- sr-users - kamailio.org

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

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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

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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

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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

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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

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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

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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

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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 );

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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

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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 >

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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

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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.

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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

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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

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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

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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.

2 4

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

2 3

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

2 4

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

2 1

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

1 0

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

async_workers vs. async workers modparam in case of async_route()
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, If I am using async_route() with delay, what is the relationship between the core async_workers parameter value and the async module 'workers' modparam? This doc change was intended to clarify it, I think: https://www.mail-archive.com/sr-dev@lists.sip-router.org/msg25002.html But it does not for me: > Number of worker processes to be started to handle the > asynchronous tasks for async_route() and async_sleep(). So, does that mean the module starts its own processes for these cases? If so, does that mean I don't need core async_workers at all? Or do I? If I do, what happens if I have fewer core async_workers than modparam workers? What if I have more? Should I aim to keep them in sync? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

log_prefix ignored in resumed routes
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, I have this log_prefix: log_prefix="[$ci $mt] " Which means xlog("L_INFO", "xyz") normally results in log entries like this: Nov 22 00:29:50 allegro-4.evaristesys.com /usr/local/sbin/kamailio[27625]: [1-27702(a)10.150.20.6 1] INFO: xyz But if xlog("L_INFO", "xyz") happens to occur in a route[] block that is resumed after async_route()—I have not tested other async functions—there is no prefix: Nov 22 00:29:52 allegro-4.evaristesys.com /usr/local/sbin/kamailio[27678]: INFO: xyz -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

flags documentation
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, I have this set up in my configuration boilerplate: -- flags DUMMY : 1, PDD_DAMPEN : 2, PROXY_MEDIA_SET : 3, #!ifdef WITH_NAT_TRAVERSAL SIPPING : 4, #!endif DUMMY2 : 12 -- I then use these aliases in transaction flags, e.g. setflag(PDD_DAMPEN). But I can honestly say I don't remember how I arrived at this approach. I wrote it somewhere in early 3.x days. However, when I look at the core documentation for the 'flags' declaration, all I find is: http://www.kamailio.org/wiki/cookbooks/4.4.x/core#flags Or namely: -- flags Alias name: bool -- And there is no entry for "bool". So, I couldn't have possibly arrived at this usage from the documentation. Can anyone help me reverse-engineer this historical phenomenon? And, given that the documentation doesn't support it particularly, is it still an encouraged practice? Or should one use #!defined constants instead, as in the stock config? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Message flags preservation for replies in onsend_route
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, I have -- flags BLAH : 1, DUMMY : 12 onsend_route_reply=yes route { ... setflag(BLAH); if(!t_relay()) sl_reply_error(); } onsend_route { if(is_reply() && isflagset(BLAH)) do_things(); } -- My discovery has been that isflagset(BLAH) always == FALSE _unless_ an onreply_route has been armed for the request, regardless of whether it is necessary. In other words, the following causes isflagset(BLAH) to == TRUE (if set, when appropriate) in the onsend_route for the reply: -- flags BLAH : 1, DUMMY : 12 onsend_route_reply=yes onreply_route[BOGUS] { exit; } route { ... setflag(BLAH); t_on_reply("BOGUS"); if(!t_relay()) sl_reply_error(); } onsend_route { if(is_reply() && isflagset(BLAH)) do_things(); } -- But without the onreply_route, it doesn't work. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

Registering Trunk help please
by Andrei Mahalean 22 Nov '16

22 Nov '16

Hello, I am having a bit of trouble finding information on how to setup a registering trunk with Kamailio. By registering trunk, I mean a trunk between a PBX and Kamailio, of say 15 DID's, where the main DID (Pilot) registers with Kamailio and all the other 14 DID are able to make/receive calls without having to register all DID's individually. If the pilot DID registration goes away, the trunk is then considered to be unreachable/down. I have seen this in the past with a Broadsoft setup, but I can't figure out a way to do it with Kamailio. I have my pilot DID setup and registered with Kamailio ok and I can make calls in and out through my PSTN gateway, but I'm unsure how to group all the other DID's so they are matched against the pilot DID. Is this possible? Any help is appreciated. Regards, Andrei. -- Regards, Andrei.

