- sr-users - kamailio.org

kamailio with cassandra
by Никитенко Виталий 08 Feb '17

08 Feb '17

Hi! I want start kamailio with db_cassandra. do as it written in https://www.kamailio.org/docs/modules/4.2.x/modules/db_cassandra.html. Then trying kamctl start, failed with error "table version does not exist!". >From CQLSH table is view cqlsh:kamailio> Describe tables; version location cqlsh:kamailio> select * from version; table_name | table_version ------------+--------------- version | 1 Tell me please wherein may be trouble? Full log below ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/location ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/version INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.allow INFO: permissions [permissions.c:608]: mod_init(): default allow file (/usr/local/etc/kamailio//permissions.allow) not found => empty rule set INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.deny INFO: permissions [permissions.c:617]: mod_init(): default deny file (/usr/local/etc/kamailio//permissions.deny) not found => empty rule set INFO: misc_radius [misc_radius.c:164]: mod_init(): initializing... ERROR: db_cassandra [dbcassa_base.cpp:449]: cassa_translate_query(): table version does not exist! ERROR: db_cassandra [dbcassa_base.cpp:962]: db_cassa_query(): Failed to query Cassandra cluster ERROR: <core> [db.c:401]: db_table_version(): error in db_query ERROR: <core> [db.c:440]: db_check_table_version(): querying version for table location ERROR: usrloc [dlist.c:640]: register_udomain(): error during table version check. ERROR: registrar [reg_mod.c:554]: domain_fixup(): failed to register domain ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:982 ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:987 ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:992

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Kamailio + Asterisk with multidomain
by Tomas Zanet 07 Feb '17

07 Feb '17

Hello, thanks to this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I successfully installed Asterisk 11.6 and Kamailio 4.4.4 on the same machine. Everything works fine. Right now I would like to extend this scenario adding MULTIDOMAIN support, which is not enabled by default Do you see any problems / technical limitations to add multidomain support in this scenario? Before enhancing Kamailio cfg file and Asterisk configuration I would like to know: is this possible? As far as I know Kamailio works fine with multidomain support but I don’t know if Asterisk or both processes can support it. Thanks in advance Regards, T. Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

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DMQ Cluster - hosts loose network connectivity
by José Seabra 07 Feb '17

07 Feb '17

Hello there, My DMQ cluster has 4 nodes and by some reason 2 of them lost the network connectivity for long time(~ 4 hours), after the network of these 2 nodes come back they didn't get connected on DMQ cluster automatically, i had to restart kamailio to get them again on DMQ list. My doubts here are: - Is this an expected behavior of DMQ module? - Is there any way of put them again on DMQ bus without need restart kamailio? Thank you Best Regards José Seabra

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Kamailio parse failed message
by Hai Bui Duc Ha 07 Feb '17

07 Feb '17

Hi all, I have problem when make call with my Android mobile use PJSIP library. Scenario: my client -> Kamailio -> Freeswitch (media server) -> another client (soft phone on Windows) my client: + use Bluestack + Capture via Wireshark + use Wifi Issue: The call will be drop after ~ 30 second. I see the error on Kamailio: *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<p:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0#015#012Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias#015#012Max-Forwards: 70#015#012From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO#015#012To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Abuiduchahai(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj#015#012Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB#015#012CSeq: 29055 ACK#015#012Route: <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>#015#012Content-Length: 0#015#012#015#012>* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [receive.c:129]: receive_msg(): core parsing of SIP message failed (49.156.54.54:50785/2 <http://49.156.54.54:50785/2>)* Seem to the server error when parse (on INVITE SDP) *a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16* (on new ACK message) *ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0* *Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias* *Max-Forwards: 70* *From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO* *To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Abuiduchahai(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj* *Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB* *CSeq: 29055 ACK* *Route: <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>* *Content-Length: 0* I think the SIP message is fragmented but when resume package is not correct. Do you have any advice ? Thank you for watching ! Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

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FOSDEM 2017
by Daniel-Constantin Mierla 07 Feb '17

07 Feb '17

Hello, another edition of FOSDEM is approaching, about 2 weeks left: - https://fosdem.org/2017/ Couple of devs and many Kamailio friends will be around. Like past year, there is a Realtime Communications devroom, this edition on Sunday. Very interesting presentation for our ecosystem: Olle giving a presentation about IPv4/IPv6, Inaki introducing its SFU media server project, Jose talking about JsSIP, Daniel debating about fundraising FreeRTC, Lorenzo with Janus SIP-WebRTC gateway, the other Lorenzo with Homer Sipcature, Saul with Jitsi, Dan with cgrates, Matt with Asterisk, Giovanni with FreeSwitch... Schedule at: - https://fosdem.org/2017/schedule/track/real_time_communications/ On Sunday morning, part of Lua devroom, I will present about using Lua for building RTC services with Kamailio: - https://fosdem.org/2017/schedule/track/lua/ - https://fosdem.org/2017/schedule/event/luartcserviceskamailio/ During the past editions we organized a dinner on Saturday evening. Shall we attempt to do it again in advance this year? Or should we do it on spot based on the weather and mood at the end of Saturday? 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

