https://paste.fedoraproject.org/497911/48081139/ Any help thank you. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 02:12:21 Subject: Re: [SR-Users] msilo Hello Daniel, I am trying more simple example of $dbr in REGISTER first. Insert into location temporary data. The problem is not looping though all entries in database only last one. Which compare same entry 100 times. if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query("cb", "SELECT * FROM location", "result"); xlog("rows: $dbr(result=>rows) cols: $dbr(result=>cols)\n"); $var(result) = 0; while($var(i)<$dbr(result=>rows)) { sql_pvquery("cb", "SELECT ruid FROM location WHERE callid = '$ci'", "$var(dbruid)"); if(fnmatch("$var(dbruid)", "$var(ruid)")) { xlog("L_INFO", "AOR: Found match --> [$var(dbruid)] = [$var(ruid)]\n"); } else { sql_query_async("cb","DELETE FROM location WHERE ruid = '$var(dbruid)'"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', NOW(), '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } } } From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:58:29 Subject: Re: [SR-Users] msilo Ok lua works right now. I was need remove "" in the value. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:29:39 Subject: Re: [SR-Users] msilo Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Rtpproxy settings without NAT in Kamailio
by shantanu saha 03 Dec '16

03 Dec '16

Hello, I am trying to configure rtpproxy with Kamailio. kamailio.cfg is already configured (default) for rtpproxy. I just add the following line at the top of the cfg file. *#!define WITH_NAT* After that rtpproxy is working in local network. Both client and sip server are running in local network. But when I run this in public network, rtpproxy didn't trigger. Rtppoxy and Kamailio are running in same public IP, Sip client also running in another *public IP. *So there is *no NAT* *.* Both local and public version was Centos 6 Kamailio version : 4.4 rtpproxy version: 2.1 Note: udp connection between rtpproxy and Kamailio was established (both in public and private network) Thanks shantanu saha

2 1

Dual Stack Environments
by Colin Morelli 02 Dec '16

02 Dec '16

Hey all, Sorry for two questions out at the same time! I'm using kamailio in front of FS, and I'm about to enable IPv6, but I have a question that I'm not entirely clear on. What's the proper way to forward traffic using the same interface that it came in on? Specifically, if a client connects to kamailio on IPv6, it should connect to FS on IPv6, so that FS knows to advertise its v6 address in the SDP. The same should be true for v4. Is this possible? Best, Colin

2 1

Local messages are malformed in Kamailio 4.4.X
by Helio Okuyama 01 Dec '16

01 Dec '16

Local messages are malformed in Kamailio 4.4.X. For example after receiving an error response (603) to an INVITE message, the ACK is generated with header Max-Forwards concatenated to header CSeq and double CRLF, which is wrong. Via: SIP/2.0/TLS 173.16.10.61:5061 ;branch=z9hG4bKc782.5b960c7c48a4363a0126963daa797eb0.0;i=7 From: "22861"<sip:22861@173.16.10.61;transport=tls>;tag=snl_OIPEOzJUNy Call-ID: 61b500a71b510a1Pj61qq1k9dPW To: <sip:551138174108@174.16.4.108:5061;transport=tls>;tag=435012226 CSeq: 2351 ACKMax-Forwards: 70 User-Agent:... The fix for that seems to be just the order of append_str in function build_local() from tm\t_msgbuilder.c. append_str( p, method, method_len ); + append_str( p, CRLF, CRLF_LEN ); append_str( p, MAXFWD_HEADER, MAXFWD_HEADER_LEN ); - append_str( p, CRLF, CRLF_LEN );

4 7

help with string modification
by Sergey Basov 01 Dec '16

01 Dec '16

Hello All. I have some troubles with upstream sip switch. It ignores SIP packets which contains: User-Agent: FPBX-2.11.0(11.17.1) or Server: User-Agent: FPBX-2.11.0(11.17.1) If space is present before first "(" then sip switch works as expected So my question is: how corektly make analyze and modify this headers using kamailio? In which routes i must parse it? i need such modifications: User-Agent: FPBX-2.11.0(11.17.1) -> User-Agent: FPBX-2.11.0 (11.17.1) Server: User-Agent: FPBX-2.11.0(11.17.1) -> Server: User-Agent: FPBX-2.11.0 (11.17.1) Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 4

Re: [SR-Users] TSILO question
by Federico Cabiddu 01 Dec '16

01 Dec '16

Hi, about your questions: Q1: yes it's feasible as long as the two Registrars (Kamailio) have the location information. Please see my reply on a thread on this topic: http://lists.sip-router.org/pipermail/sr-users/2016-February/091612.html Q2: given that TSILO depends on the users' location you have to put it in the Registrar. Best regards, Federico On Wed, Nov 30, 2016 at 5:32 AM, C. J Hsu (徐熙杰) <CJ.Hsu(a)quantatw.com> wrote: > hi Federico > > i'm stuck at TSILO and wondering if you could give me some advises. > > Q1: > > in lab1, I'm having two Kamailios with proxy and registrar together, UA1 > and UA2. > UAs go thru hardware load-balancing device (F5) to register themselves in > Kamailio, > > I wonder, UA1 calling to UA2, invite suspended in Kamailio 1, waiting for > UA2; > and when UA2 come to register in Kamailio 2, TSILO in Kamailio 1 resumes > the invite and sending it to Kamailio 2, is this a no problem scenario? > > Q2: > > if we decide to separate Proxy and Registrar in different Kamailio, when > should I put TSILO in? > > > RGDS > > C J >