2 1

Roadmap to Kamailio v5.0
by Daniel-Constantin Mierla 21 Nov '16

21 Nov '16

Hello, following the last IRC devel meeting, I am writing the summary of the main points being discussed there and what's planned to get to Kamailio v5.0. 1) Freezing should be done either before Chrismas holidays or at beginning of January 2017 to aim meeting the deadline for next Debian stable distro 2) Restructuring the source tree To be done in the weeks before freezing, so eventual fixes won't have hard time to be backported to 4.4. This means it will be done in December. Open issues: a) anyone aware of a tool that can update include directives upon relocation of the file? b) who is interested to join an eventual online session to perform the restructuring? C coding is not a must here, but more knowledge of sed/awk/perl/python/... and Makefiles ... 3) Defining a template for generating main file of a module The scope is to have a descriptive file for the module where one lists exported symbols (config params, functions, kemi functions, rpc/mi commands, etc.) which along with a template will generate the c code. The effective implementation of the functions/commands will be done separately, in a different file (e.g, the main file that is generated can have references to parameter variables with extern ... and will also generate the .h which will have only the prototypes for the functions). The main goal here is to get a way of easily generated an index of exported symbols that can be used to discover missing documentation. Right now we sort of do 'gentlemen agreement' on documenting functions and parameters, but many rpc commands are not documented (they are extracted with some tool scanning compiled objects). Also, same concept as with c code can be used to generate some stubs for documentation that can be included in docbook xml files. Few proposals of templating engines were mentioned, coming in my mind now being jinja2 and mustache. Anyone else with more suggestions? Or even propose alternatives of what can be used to solve this relation between code and docs? Cheers, Daniel -- 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

usage of app_mono : how to communicate results back to config file
by Pranathi Venkatayogi 21 Nov '16

21 Nov '16

Hi, I am using app_mono module and building my logic in c# app. I am unable to understand how to send the decision back to kamalio.cfg. More precisely: I would like my routing logic to allow specific user names though they are not registered as subscribers. Like: translate(a)myip.cyracom.com<mailto:translate@myip.cyracom.com> I have the logic in c# app which takes $ru and validates such that: Host is myip.cyracom.com User is translate How do I communicate the decision if it is valid or not back to my config script. Do I need to introduce new pseudo variable for this purpose? Thanks, Pranathi

2 1

Is there a way to detect inactive MSRP chat sessions without using SIP session timers
by Pranathi Venkatayogi 18 Nov '16

18 Nov '16

Hi, I am using MSRP feature of Blink client. I want to track when the chat starts and ends so I can free up server resources. Apparently Blink client does not send BYE when MSRP chat window closes. It does not support SIP session timers even. So potential use of SIP session timers (sst and dialog modules) is ruled out. Only possible way is to track every message that goes through the server and keep the session alive. If no message comes within “expiry time period”, server can send “bye” to all parties and tear up the session. Can you please comment if there are other approaches to solve this problem? If not, are they any modules that will help me implement the above approach easily? Thanks Pranathi

2 2

tcp_set_connection_lifetime - does not close the connection immediately
by Andy 18 Nov '16

18 Nov '16

I am calling tcp_set_connection_lifetime(conid, 1), which I understand is one sec after now, but I see it taking upto 60 secs to kill the connection, is there any other timer which has a larger interval which might affect this. -- View this message in context: http://sip-router.1086192.n5.nabble.com/tcp-set-connection-lifetime-does-no… Sent from the Users mailing list archive at Nabble.com.

2 2

Htable sht_rm_name_re usage
by Grant Bagdasarian 18 Nov '16

18 Nov '16

Hello Community, We were facing a weird issue in one of our Kamailio servers where custom htable entries were removed which did not belong to a particular call. Background Information: We use htable to store the call-id and a uuid when an INVITE message is received and remove these entries when the call is completed/failed (so on CANCEL, BYE, 4XX, 5XX, 6XX, etc). In this case the function sht_rm_name_re was causing incremented call-id's belonging to different calls to be removed from the htable, since the removal is based on regex and not the exact key. On INVITE we do this: $var(reference) = $uuid(g); $sht(calls=>$ci) = $ci; $sht(calls=>$ci::reference) = $var(reference); On BYE/4XX,5XX,6XX, etc we did this: sht_rm_name_re("calls=>$ci"); sht_rm_name_re("calls=>$ci::reference"); SIPP generates incremented call-id's, like this: 1-14875(a)1.0.0.1<mailto:1-14875@1.0.0.1>. When you generate enough calls, for instance 30 calls, the 21st call will have the call-id 21-14875(a)1.0.0.1<mailto:21-14875@1.0.0.1>. If for instance the BYE for the call-id 1-14875(a)1.0.0.1<mailto:1-14875@1.0.0.1> arrives before 21-14875(a)1.0.0.1<mailto:21-14875@1.0.0.1> , both htable entries will be removed by the sht_rm_name_re function, since it matches based on regex. To fix this we made sure the function matches on the exact keys by doing this: $var(callid) = '^' + $ci + '$'; $var(callid_reference) = '^' + $ci + '::reference' + '$'; sht_rm_name_re("calls=>$var(callid)"); sht_rm_name_re("calls=>$var(callid_reference)"); Hope this may help someone in the future facing the same issue. Regards, Grant Bagdasarian CM

2 1

Kamailio for SIP Capture With Registration
by Ali Dakroub~Vanrise Technical Support 18 Nov '16