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kamailio proxy behind firewall
by JBF 07 Feb '17

07 Feb '17

Hello, we have the following Configuration for our kamailio installation (we are using TLS and not udp) (1) F5 Firewall (configured as message fowarding), opening a TLS server on the outside (2) SIP proxy, with a TLS server accessed by the F5 . The SIP proxy doesnt see the F5 TLS server (3) SIP registrar REGISTER works find We have the following issue on INVITE: A sends an INVITE to B. The Registrar patches the R-URI with the content of location, which contains the publi ip of the Device (because the device used stun) we force the routing from registrar to proxy by using t_relay (SIP_PROXY_IP) /The proxy tries to route to this R-URI, which is not visible/ I am not sure how to fix that: Record Route is for a true sip proxy, but the Firewall does not have an server facing the SIP proxy: the sip proxy needs to find the proper client socket opened at register to route the INVITE We have arranged for the Firewall to add its own Via, but if i understand correctly, this is used for replies, and here we are dealing with a request forwarding, and t_relay uses the r-ruri to route requests. IT might be why REGISTER works correctly (ie the 200 OK is routed correctly from proxy to firewall) I could arrange for the location table to contain the private ip and port of the firewall connection (through the use of the received/rport info inserted in the Via by the proxy ) That would mean, however that the contact of the user will contain the private interface of the F5 which i found weird. How do you think i should proceed ? any advices are welcome Thank you -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamailio-proxy-behind-firewall-tp15… Sent from the Users mailing list archive at Nabble.com.

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RTPEngine disable/enable crash kamailio
by Sergey Basov 07 Feb '17

07 Feb '17

Hi All. I ave just try to temporary disable 1 of 2 RTPEngines. It was disabled successfully all media dialogs was finished. But when I have try to enable it I have kamailio crashed with error: CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 121 Does anyone have such issue? kamailio 4.4.5 on RHEL 7.2 x86_64 rtpengine 4.4.0 on RHEL 7.2 x86_64 # kamctl fifo nh_show_rtpp all which: no gdb in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/) udp:10.1.23.24:2223:: set=0 index:: 0 disabled:: 0 weight:: 1 recheck_ticks:: 0 udp:10.1.23.25:2223:: set=0 index:: 1 disabled:: 0 weight:: 1 recheck_ticks:: 0 after # kamctl fifo nh_enable_rtpp udp:10.1.23.25:2223 0 it was successfully disabled with media going on after # kamctl fifo nh_enable_rtpp udp:10.1.23.25:2223 1 I got kamailio crash with error CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 121 sorry I does not have coredump, it is production system. my config regarding rtpengine: modparam("rtpengine", "rtpengine_allow_op", 1) modparam("rtpengine", "rtpengine_sock", "udp:10.1.23.24:2223 udp:10.1.23.25:2223") Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

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execute config route by ctp command?
by Juha Heinanen 07 Feb '17

07 Feb '17

Is there a way to execute a named config route, such as an event_route, by an rpc command? -- Juha

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Re: [SR-Users] Spurious dialog timeouts
by Pranathi Venkatayogi 07 Feb '17

07 Feb '17

Any clue on this? Or to work around this, can someone shed light on how to differentiate when “Dialog:end” is called due to actual call drop - Bye vs some timeout. I need to do some special logic only on call drop. Relevant snippet from the logs are: 7719 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7720 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7721 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7722 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7723 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7724 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent2@translation.sms-test.cyracom.com. Marking free 7725 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7726 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7727 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '1478097bad2847cfaa60a768d76044b0' and tags '57315e78d6514251876bd5a9e32347d2' '00a1eac150f04e2592527372412f059e' ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ -----Original Message----- From: Pranathi Venkatayogi Sent: Friday, February 03, 2017 2:29 PM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Subject: Spurious dialog timeouts Hi, I have two kamailio proxies between the customer and agent. Cust -> proxy1 -> proxy2 -> agent. The dialogs are successfully established and remain undeleted within the customer / agent client. However in Proxy2 I am seeing spurious dialog timeout. I am attaching kamailio log trace below. Also attached is the siptrace on the agent/customer side. Question: How is this dlg_ontimeout() generated despite dialog being successfully established and undelete? How to avoid it. Kamailio (proxy2) log: 7719 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7720 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7721 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7722 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7723 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7724 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent2@translation.sms-test.cyracom.com. Marking free 7725 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7726 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7727 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '1478097bad2847cfaa60a768d76044b0' and tags '57315e78d6514251876bd5a9e32347d2' '00a1eac150f04e2592527372412f059e' 7728 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7729 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7730 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7731 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7732 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7733 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent5@translation.sms-test.cyracom.com. Marking free 7734 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7735 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7736 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '04d924dc160d4e0c9366d4699f579793' and tags '9b107772c82148c0a09182dc86804c51' '44b0a200438a42f8a6ac931ed72f795a' Thanks for your input, Pranathi