1 0

Real error?
by Olvera, Victor 01 Dec '16

01 Dec '16

Is this a real error? 9(1899) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket Great regards, Victor Olvera victor.olvera(a)verizonwireless.com<mailto:victor.olvera@verizonwireless.com> mobile # 001-214-923-9405

3 2

In dialog messages cannot be routed after re-registration with outbound and gruu
by Robert 01 Dec '16

01 Dec '16

Hi, I'm using outbound with gruu enabled and with edge proxy, registrar and auth proxy all running as separate entities. My problem is once a dialog is formed between two UAs, if one side re-registers and gets assigned a new flow token, all subsequent in-dialog messages cannot be routed because the flow token in the Route set is no longer valid. What is the proper way to update the binding? I read in this thread <http://sip-router.1086192.n5.nabble.com/Outbound-and-Registrar-No-binding-u…> that "using GRUU means that Outbound is also used, so the Contact has a +sip.instance and reg-id params which are inspected by the registrar to create/update/delete a binding". How is the binding updated exactly? Any help would be appreciated. -- View this message in context: http://sip-router.1086192.n5.nabble.com/In-dialog-messages-cannot-be-routed… Sent from the Users mailing list archive at Nabble.com.

1 0

Kamailio Processes Stop Responding
by Peter DiSanto 30 Nov '16

30 Nov '16

We have a Kamailio 4.2 machine with a custom config file we've been using for the better part of two years without issue. Suddenly we've run into an issue where Kamailio just stops responding to any INVITEs, and stops any logging. The processes keep running but stop doing anything. It is a full production machine which naturally causes problems. I have the memlog from the restart after a hang if that helps, but I presume it is too large for this list. We have increased shared memory to 1536 and Pkg memory to 32 but that didn't help, and in any case the shmem stats look OK to us. It is a full production machine doing about 200 CPS, the load is barely measurable but turning logging to debug will surely kill it. We use TOPOH and do get lots of log entries like : /usr/sbin/kamailio[17865]: ERROR: <core> [parser/parse_via.c:2048]: parse_via(): ERROR: parse_via: bad char <▒> on state 100 /usr/sbin/kamailio[17865]: ERROR: <core> [parser/parse_via.c:2738]: parse_via(): ERROR: parse_via on: <▒#010d▒▒_▒#0117Jt▒#015#012X> /usr/sbin/kamailio[17865]: ERROR: <core> [parser/parse_via.c:2744]: parse_via(): ERROR: parse_via: via parse error /usr/sbin/kamailio[17865]: ERROR: topoh [th_msg.c:438]: th_unmask_via(): cannot find cookie in via2 /usr/sbin/kamailio[17865]: ERROR: topoh [th_mask.c:165]: th_mask_decode(): invalid input string"642525435_130081688(a)xxx.xxx.xxx.xxx" /usr/sbin/kamailio[17865]: ERROR: topoh [th_msg.c:484]: th_unmask_callid(): cannot decode callid and CRITICAL: sl [../../forward.h:279]: msg_send(): unknown proto 0 However, I believe these are not new and do not seem to affect overall call processing. Anyone have any idea what to look for or how to figure this out? shmem:fragments = 4019 shmem:free_size = 1538943936 shmem:max_used_size = 84349120 shmem:real_used_size = 71668800 shmem:total_size = 1610612736 shmem:used_size = 54855384

2 1

Offload SSL from backends with Kamailio
by Vladyslav Zakhozhai 30 Nov '16

30 Nov '16

Hi, I'm trying to understand the best (or reasonable) approach of offloading SSL encryption from backend to Kamailio. Let me simplify a little bit: UAC == SIP/TLS ==> Kamailio == SIP/UDP ==> FreeSWITCH My main problem is in Contact header of SIP packet which passes through Kamailio SIP proxy and remains unmodified. For example, REGISTER request. There is FreeSWITCH backend which is registrar server as well. UAC send REGISTER request to it through Kamailio SIP proxy via SIP/TLS. This request dispatches to backend(s) by Kamailio with dispatcher module. Backend does not configured to support TLS. In this case everything works fine: I see REGISTER requests on FreeSWITCH. But Contact header of SIP message which is passing Kamailio remains unmodified. And as result I see on FreeSWITCH something like the following: Call-ID: Jpmjp4ruHI User: user_name@domain_name Contact: "" <sip:user_name@uac_ip :27026;transport=tls;fs_path=sip%3Akamailio_ip%3A5060> Agent: Linphone/3.10.2 (belle-sip/1.5.0) Status: Registered(TLS)(unknown) EXP(2016-11-28 11:48:28) EXPSECS(110) Ping-Status: Reachable Ping-Time: 0.00 Host: kamailio_host IP: kamailio_ip Port: 5060 Auth-User: unknown Auth-Realm: domain_name MWI-Account: user_name@domain_name As a result FreeSWITCH tries to originate call over SIP/TLS and it fails because FreeSWITCH does not configured to work with TLS. I want to understand what is correct workaraound of this issue. Do I need to modify Contact header manually on kamailio host and this is right approach? Or kamailio in case of correct config rewrites this header itself? If parts of my kamailio config would be useful I will post it later. Thanks in advance. -- С уважением, Владислав Захожай