18 Nov '16

Hello, Kindly note that I am configuring Kamailio for SIP capture with "Registration" module (registrar.so). Before loading the "Sip Capture" module (sipcapture.so) into Kamailio.cfg file, the registration of the SIP Phone (X-lite) is successful. Yet, when we add the "Sip Capture" module (sipcapture.so) into Kamailio.cfg file with the below parameters: modparam("sipcapture", "db_url", "postgres://postgres:postgres@localhost/kamailio") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_moni_capture_on", 1) /* IP to listen. Port/Portrange apply only on mirroring port capturing */ modparam("sipcapture", "raw_socket_listen", "10.0.10.1:5060") /* Name of interface to bind on raw socket */ modparam("sipcapture", "raw_interface", "eth0") /* children for raw socket */ modparam("sipcapture", "raw_sock_children", 4) /* insert delayed */ #modparam("sipcapture", "db_insert_mode", 1) /* Linux only */ /* Promiscious mode RAW socket. Mirroring port. */ modparam("sipcapture", "promiscious_on", 1) /* activate BPF */ modparam("sipcapture", "raw_moni_bpf_on", 1) With the following routing logic: ####### Routing Logic ######## request_route { #sip_capture(); # CANCEL processing # 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(); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service } route[REGISTRAR] { if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } } The registration of the SIP Phone fails and we are getting the following error in Kamailio.log file: 2016-11-16T09:49:28.809247-08:00 ubuntu /usr/local/sbin/kamailio[10612]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f60c7593910,481,0,192.168.26.4:14661,16): Bad file descriptor(9) 2016-11-16T09:49:28.856080-08:00 ubuntu /usr/local/sbin/kamailio[10612]: ERROR: sl [../../forward.h:192]: msg_send(): udp_send failed 2016-11-16T09:49:28.860251-08:00 ubuntu /usr/local/sbin/kamailio[10612]: ERROR: registrar [reply.c:734]: reg_send_reply(): failed to send 200 OK It would be highly appreciated to check what went wrong here. Thanks in advance. Ali Dakroub

2 1

Kamctl stats
by Alexandru Covalschi 18 Nov '16

18 Nov '16

Hello list, Where can I found any information to completely understand what do values returned by 'kamctl stats' represent? Cheers, Alex

2 1

Re: [SR-Users] msilo
by Slava Bendersky 17 Nov '16

17 Nov '16

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: 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

2 1

Possible memory leak in mysql driver
by Alexandru Covalschi 17 Nov '16

17 Nov '16

Hello list, We’re using dev version of Kamailio: version: kamailio 5.0.0-dev4 (x86_64/linux) ff63e5 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: ff63e5 compiled on 15:46:49 May 31 2016 with gcc 4.9.2 Sometimes we encounter such issue: Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: <core> [db_row.c:114]: db_allocate_row(): no private memory left Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_row.c:57]: db_mysql_convert_row(): could not allocate row Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_res.c:188]: db_mysql_convert_rows(): error while converting row #16 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_res.c:217]: db_mysql_convert_result(): error while converting rows Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_dbase.c:261]: db_mysql_store_result(): error while converting result Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: <core> [db_query.c:139]: db_do_query_internal(): error while storing result Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: permissions [trusted.c:91]: reload_trusted_table(): failed to query database Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: <core> [db_row.c:114]: db_allocate_row(): no private memory left Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_row.c:57]: db_mysql_convert_row(): could not allocate row Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_dbase.c:444]: db_mysql_fetch_result(): error while converting row #15 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: htable [ht_db.c:234]: ht_db_load_table(): Error while fetching result Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f5ebda8ae18), called from db_mysql: km_dbase.c: db_mysql_free_result(305), first free db_mysql: km_dbase.c: db_mysql_free_result(305) - aborting Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12281]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 16 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT: <core> [main.c:739]: handle_sigs(): child process 12276 exited by a signal 6 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT: <core> [main.c:742]: handle_sigs(): core was not generated Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD The thing is we heavily use mysql module, but only to update the in-memory tables by kamcmd. Each N minutes a special script updates the trusted,address and htable executing kamcmd. Kamailio (and kamcmd as well) talks only with localhost mysql server. What I saw when encountered that issue on a live machine is that issue happens only with one of child processes, any other are ok. Interesting thing is that happens at the same time with machines on the same «set», I mean that issue happened simultaneously with two our test machines which actually didn’t have any load on them. The common thing between those machines is that they are in same subnet and local mysql databases are filled by scripts which query same external db. I can’t confirm if there were or there weren’t any networking issues at that time with those machines, but as soon as kamcmd queries localhost that shouldn’t be the source of the issue. So my questions are: 1. Has anyone encountered such thing? 2. Maybe the issue is already localized so it has sense to update? We actually use that on production (pls don’t throw too much rocks at me), so maintenance should be properly planned and I must be sure update won’t break anything. 3. If update is proposed - how to do it? I mean - follow the guide https://www.kamailio.org/wiki/install/devel/git or there are some other tips? I suppose in ideal world I don’t even stop the binary, only restart after make all && make install are done, as everything is in-memory. Am I correct? 4. When can we expect stable 5.0 version? (at least tell if it’s months/years) Thanks in advance!