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SIP message order
by Andy Chen 06 Feb '17

06 Feb '17

Hello sr-users, First of all, I'm new to this forum so any help would be greatly appreciated as my Kamailio knowledge is somewhat limited. Thanks for your patience. So to the question at hand. If the originator of the SIP method sends SIP messages out of order, does the Kamailio put them back in order before relaying it to the destination device? I'm seeing where the 200 OK was sent first before the SIP UPDATE within the same CSeq number. The reason I'm asking is because I'm seeing this behavior now with UDP transport protocol and I'm trying to justify why we don't need to switch to TCP to fix this issue. Thanks. --Andy

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Kamailio 4.2.3-3.1 Segfaulting - Need debuginfo RPM for kamailio-4.2.3-3.1.x86_64
by Peter DiSanto 06 Feb '17

06 Feb '17

Our Kamailio 4.2.3 recently began seg-faulting in production. I ran a bt full, but without the debuginfo I presume it isn't much use. We cannot locate the original RPMs for this version to be able to obtain the debuginfo RPM. We are running the kamailio-4.2.3-3.1.x86_64 RPM for CentOS6. Anybody have it or can help us find the debug, or offer other advice?

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loose_route ACK and BYE
by Ben Hood 06 Feb '17

06 Feb '17

Hi, I've fixed a loose_route problem with a broken gateway. Essentially gateway sends ACKs and BYEs with an unroutable contact header (i.e. it sticks the address of the proxy into the header, rather than the destination UA address). This is all on the same network, so there is not NAT. So by caching the contact URI to for replies back to the gateway, I can populate $ru to forward the ACK or BYE back to the correct UA. However, what I've noticed is that for BYEs, I also need to set the $du variable in addition to $ru. But this is required for ACKs. This is not a problem per se, but I'm interesting to know why - could anybody explain? Cheers, Ben

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Os on virtual machine
by Diego Nadares 04 Feb '17

04 Feb '17

Hi guys, We are using vmware to run kamailio. The thing is that they give us support only if we install Red Hat. Did you test kamailio on RH? Any issues or things to have in mind? What linux dist do you recommend? Thanks! Diego.

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Fosdem 2017 dinner
by Daniel-Constantin Mierla 04 Feb '17

04 Feb '17

Hello, the dinner at Fosdem 2017 is planned to start at 20:00, on Saturday, Feb 4, 2017. The address is: Le Pickwick Avenue Adolphe Buyl 79 1050 Ixelles, Belgium * https://en.yelp.be/biz/le-pickwick-ixelles The place is nearby Fosdem location. If you announced the intention to participate at the dinner before Feb 1, 2017, your seat is reserved. If you did it afterwards, please contact me during RTC Devroom to confirm the availability of the seats. 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

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Spurious dialog timeouts
by Pranathi Venkatayogi 03 Feb '17

03 Feb '17

Hi, I have two kamailio proxies between the customer and agent. Cust -> proxy1 -> proxy2 -> agent. The dialogs are successfully established and remain undeleted within the customer / agent client. However in Proxy2 I am seeing spurious dialog timeout. I am attaching kamailio log trace below. Also attached is the siptrace on the agent/customer side. Question: How is this dlg_ontimeout() generated despite dialog being successfully established and undelete? How to avoid it. Kamailio (proxy2) log: 7719 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7720 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7721 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7722 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7723 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7724 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent2@translation.sms-test.cyracom.com. Marking free 7725 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7726 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7727 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '1478097bad2847cfaa60a768d76044b0' and tags '57315e78d6514251876bd5a9e32347d2' '00a1eac150f04e2592527372412f059e' 7728 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7729 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7730 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7731 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7732 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7733 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent5@translation.sms-test.cyracom.com. Marking free 7734 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7735 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7736 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '04d924dc160d4e0c9366d4699f579793' and tags '9b107772c82148c0a09182dc86804c51' '44b0a200438a42f8a6ac931ed72f795a' Thanks for your input, Pranathi

1 0

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

1 0

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

4 4

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

2 2

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 ?

2 3

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.

2 3

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

3 2

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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