4 8

kazoo module
by Ján Füri 30 Nov '16

30 Nov '16

Hello, I am trying to install kazoo module. According to documentation I have installed libraries : librabbitmq, libjson, libuuid, but It always fails on missing json.h : CC (gcc) [M kazoo.so] kz_trans.o In file included from kz_trans.c:53:0: kz_json.h:12:18: fatal error: json.h: No such file or directory #include <json.h> ^ compilation terminated. ../../Makefile.rules:97: recipe for target 'kz_trans.o' failed make[1]: *** [kz_trans.o] Error 1 Makefile:511: recipe for target 'modules' failed make: *** [modules] Error 1 Can anybody help me ? BR Jan

3 3

Re: [SR-Users] Kamailio IMS virtual machine
by Franz Edler 29 Nov '16

29 Nov '16

Hi Rafael, There should not be a problem with the VMware mage. Regarding your question: you have to check kamailio-pcscf.cfg to recognize when and it how forward the SIP message towards the I-CSCF. I guess it is the statement "t_relay();" in line 896. You may provide a core tracefile (generated with tcpdump), then I may have aquick look ate your ise. BR Franz Von: Rafi Aizenstain [mailto:rafi_a@cellowireless.com] Gesendet: Montag, 28. November 2016 16:12 An: franz.edler(a)technikum-wien.at Betreff: Kamailio IMS virtual machine Hello Mr. Franz I'm trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ But when running the machine as-is I keep getting errors from the P-CSCF module and it doesn't forward the SIP message towards the I-CSCF. Please see the following log: root@server:~# kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2096) INFO: <core> [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: server.net1.test [10.0.0.9]:4060 Aliases: *: pcscf.net1.test:* WARNING: no fork mode 0(2096) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(2096) INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available 0(2096) INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 512 0(2096) INFO: <core> [udp_server.c:150]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(2096) INFO: <core> [udp_server.c:200]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(2096) WARNING: ims_usrloc_pcscf [usrloc_db.c:59]: connect_db(): DB connection already open... continuing 0(2096) INFO: <script>: [REGISTER] from [sip:bob@net1.test] to [sip:bob@net1.test] 0(2096) ERROR: <script>: REGISTER (sip:bob@net1.test (10.0.0.200:59721) to sip:bob@net1.test, 7013b8fc-11a0-2b70-3654-9ce68424e9a7) Looking into the configuration files I couldn't understand how does the P-CSCF determine that it has to forward the SIP message towards the I-CSCF module. Are you aware of any issue with the VMware machine you have shared. Please advise. Best Regards, Rafael A.