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Error message not seen before
by Kevin Greene 17 Nov '16

17 Nov '16

Hi fellow Kamailio people: I was using the Siremis (v2.0) to manage my Kamailio application. I received a message labeled "System Internal Error" It seems, assuming I understand it correctly, that a table called "kamailio.uacreg" does not exist in the database. I receive this error when I click on the following: Main System > SIP Admin Menu > Server Services > UACReg List function. Any help in pointing me in the right direction would be great. Thanks. Kevin 2016-11-16 22:29:23 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.l_uuid, T0.l_username, T0.l_domain, T0.r_username, T0.r_domain, T0.realm, T0.auth_username, T0.auth_password, T0.auth_proxy, T0.expires FROM uacreg T0 LIMIT 10. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'kamailio.uacreg' doesn't exist Script name and line number of error: /var/www/siremis/openbiz/bin/data/BizDataObj_Lite.php:447 *function:* _run_search ( Array(2) ) @ /var/www/siremis/openbiz/bin/data/BizDataObj_Lite.php 243 *function:* fetch ( ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 690 *function:* fetchDataSet ( ) @ /var/www/siremis/openbiz/bin/easy/FormRenderer.php 69 *function:* renderSmarty ( Object(EasyForm), "/var/www/siremis/siremis/modules/ser/template/grid.tpl" ) @ /var/www/siremis/openbiz/bin/easy/FormRenderer.php 45 *function:* render ( Object(EasyForm) ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 1648 *function:* renderHTML ( ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 1538 *function:* render ( ) @ /var/www/siremis/openbiz/bin/easy/ViewRenderer.php 83 *function:* renderSmarty ( Object(EasyView), "/var/www/siremis/siremis/modules/ser/template/view.tpl" ) @ /var/www/siremis/openbiz/bin/easy/ViewRenderer.php 40 *function:* render ( Object(EasyView) ) @ /var/www/siremis/openbiz/bin/easy/EasyView.php 274 *function:* _render ( ) @ /var/www/siremis/openbiz/bin/easy/EasyView.php 237 *function:* render ( ) @ /var/www/siremis/openbiz/bin/BizController.php 221 *function:* renderView ( "ser.view.UacregListView", "", "", Null, "" ) @ /var/www/siremis/openbiz/bin/BizController.php 107 *function:* dispatchRequest ( ) @ /var/www/siremis/openbiz/bin/BizController.php 32 *function:* include_once ( "/var/www/siremis/openbiz/bin/BizController.php" ) @

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[CFP] reminder! FOSDEM RTC dev-room talks: deadline tomorrow
by FOSDEM RTC Team 16 Nov '16

16 Nov '16

Reminder: speaker's deadline tomorrow, 17 November at 23:59 UTC The Free RTC dev-room has already received some really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak. Many other dev-rooms also have a deadline in the next few days and if your topic is applicable to more than one dev-room, you are welcome to make more than one submission. Please contact us or put a note in the memo field at the top of the talk proposal if you do that. All projects are encouraged to consider making a lightning talk too, it is an excellent opportunity to get exposure for your project: even though you only have 15 minutes, it can be a much larger and more diverse audience than in some dev-rooms. For full details, please see the original call for participation: https://danielpocock.com/fosdem-2017-rtc-cfp We invite all potential speakers and participants to discuss the selection process and other aspects of FOSDEM on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc

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Re: [SR-Users] msilo
by Federico Cabiddu 16 Nov '16

16 Nov '16

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: 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: 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlahttp://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

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

16 Nov '16

Hello Everyone, I am trying just forward MESSAGE through kamailio and I reported issue with mod_sms in freeswitch https://freeswitch.org/jira/browse/FS-9701. So I was need add work around on kamailio side. The issue that my setup MSILO should work in intercept mode meaning check when user online of offline and act as m_store() or m_dump() without rely on lookup("location");, because registration on B2BUA side. If m_store() will have additional parameter to store body this will allow check user state and send when need it. On reply I put something like this to fix To, as work around and forward MESSAGE to client, but when client send MESSAGE kamailio receive on public interface and push out of public interface instead RELAY through local interface and then to B2BUA. 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")) { $avp(from) = $fu; $avp(to) = $tu; $avp(new_to) = "<" + $avp(to) + ">" + ";messagetype=SMS"; xlog("L_INFO", "Message Accepted by B2BUA --> [$rs] with [$rr] from [$avp(from)] to [$avp(to)]\n"); if(fnmatch("$fU", "offline")) { xlog("L_INFO", "OFFLINE SMS: from [$fU] user --> $fU\n"); $avp(from) = $(avp(to){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"; } $sht(a=>to) = $avp(new_to); xlog("SMS from --> $fU domain $fd\n"); xlog("-------------------------------------\n"); xlog("L_INFO", "FROM --> $avp(from)\n"); xlog("L_INFO", "OLD_TO --> $avp(to)\n"); xlog("L_INFO", "NEW_TO --> $avp(new_to)\n"); } if(compare_ips("$td", "10.18.130.27")) { xlog("L_INFO", "Message from B2BUA contain domain --> $td . Updating...\n"); $avp(new_dst) = $(hdr(Route){param.value,received}); xlog("L_INFO", "New destination --> $avp(new_dst)\n"); $avp(new_to) = "<" + $avp(new_dst) + ">" + ";messagetype=SMS"; $sht(a=>to) = $avp(new_to); remove_hf("To"); insert_hf("To: $sht(a=>to)\r\n", "To"); $du = $sht(a=>to); } } From: "Federico Cabiddu" <federico.cabiddu(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, 15 November, 2016 11:07:44 Subject: Re: [SR-Users] msilo 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: 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 _______________________________________________ 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 15 Nov '16