2 1

LCR modules troubles
by Laura 29 Nov '16

29 Nov '16

Dear list, i have a strange situation with module LCR on kamailio 4.3.6. A customer send a call to my first kamailio and i forward it with some tech prefix to my supplier. Here the 2 correct invites.. customer is sending to me with prefix 9001 14:34:29.155719 CUSTOMER_IP.101:5060 -> EU_KMAILIO_IP.40:5060 INVITE sip:900133139864069@EU_KMAILIO_IP.40 SIP/2.0. Record-Route: <sip:CUSTOMER_IP.101;lr;ftag=33F431FC-2103;vst=AAAAAAwFBQQFAwEACgkFDQgADXADCG4DDQEABQUWHwYGLjEwMQ-->. Via: SIP/2.0/UDP CUSTOMER_IP.101;branch=z9hG4bK2d4.4853821c2be0bf7d75377fce7683f593.0. Via: SIP/2.0/UDP CUSTOMER_IP.108:5060;branch=z9hG4bK504DC634. Remote-Party-ID: <sip:441753825186@CUSTOMER_IP.108>;party=calling;screen=yes;privacy=off. From: <sip:441753825186@CUSTOMER_IP.108>;tag=33F431FC-2103. To: <sip:900133139864069@EU_KMAILIO_IP.40>. Date: Mon, 28 Nov 2016 14:34:29 GMT. Call-ID: 9A8A4545-B4AE11E6-9238E2D3-6BBD43E(a)CUSTOMER_IP.108. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2592672861-3031306726-2677997592-0419390928. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 69. ..... and i forward to my supplier U 2016/11/28 14:34:29.157516 EU_KMAILIO_IP.40:5060 -> EU_SUPPLIER_IP.198:5060 INVITE sip:00233139864069@EU_SUPPLIER_IP.198 SIP/2.0. Record-Route: <sip:EU_KMAILIO_IP.40;lr;ftag=33F431FC-2103;did=ff5.6f01;vst=AAAAAAkAAgIAAgIKAQ4CBAYPeXIECQAABAcAAwQIAAUJOA--;nat=yes>. Record-Route: <sip:CUSTOMER_IP.101;lr;ftag=33F431FC-2103;vst=AAAAAAwFBQQFAwEACgkFDQgADXADCG4DDQEABQUWHwYGLjEwMQ-->. Via: SIP/2.0/UDP EU_KMAILIO_IP.40:5060;branch=z9hG4bK2d4.782830bfdde5e873d3dc6a71bcbba70f.0. Via: SIP/2.0/UDP CUSTOMER_IP.101;rport=5060;branch=z9hG4bK2d4.4853821c2be0bf7d75377fce7683f593.0. Via: SIP/2.0/UDP CUSTOMER_IP.108:5060;branch=z9hG4bK504DC634. Remote-Party-ID: <sip:441753825186@CUSTOMER_IP.108>;party=calling;screen=yes;privacy=off. From: <sip:441753825186@CUSTOMER_IP.108>;tag=33F431FC-2103. To: <sip:00233139864069@EU_SUPPLIER_IP.198>. Date: Mon, 28 Nov 2016 14:34:29 GMT. Call-ID: 9A8A4545-B4AE11E6-9238E2D3-6BBD43E(a)CUSTOMER_IP.108. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2592672861-3031306726-2677997592-0419390928. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 68. Timestamp: 1480343669. Contact: <sip:441753825186@CUSTOMER_IP.108:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 310. On both invites the To field respect the correct information from INVITE so that the kamailio CDR is working correct. After that the supplier reply me 503 and call is come back to my kamailio. LCR modules send the call to second supplier that on this case is my second box kamailio on another countrie.. Here there is this problem that i signed on RED U 2016/11/28 14:34:43.191618 EU_KMAILIO_IP.40:5060 -> EU_SUPPLIER2_IP.242:5060 *INVITE sip:905233139864069@EU_SUPPLIER2_IP.242 SIP/2.0.* Record-Route: <sip:EU_KMAILIO_IP.40;lr;ftag=33F431FC-2103;did=ff5.6f01>. Record-Route: <sip:CUSTOMER_IP.101;lr;ftag=33F431FC-2103;vst=AAAAAAwFBQQFAwEACgkFDQgADXADCG4DDQEABQUWHwYGLjEwMQ-->. Via: SIP/2.0/UDP EU_KMAILIO_IP.40:5060;branch=z9hG4bK2d4.782830bfdde5e873d3dc6a71bcbba70f.1. Via: SIP/2.0/UDP CUSTOMER_IP.101;rport=5060;branch=z9hG4bK2d4.4853821c2be0bf7d75377fce7683f593.0. Via: SIP/2.0/UDP CUSTOMER_IP.108:5060;branch=z9hG4bK504DC634. Remote-Party-ID: <sip:441753825186@CUSTOMER_IP.108>;party=calling;screen=yes;privacy=off. From: <sip:441753825186@CUSTOMER_IP.108>;tag=33F431FC-2103. *To: <sip:900133139864069@EU_KMAILIO_IP.40>.* Date: Mon, 28 Nov 2016 14:34:29 GMT. Call-ID: 9A8A4545-B4AE11E6-9238E2D3-6BBD43E(a)CUSTOMER_IP.108. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2592672861-3031306726-2677997592-0419390928. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 68. Timestamp: 1480343669. Contact: <sip:441753825186@CUSTOMER_IP.108:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. .... The LCR modules make the new INVITE using the original To field and don't rewrite it like on the first request.. How is that possibile ? Any idea ? Best regards Laura

3 2

XAVP Null
by Linux Vince 29 Nov '16

29 Nov '16

Hi, I am sure i am doing something very basic wrongly. Here is my config file snippet $xavp(caller_peer[0]=>accountname) = "AAA"; xlog("L_WARN", "XAVP ACCOUNT NAME $xavp(caller_peer[0]=>accountname) \n"); No matter what i set this XAVP at, it always prints null in log file as below WARNING: <script>: XAVP ACCOUNT NAME <null> Please help how i can set XAVP and use it further in config file.

3 2

MSRP auth does not work with blink
by Pranathi Venkatayogi 29 Nov '16

29 Nov '16

Hi, I am trying to use MSRP feature of blink client. In the server config when auth is challenged, I see the following error in the client. Can you please advise if I am configuring the server right? Note: the return value of pv_www_authenticate is "-5". 13 2016-11-07 17:57:26.494124 [blink.exe 24976]: Closed connection to 10.0.16.51:2855 ('NotificationProxyLogger' object has no attribute 'received_illegal_data') Find attached full set of client logs. Here is the relevant portion of kamailio.cfg: if (!pv_www_authenticate("MY_DOMAIN", "$var(passwd)", "0", "$msrp(method)")) { www_challenge("MY_DOMAIN", "1"); xlog("L_ERR", "MSRP auth failed with: $var(retValue)\n"); exit; } Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

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Problems with Siremis and Kamailio - PHP7
by Diogenes Aviles 29 Nov '16