15 Nov '16

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: 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

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How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 15 Nov '16

15 Nov '16

Hello I have tried $Rp before sending this to mailing list, but with $Rp I have different values for example 188, 6847, 3163, 9175 for every call ?! Don't know where are they come from, because in URI I see :6060, and using $rp give my 6060 for examples with $Rp. Only have problem with UAC Panasonic-MPR12-V004.41009/VSIPGW-V3.0000 with doesn't send $rp in URI. Greetings

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RTPProxy benchmark
by Gholamreza Sabery 15 Nov '16

15 Nov '16

I managed to create about 1900 concurrent calls using a single Kamailio and RTPProxy server. But after this number RTPProxy returns 0 and the following error is shown in the Kamailio log files: incorrect port 0 in reply from rtp proxy What is the problem here? Also number of file descriptors that RTPProxy can use are set to a million.

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How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 15 Nov '16

15 Nov '16

Hello My point is to log INVITEs in my Network. My SIP Network work on port 6060. Im am using HEP to do that, and hep clients that are listening on port range 5060-6066 to detect some SIP attack to port 5060 and others. But when I have attack to port 5060 I don't want to insert that INVITE to my "good traffic" table, but place it to "fraud" table. Some devices like Panasonic PBX send INVITE to port 6060 so it is "good traffic" but there is no port 6060 in URI, so I can't detect it in right way, because $dp or $rp are 5060 by default. This kamailio work as capture server as "promiscuous_on", so I can't use any "force_rport" etc. because I am only listening. Tcpdump for that SIP session show me that client send traffic to 6060, but I can't get that information from INVITE header. Greetings

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[CFP] FOSDEM 2017, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 15 Nov '16

15 Nov '16

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2017 takes place 4-5 February 2017 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Saturday, 4 February 2017. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 17 November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real-Time devroom". https://penta.fosdem.org/submission/FOSDEM17/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2017/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 31 October. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 4 February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists See the mailing list discussion for more details about volunteering: https://lists.fsfe.org/pipermail/free-rtc/2016-October/000285.html Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on 2 and 3 February 2017 http://wiki.xmpp.org/web/Summit_21 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit We are also considering a more general RTC or telephony summit, potentially in collaboration with the XMPP summit. Please join the Free-RTC mailing list and send an email if you would be interested in participating, sponsoring or hosting such an event. Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 3 February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

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INVITE timeout timer
by Diogenes Aviles 15 Nov '16

15 Nov '16

I have changed the following tm parameters in order to have an timeout INVITE equal to 20 sec (20000 ms). # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 0) # default retransmission timeout: 20sec modparam("tm", "fr_timer", 20000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) However, the timeout is ever 3 min (180 sec.) In accord to tm module the timeout is defined only by "r_timer" Then, I don't know where is the problem. Thanks a lot for your help. -- Diogenes

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msilo
by Slava Bendersky 15 Nov '16

15 Nov '16

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.

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 14 Nov '16

14 Nov '16

Hello Daniel, I resolved all configuration issues and right now call goes both directions include rtp. Again huge thank you for help. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 12:24:57 Subject: Re: [SR-Users] BYE dispatcher Hello, this logic is definitely wrong -- FreeSwitch can send also a request, it means that you send it back to it. Only the initial request of a dialog should be routed with rules like dispatcher/load balancer/least cost routing/... The requests within dialog should be routed based on loose routing. Of course, one can think of exceptions, but then you should be fully aware of what kind of routing you do. Cheers, Daniel On 10/11/16 16:25, Slava Bendersky wrote: Hello Daniel, My setup is proxy all requests to freeswitch via dispatcher. 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, 10 November, 2016 04:56:53 Subject: Re: [SR-Users] BYE dispatcher Hello, as I said before, the registrations have little to do with calls in sip, unless there is gruu in use. Cheers, Daniel On 09/11/16 18:07, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj To: < sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. 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 1

Homer's kamailio config + enabling siptrace module
by Ján Füri 11 Nov '16

11 Nov '16

Hi guys, I'm using captagent on my all kamalios and freeswitches to capture and send sip to homer server. That works very well. But I would like to enable, on homer's kamailio, the module SipTrace (http://www.kamailio.org/docs/modules/4.4.x/modules/siptrace.html) and save captured data into mysql as well. I already set it up using this mysql structure : http://www.kamailio.org/docs/db-tables/kamailio-db-devel.html#idp1279328 Almost everything works except the column "DIRECTION". Each row is saved as "IN" direction. Do you guys think that can be solved ? If so, can you pleas direct me HOW ... Thx Jan F.