29 Nov '16

I have Kamailio 4.4 installed in ubuntu 16.04.03, with Siremis 4.2.0, PHP7 and apache2. All is OK, but when I try to install the Siremis Wizard, the step 2 (Data Base Configuration) doesn't work, and there is the message "Something Went Wrong". I tried all the option on Step 2, but the problem continuous. Then I can't go to step 3 Application Configuration. The Siremis DataBase is created and the user also. I appreciate any help. -- Diogenes

2 1

Siremis v4.3.0 Released
by Daniel-Constantin Mierla 29 Nov '16

29 Nov '16

Hello, a new release of Siremis, the web management interface for Kamailio, has been released recently. The main work was towards making it compatible with PHP 7 and the new constraints from MySQL 5.7+. The announcement is available at: - http://siremis.asipto.com/2016/11/26/siremis-v4-3-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World 2017 - May 8-10, in Berlin - www.kamailioworld.com

1 0

custom version database/table
by David Villasmil 28 Nov '16

28 Nov '16

Hello all, I have a databse where i created a "version" table only with a version, a dispatcher records. I only want to use dispatcher on a db. I don't know if kamailio is looking for the version tables somewhere else, or it does find it in my database (not named kamailio) but the format is wrong. I used: CREATE TABLE `version` ( `table_name` VARCHAR(32) NOT NULL, `table_version` INT UNSIGNED DEFAULT 0 NOT NULL, CONSTRAINT table_name_idx UNIQUE (`table_name`) ); INSERT INTO version (table_name, table_version) values ('version','1'); INSERT INTO version (table_name, table_version) values ('dispatcher','4'); To create an populate the table, but kamailio returns: mod_init(): could not initiate a connect to the database but it does connect properly... any thoughts? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Recent problem with fix_nated_sdp and kamailio 4.4.4
by Anthony Messina 28 Nov '16

28 Nov '16

Prior to the upgrade to Kamailio 4.4.4, I was using 4.4.3 with updates through 81df84b from the 4.4 branch and the following fix_nated_sdp call worked through the initial INVITE and when a voicemail branch was appended from the failure_route. route[NATMANAGE] { ... # RTPEngine is not needed but if STUN addresses are being used on our # internal network, rewrite the addresses to the internal source address if(!isbflagset(FLB_RTPENGINE) && has_body("application/sdp")) { if(dst_ip==10.1.1.2 && compare_pure_ips($sel(contact.uri.host), "<EXTERNAL_IP>")) { fix_nated_sdp("10"); } } ... } Since the upgrade to 4.4.4, when the original INVITE fails over to voicemail (branch appended), fix_nated_sdp creates the SDP below, doubling the replaced IP addresses. Asterisk replies with 488, as the media address is completely invalid. I need something like the above (which worked for the request and the reply) but can't figure out what the problem is. v=0 o=Zoiper 0 0 IN IP4 10.1.1.18210.1.1.182 s=Zoiper c=IN IP4 10.1.1.18210.1.1.182 t=0 0 m=audio 63732 RTP/SAVP 9 3 0 97 101 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNDOcpFdtqu5DQ== a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNDOcpFdtqu5DQ== a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNA= a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNA= a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDI a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDI a=oldmediaip:<EXTERNAL_IP> a=oldmediaip:<EXTERNAL_IP> a=oldmediaip:<EXTERNAL_IP> a=oldmediaip:<EXTERNAL_IP> -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

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kamailio core dumps once every few days
by davy van de moere 28 Nov '16