2 3

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 11 Nov '16

11 Nov '16

Hello I have some UAC like Panasonic PBX, that send traffic to port 6060 that I am listening on, but that port isn't included to R-URI. I can see that only at tcpdump, or sip_trace from sip_capture module. Kamailio variables like $dp or $rp have default value of 5060. Thank You,

3 2

Execution Error while getting profile size with jsonrpc-s
by Aqs Younas 11 Nov '16

11 Nov '16

Greetings list, I am trying to get profile size with jsonrpc-s module. Below is jsonrpc-s configuration and a curl command to get the profile size. listen=tcp:0.0.0.0:5060 loadmodule "xhttp" loadmodule "jsonrpc-s" modparam("xhttp", "url_match", "^/rpc_path/") modparam("jsonrpc-s", "pretty_format", 1) event_route[xhttp:request] { if($hu =~ "^/rpc_path") jsonrpc_dispatch(); else xhttp_reply("404", "Not Found", "text/html", ""); } curl -v -H 'Content-Type: application/json' -H 'Call-ID: abc12' -d '{"jsonrpc": "2.0", "method": "dlg.profile_get_size","params":{"accountno": "1234567"}, "id": "0deadb33f"}' http://127.0.0.1:5060/rpc_path/ I am getting this in respose. * Hostname was NOT found in DNS cache * Trying 127.0.0.1... * Connected to 127.0.0.1 (127.0.0.1) port 5060 (#0) > POST /rpc_path/ HTTP/1.1 > User-Agent: curl/7.38.0 > Host: 127.0.0.1:5060 > Accept: */* > Content-Type: application/json > Call-ID: abc12 > Content-Length: 106 > * upload completely sent off: 106 out of 106 bytes *< HTTP/1.1 404 Profile not found: 1234567* < Sia: SIP/2.0/TCP 127.0.0.1:49374 < Call-ID: abc12 < Content-Type: application/json * Server kamailio (4.4.1 (x86_64/linux)) is not blacklisted < Server: kamailio (4.4.1 (x86_64/linux)) < Content-Length: 106 < { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": "0deadb33f" * Connection #0 to host 127.0.0.1 left intact Whereas fifo command gives me this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo profile_get_size accountno 1234567 profile:: name=accountno value=1234567 count=2 Why jsonrpc-s is search for profile 1234567 whereas profile should be accountno. Anyhelp what i am doing wrong here is much appreciated. Best Regards.

2 4

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 11 Nov '16

11 Nov '16

Hello Daniel, I really ask for help, here are configuration file https://paste.fedoraproject.org/477652/88413891/ I spent quite a lot of time trying understand loose_route() /record_route() mix. I can get signalling working, call is not disconnects, but no RTP. Or I can get rtp and signalling BYE is not routed properly. My setup is just proxy all requests to freesiwtch boxes base on dispatcher selection where kamailio setup with 2 interfaces public and private. I really appreciate on you time and help. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Cc: miconda(a)gmail.com Sent: Thursday, 10 November, 2016 23:54:40 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, What From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 11:25:19 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, My setup is proxy all requests to freeswitch via dispatcher. 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, 10 November, 2016 04:56:53 Subject: Re: [SR-Users] BYE dispatcher Hello, as I said before, the registrations have little to do with calls in sip, unless there is gruu in use. Cheers, Daniel On 09/11/16 18:07, Slava Bendersky wrote: Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj To: < sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- 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 -- 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

Kamailio 4.4 VoiceMail
by Gonzalo Gasca Meza 10 Nov '16

10 Nov '16

I have the following call flow: INVITE -> sip:gonzalo@sip.parzee.io -- TLS/TCP/UDP -> *KAMAILIO* - DB Lookup -> INVITE sip:gonzalo58@test.external.com;transport=tls (Phone1) If Phone1 is Busy or No answer, I want call to go to VM. Phone1, is not registered to Kamailio, nor I'm using Realtime Integration. This Phone1 is registered to an external PBX. Currently in sample configuration script, seems to be that value: $avp(oexten) is used to redirect to VM, but in my case this value is null. I didnt find any documentation for this. *Questions:* a) What is $avp(oexten) ? b) What is the best way to pass a Redirect number in SIP INVITE to VoiceMail system (Asterisk or Freeswitch) c) Is there a way to configure CFNA timer per alias/uri ? *Example*: route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { xdbg("SIP Request: method [$rm] from [$fu] to [$tu]\n"); xlog("VoiceMail routing enabled $avp(oexten)\n"); if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("SCRIPT: VoiceMail to $tu\n"); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; }

2 2

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, My setup is proxy all requests to freeswitch via dispatcher. 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, 10 November, 2016 04:56:53 Subject: Re: [SR-Users] BYE dispatcher Hello, as I said before, the registrations have little to do with calls in sip, unless there is gruu in use. Cheers, Daniel On 09/11/16 18:07, Slava Bendersky wrote: Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj To: < sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- 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 -- 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 1

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 10 Nov '16

10 Nov '16

Hello Thank You. In using Kamailio only as: route { if (is_method("INVITE")) { sql_query("cb", "insert into invites (from_user,to_user,callid,src_ip,dst_ip,src_port,dst_port) values ('$fU','$rU','$ci','$si','$rd','$sp','$rp')"); } } I know that sip_capture(); In some way get real DST port even if there is no port in r-URI. Greetings

2 1

Change CLID
by Nelson Migliaro 10 Nov '16

10 Nov '16

Hello, I have several SIP vendors and I must present a different CLID for each of them. I am planning to use failure routers in Kamailio to route calls to a different vendor in case of failure. The problem I face is changing clid in each route. I am planning to use a mysql database to select the clid based on a prefix and sip vendor and in case of failure, select new clid base on new sip vendor. Is mysql the right way to do that, is there a module for this? In order to change the clid I am thinking of using: uac_replace_to(display,uri). Is this correct way to do that? And finally, in case of using uac_replace_to(display,uri), do I have to use uac_restore_from() to restone the correct from to avoid further issues? Thank you in advance Nelson.-

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- 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

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- 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

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- 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] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- 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

usrloc module expires column issues
by Grant Bagdasarian 10 Nov '16

10 Nov '16

Hello, When saving in the location table, the expires and last_modified columns contain the value 000-00-00 00:00:00 which causes issues when Kamailio is restarted and the preload modparam is set to load the location table from db in memory at startup. I've already upgraded Kamailio to 4.4.3, and used the dbtables script from version 4.4 to recreate the location table, but without success. Anyone else having this issue? Any resolution for it? Regards, Grant Bagdasarian CM

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, Here more clear trace https://www.dropbox.com/s/2z6ck71ulidqelh/kamailio-BYE-flow.png?dl=0 BYE send by kamailio to wrong freeswitch box. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 20:09:53 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- 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] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- 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

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Kamailio v4.4.4 Released
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, Kamailio SIP Server v4.4.4 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.3. 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 v4.4.4. For more details about version 4.4.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/11/kamailio-v4-4-4-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

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Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. for ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info againts a gateway.? Or any way to insert early dialog into database with callee_contact not empty? Best Regards, Aqs Younas

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Planning the release of Kamailio v4.4.4
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, I am considering to package a new release out of latest stable branch 4.4, respectively v4.4.4. The target date is one of the days between Nov 8-10, a matter of work load that needs to be done for getting it out. As usual, if you are aware of any issue not yet reported on github bug tracker, do it as soon as possible to give it a chance to be fixed: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- 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

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Re: [SR-Users] fnmatch
by Slava Bendersky 09 Nov '16

09 Nov '16

Hello Daniel, Yes, reply was 202 not 200. Right now it work as expected. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, 7 November, 2016 04:51:11 Subject: Re: [SR-Users] fnmatch Hello, for clarification -- what's the actual problem here? fnmatch("$rs", "200") is matching the 202? Cheers, Daniel On 07/11/16 08:52, Serge S. Yuriev wrote: Hello, In debug you can see 202 Accepted but match configured on 200. Not matched -- Wbr, Serge via mobile 07.11.2016, 01:08, "Slava Bendersky" <volga629(a)skillsearch.ca> : BQ_BEGIN Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? 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 _______________________________________________ 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 _______________________________________________ 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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BYE dispatcher
by Slava Bendersky 09 Nov '16