28 Nov '16

Gents, On a not really loaded kamailio (less than 50 users at the moment), I'm getting a core dump every few days. I've tried multiple verisons of Kamailio, 4.4.x , 4.3.x and also on 5.0.0-dev7 . I can't reproduce it adhoc, but every few days it happens... Is that something someone wants to jump in? grtz, davy Version: version: kamailio 5.0.0-dev7 (x86_64/linux) 7360b0 [New LWP 29728] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -f /usr/local/etc/kamailio//k'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f92c5a0be98 in __dialog_sendpublish (dlg=0x7f92c0e3a6c0, type=64, _params=0x7f92c0423500 <params>) at pua_dialoginfo.c:264 264 uri = dlginfo->to_uri; (gdb) backtrace #0 0x00007f92c5a0be98 in __dialog_sendpublish (dlg=0x7f92c0e3a6c0, type=64, _params=0x7f92c0423500 <params>) at pua_dialoginfo.c:264 #1 0x00007f92c01e2563 in run_dlg_callbacks (type=64, dlg=0x7f92c0e3a6c0, req=0x7f92cc6d5db8, rpl=0x0, dir=1, dlg_data=0x0) at dlg_cb.c:256 #2 0x00007f92c01a92b1 in dlg_terminated (req=0x7f92cc6d5db8, dlg=0x7f92c0e3a6c0, dir=1) at dlg_handlers.c:366 #3 0x00007f92c01b610f in dlg_onroute (req=0x7f92cc6d5db8, route_params=0x7ffc7de27ba0, param=0x0) at dlg_handlers.c:1409 #4 0x00007f92c9292171 in run_rr_callbacks (req=0x7f92cc6d5db8, rr_param=0x7ffc7de27c20) at rr_cb.c:96 #5 0x00007f92c9287ac2 in after_loose (_m=0x7f92cc6d5db8, preloaded=0) at loose.c:928 #6 0x00007f92c9288168 in loose_route (_m=0x7f92cc6d5db8) at loose.c:962 #7 0x00007f92c928b7ee in w_loose_route (msg=0x7f92cc6d5db8, p1=0x0, p2=0x0) at rr_mod.c:266 #8 0x000000000041f5d9 in do_action (h=0x7ffc7de285b0, a=0x7f92cc5f7a70, msg=0x7f92cc6d5db8) at action.c:1054 #9 0x000000000042bcde in run_actions (h=0x7ffc7de285b0, a=0x7f92cc5f7a70, msg=0x7f92cc6d5db8) at action.c:1552 #10 0x000000000042c383 in run_actions_safe (h=0x7ffc7de29c60, a=0x7f92cc5f7a70, msg=0x7f92cc6d5db8) at action.c:1620 #11 0x000000000061fe86 in rval_get_int (h=0x7ffc7de29c60, msg=0x7f92cc6d5db8, i=0x7ffc7de28cc0, rv=0x7f92cc5f7bc8, cache=0x0) at rvalue.c:912 #12 0x00000000006243c3 in rval_expr_eval_int (h=0x7ffc7de29c60, msg=0x7f92cc6d5db8, res=0x7ffc7de28cc0, rve=0x7f92cc5f7bc0) at rvalue.c:1910 #13 0x000000000041f0d1 in do_action (h=0x7ffc7de29c60, a=0x7f92cc600430, msg=0x7f92cc6d5db8) at action.c:1030 #14 0x000000000042bcde in run_actions (h=0x7ffc7de29c60, a=0x7f92cc600430, msg=0x7f92cc6d5db8) at action.c:1552 #15 0x000000000041f548 in do_action (h=0x7ffc7de29c60, a=0x7f92cc6006d0, msg=0x7f92cc6d5db8) at action.c:1045 #16 0x000000000042bcde in run_actions (h=0x7ffc7de29c60, a=0x7f92cc6006d0, msg=0x7f92cc6d5db8) at action.c:1552 #17 0x000000000041c0fb in do_action (h=0x7ffc7de29c60, a=0x7f92cc545730, msg=0x7f92cc6d5db8) at action.c:678 #18 0x000000000042bcde in run_actions (h=0x7ffc7de29c60, a=0x7f92cc537760, msg=0x7f92cc6d5db8) at action.c:1552 #19 0x000000000042c44b in run_top_route (a=0x7f92cc537760, msg=0x7f92cc6d5db8, c=0x0) at action.c:1641 #20 0x00000000005726a2 in receive_msg ( buf=0xa1c200 <buf> "BYE sip:echo@123.48.11.215:5070;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 0.0.0.0:5060;rport;branch=z9hG4bKPj26b505ba-7ede-4198-a917-bde966b6d8b6\r\nMax-Forwards: 69\r\nFrom: sip:192.168.60.1;tag=9a10d8ab-"..., len=462, rcv_info=0x7ffc7de29fb0) at receive.c:264 #21 0x000000000047bcdc in udp_rcv_loop () at udp_server.c:507 #22 0x0000000000509de1 in main_loop () at main.c:1618 #23 0x00000000005112de in main (argc=5, argv=0x7ffc7de2a478) at main.c:2639 (gdb)

2 2

Balancer with Dispacther Module
by Rodrigo Moreira 28 Nov '16

28 Nov '16

I need help. I'm building a load balancer using the dispatcher. It does NAT with too many SIP Servers. But when the balancer receives an ACK from UAC, it immediately returns the message "404 Not Here" to UAC. Help me solve this, since this message (ACK) has to be forwarded to SIP Servers and finally to UAS to start an RTP session. I thank you. Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

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Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Hello Daniel, I am trying more simple example of $dbr in REGISTER first. Insert into location temporary data. The problem is not looping though all entries in database only last one. Which compare same entry 100 times. if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query("cb", "SELECT * FROM location", "result"); xlog("rows: $dbr(result=>rows) cols: $dbr(result=>cols)\n"); $var(result) = 0; while($var(i)<$dbr(result=>rows)) { sql_pvquery("cb", "SELECT ruid FROM location WHERE callid = '$ci'", "$var(dbruid)"); if(fnmatch("$var(dbruid)", "$var(ruid)")) { xlog("L_INFO", "AOR: Found match --> [$var(dbruid)] = [$var(ruid)]\n"); } else { sql_query_async("cb","DELETE FROM location WHERE ruid = '$var(dbruid)'"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', NOW(), '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } } } From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:58:29 Subject: Re: [SR-Users] msilo Ok lua works right now. I was need remove "" in the value. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:29:39 Subject: Re: [SR-Users] msilo Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Ok lua works right now. I was need remove "" in the value. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:29:39 Subject: Re: [SR-Users] msilo Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

tm:local-request does not intercept locally generated CANCEL
by Andy 25 Nov '16

25 Nov '16

should tm:local-request intercept CANCELs, if not then where can the locally generated CANCELs from t_cancel.c be intercepted -- View this message in context: http://sip-router.1086192.n5.nabble.com/tm-local-request-does-not-intercept… Sent from the Users mailing list archive at Nabble.com.

2 2

protocol/port mismatch
by Yuriy Gorlichenko 25 Nov '16

25 Nov '16

Hi. Have strange issue Calling from kamailio through WS invite and resposes goes ok but with ACK to WS client has this issue get_send_socket2(): protocol/port mismatch (forced tls:kamailioIP:4443, to udp:MyWSClientIP:65451) Guess kamailio thinks that it is UDP client because $ru is sip:10002@MyWSClientIP:65451 But can not understand how to fix this...

2 2

Re: [SR-Users] msilo
by Slava Bendersky 25 Nov '16

25 Nov '16

Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ 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

2 1

Re: [SR-Users] msilo
by Slava Bendersky 25 Nov '16

25 Nov '16

Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] msilo
by Slava Bendersky 25 Nov '16

25 Nov '16

Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] msilo
by Slava Bendersky 24 Nov '16

24 Nov '16

Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Wednesday, 23 November, 2016 21:10:53 Subject: Re: [SR-Users] msilo Hello Everyone, Need another set of eyes, can't find how to To: got << in uri. Nov 23 19:50:55 cavprx00 kamailio: 3(9625) DEBUG: <core> [parser/parse_uri.c:1280]: parse_uri(): parse_uri: bad uri, state 0 parsed: <<sip> (4) / <<sip:4200@client_public_ip:49418>;messagetype=SMS> (47) Nov 23 19:50:55 cavprx00 kamailio: 3(9625) ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: [<sip:4200@ client_public_ip :49418>;messagetype=SMS] Nov 23 19:50:55 cavprx00 kamailio: 3(9625) ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches route[MESSAGE_FORWARD] { xlog("L_INFO", "Incoming new SMS [$rm] from $tU --> $td\n"); if(fnmatch("$rs", "202") && fnmatch("$rr", "Accepted") && allow_trusted("$si", "$proto")) { $sht(b=>from) = $fu; $sht(b=>oldto) = $tu; $avp(new_to) = "<" + $sht(b=>oldto) + ">" + ";messagetype=SMS"; xlog("L_INFO", "MESSAGE: B2BUA send --> [$rs] with [$rr] from [$sht(b=>from)] to [$sht(b=>oldto)] -->[$td]\n"); } if(compare_ips("$td", "10.18.130.27")) { xlog("L_INFO", "Message from B2BUA contain domain --> $td user --> $(sht(b=>oldto){uri.user}) updating...\n"); sql_pvquery("cb", "select contact from location where username = '$(sht(b=>oldto){uri.user})'","$avp(dst)"); #$avp(new_dst) = $(hdr(Route){param.value,received}); xlog("L_INFO", "New destination --> $avp(dst)\n"); $avp(new_to) = "<" + $(avp(dst){tobody.uri}) + ">" + ";messagetype=SMS"; } if(fnmatch("$fU", "offline")) { xlog("L_INFO", "OFFLINE SMS: from [$fU] user --> $fU\n"); $avp(from) = $(sht(b=>oldto){uri.user}); xlog("L_INFO", "OFFLINE SMS: Searching destination user $avp(oexten)\n"); sql_pvquery("cb", "select contact from location where username = '$avp(from)'","$avp(dst)"); xlog("L_INFO", "OFFLINE SMS: set new destination --> $avp(dst)\n"); $avp(new_to) = "<" + $(avp(dst){tobody.uri}) + ">" + ";messagetype=SMS"; } xlog("L_INFO", "OFFLINE SMS: new AVP DST $avp(new_to)\n"); $sht(a=>to) = $avp(new_to); remove_hf("To"); insert_hf("To: $sht(a=>to)\r\n", "To"); $du = $sht(a=>to); xlog("SMS from --> $fU domain $fd\n"); xlog("-------------------------------------\n"); xlog("L_INFO", "FROM --> $avp(from)\n"); xlog("L_INFO", "OLD_TO --> $sht(b=>oldto)\n"); xlog("L_INFO", "NEW_TO --> $avp(new_to)\n"); } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 17 November, 2016 05:44:22 Subject: Re: [SR-Users] msilo Hello, if you want to fetch them for config usage is ok, otherwise m_dump() is still better, because it takes care of removing delivered messages, etc... Cheers, Daniel On 16/11/16 16:08, Slava Bendersky wrote: Hello Daniel, Yes that good alternative. I will use sql to fetch or insert staff. Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 16 November, 2016 04:37:24 Subject: Re: [SR-Users] msilo Hello, as an alternative, I think it should work by using sql_query(...) to insert whatever values one would like to have in msilo table. This might be a quick workaround until someone will code the extension to get the body via a function parameter. Cheers, Daniel On 15/11/16 16:07, Federico Cabiddu wrote: BQ_BEGIN Hi Slava, if I understand correctly your scenario, you could call t_on_failure before relaying the MESSAGE. In this way you should have access to the original message in the failure route triggered after receiving a final reply and you should be able to store it with the regular msilo functions. Hope this helps. Cheers, Federico On 15 Nov 2016 3:47 pm, "Slava Bendersky" < volga629(a)skillsearch.ca > wrote: BQ_BEGIN Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. From: "Daniel-Constantin Mierla" < miconda(a)gmail.com > To: "sr-users" < sr-users(a)lists.sip-router.org > Sent: Tuesday, 15 November, 2016 03:00:19 Subject: Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ 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 -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ 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 BQ_END BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

2 2

Howto send rtp stats infos to homer
by Oliver Roth 23 Nov '16

23 Nov '16

Hi there We are using homer to analyce and monitor traffic. I did not get a solution to send x-rtp or p-rtp stats to homer. How can I find infos about jitter, sent/received packages, lost packages, etc.? Our current infos are based on Rtpstat provided by kamailio. At the moment I do $var(xrtpstat) = $(rtpstat{s.striptail,1}); # Work the stats $var(octetsent) = $(var(xrtpstat){s.select,1, }); #octetsent / OS $var(packetsent) = $(var(xrtpstat){s.select,3, }); #PS $var(octetrcv) = $(var(xrtpstat){s.select,8, }); #OR $var(packetrcv) = $(var(xrtpstat){s.select,10, }); #PR $var(errorsent) = $(var(xrtpstat){s.select,5, }); $var(errorrcv) = $(var(xrtpstat){s.select,12, }); if ($var(octetsent) != "" || $var(packetsent) != "") { xlog(,"L_INFO", "WITHINDLG PS=$var(rtp0),PR=$var(rtp1),PL=$var(rtp3) \n"); xlog(,"L_INFO", "WITHINDLG xrtpstats=$var(xrtpstat) \r\n"); xlog(,"L_INFO", "WITHINDLG rtpstats=$rtpstat \r\n"); xlog(,"L_INFO", "WITHINDLG X-RTP-Stat: PS=$var(packetsent),OS=$var(octetsent),PR=$var(packetrcv),OR=$var(octetrcv),PL=$var(errorsent) \r\n"); #append_hf("P-RTP-Stat: $rtpstat\r\n"); append_hf("X-RTP-Stat: PS=$var(packetsent);OS=$var(octetsent);PR=$var(packetrcv);OR=$var(octetrcv);PL=$var(errorsent);JI=600;LA=40;\r\n"); } Other question: Where has this to be placed? Only on bye messages? KR, Oli

1 0

Change from in a failure route
by Nelson Migliaro 23 Nov '16

23 Nov '16

Hello, My sip vendor sends me a 403 error when the caller id (CLI) sent is not valid. I would like to set up a failure route to capture that 403 and set up a default CLI. I tried to use a uac_replace_from() in a failure route but I get: "Command cannot be used in the block" Does anyone know how can I change the CLI in a failure route how how can achieve the default CLI? Thank you

2 1

Force OpenSER to send a Stop radius accounting record in failure route.
by Jan Hazenberg 23 Nov '16

23 Nov '16

Hi all, I have a issue here on a old OpenSER 1.3 system. We have some clients that respond to a BYE message with a 481 resulting in a missing radius accounting record. I have added some code to the failure route to force accounting even if a 481 response in received: if (t_check_status("481") && $rm == "BYE") { xlog("L_INFO", "INFO: [FAILURE] 481 response detected!! Forcing accounting! - SRC=$si:$sp R=$ru ID=$ci FU=$fu M=$rm S=$rs\n"); acc_rad_request("BYE with 481 response"); } Now OpenSER sends a accounting record, but this has the attribute Acct-Status-Type set to Failed. This gives a problem with our accounting service and i would like to change it to a normal Stop record. Is there a way to change this and force OpenSER/Kamailio to send a Stop record instead of a Failed record? Thanks, Jan

2 3

SDP Codec not removed with RTPengine - but with rtpproxy it worked
by Oliver Roth 23 Nov '16

23 Nov '16

Hi there I have the following problem - I need to remove a codec in the initial INVITE. This happens since I changed from rtpproxy to rtpengine. I changed all rtpproxy_manage() to rtpengine_manage(). Originating INVITE with the "clearmode" m=audio 9196 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 I do the following in the config route[SDP] { xlog(, "L_INFO", "SDP: request method $rm"); # remove CLEARMODE if Colt if($avp(s:todirection) =~"^MyCarrier+") { if (is_method("INVITE")) { xlog(, "L_INFO", "SDP remove: request method $rm"); msg_apply_changes(); sdp_remove_codecs_by_name("CLEARMODE"); #sdp_remove_codecs_by_id("125"); } } } This block gets hit as I can see with the xlog entry. In the sent INVITE the "Clearmode" is still in the sdp header a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 So with rtpproxy - it worked With rtpengine not any more ... Any idea? KR, Oli

3 10

IMS related : ims_usrloc_pcscf : "lookup_check_received" setting is useless now
by Dragos Oancea 23 Nov '16

23 Nov '16

Hi The setting "lookup_check_received" is documented here : http://www.kamailio.org/docs/modules/4.4.x/modules/ims_usrloc_pcscf.html#id… But looking through the code I saw it's not used anymore, although it can be passed as a parameter. I was wondering if this is an accident or something intentional. I found some more details here: http://lists.sip-router.org/pipermail/sr-dev/2014-June/024371.html Cheers, Dragos

2 1

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