09 Nov '16

Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). client_public_ip.28972 > proxy_public_ip.llrp: [udp sum ok] UDP, length 660 E....2..5...c....E.\q,.....CBYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Via: SIP/2.0/UDP client_public_ip:28972;branch=z9hG4bK1248337638;rport Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 70 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x10, ttl 64, id 10813, offset 0, flags [none], proto UDP (17), length 930) 10.18.130.27.sip > 10.18.130.24.5160: [bad udp cksum 0x1bf7 -> 0xd85c!] UDP, length 902 E...*=..@.4. ... ......(....BYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Record-Route: <sip:10.18.130.27;r2=on;lr=on;ftag=487304085> Record-Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=487304085> Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 69 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x0, ttl 64, id 38202, offset 0, flags [none], proto UDP (17), length 587) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 559 E..K.:..@... ... ....(...7i.SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x10, ttl 64, id 782, offset 0, flags [none], proto UDP (17), length 503) proxy_public_ip.llrp > client_public_ip.28972: [bad udp cksum 0x4c50 -> 0xd018!] UDP, length 475 E.......@.+}.E.\c.....q,..LPSIP/2.0 200 OK Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x0, ttl 64, id 10584, offset 0, flags [none], proto UDP (17), length 764) 10.18.130.26.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 736 E...)X..@.6@ ... ....(....Y^BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974;alias=client_public_ip~48147~1 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.26:5160;rport;branch=z9hG4bK33785247SrX3H Route: <sip:10.18.130.27;r2=on;lr=on;ftag=d2e84f4a> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=d2e84f4a> Max-Forwards: 70 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 802, offset 0, flags [none], proto UDP (17), length 741) proxy_public_ip.llrp > client_public_ip.48147: [bad udp cksum 0x4d3e -> 0x4f2d!] UDP, length 713 E...."..@.*{.E.\c.........M>BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H Max-Forwards: 69 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 810, offset 0, flags [DF], proto UDP (17), length 504) client_public_ip.48147 > proxy_public_ip.llrp: [udp sum ok] UDP, length 476 E....*@.4..oc....E.\........SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 IP (tos 0x10, ttl 64, id 63040, offset 0, flags [none], proto UDP (17), length 413) 10.18.130.27.sip > 10.18.130.26.5160: [bad udp cksum 0x19f4 -> 0x6386!] UDP, length 385 E....@..@.j. ... ......(....SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 Slava.

2 1

Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. For ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info against each gateway or every dialog stat insertion in database dialog table. Best Regards.

1 0

using set_body_multipart and append_body_part on on_reply routes
by david 08 Nov '16

08 Nov '16

hello all i'm using kam version 4.4.1 and i would like to generate an ISUP part on the 1xx,2xx responses to a remote carrier since the fsw i send the calls to cannot handle ISUP SDP. despite seeing in the doc that those functions cannot be used there and only in REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE, i tried and it works.... partially, because it seems it doesnt create the last boundary and it creates a next boundary with an empty "Encapsulated multipart part: " this the the route i call from onreply_route route[SIPISUP] { set_body_multipart(); msg_apply_changes(); $var(isup) = "\x01\x12\x49\x01\x0a\x03\x02\x0a\x08\x84\x90\x33\x41 \x72\x17\x01\x06\x0a\x08\x04\x13\x93\x70\x21\x73\x23\x10\x08\x01\x01 \x00"; append_body_part("$var(isup)","application/isup;version=itu-t92 +","signal;handling=optional"); msg_apply_changes(); sipt_destination($rU, 31, 4); sipt_set_calling($fU, 4, 0, 3); msg_apply_changes(); } am i doing something wrong? how can i set this to end the SDP with last boundary? thanks a lot and regards david

2 3

Re: [SR-Users] kamailio + farm of Elastix + rtp direct
by Kato 08 Nov '16

08 Nov '16

Any comments on this please? Thanks! De: Kato [mailto:elkato@gmail.com] Enviado el: domingo, 6 de noviembre de 2016 11:29 a. m. Para: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Asunto: kamailio + farm of Elastix + rtp direct Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Next Kamailio IRC Devel Meeting, Nov 09, 2016
by Daniel-Constantin Mierla 08 Nov '16

08 Nov '16

Hello, I am proposing a new IRC meeting to discuss the current major issues and the plans for next Kamailio release, on Wednesday, Nov 09, 2016, with alternatives for previous or next day. If many developers are not available, we can postpone it to another date in the near future (make proposals if that is the case for you). I created a wiki page for it: * https://www.kamailio.org/wiki/devel/irc-meetings/2016b Add there the topics that you want to be discussed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 4

Contacts replacing each other when registering
by Sebastian Damm 08 Nov '16

08 Nov '16

Hi, we are running a Registrar server with 4.4.3, the registrar module is configured with the following parameters: modparam("registrar", "max_contacts", 50) modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("registrar", "received_avp", "$(avp(i:801))") modparam("registrar", "default_expires", 600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 600) modparam("registrar", "gruu_enabled", 0) A customer has configured 4 phones with the same username. Normally, all 4 phones would be visible in the location database, and incoming calls would fork out to all 4 phones. However, only one of the devices is always online, the others always replace each other on each registration. Client details: * All phones are snom 715 devices with the same firmware version. * All phones sit behind the same router. Registrations: client 1: * Via: SIP/2.0/UDP 192.168.208.91:35812;branch=z9hG4bK-4q0mmxgu6gm7;rport * Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 2: * Via: SIP/2.0/UDP 192.168.208.90:41331;branch=z9hG4bK-3cbxdvbmk0di;rport * Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 3: * Via: SIP/2.0/UDP 192.168.208.92:5360;branch=z9hG4bK-4drsogakez94;rport * Contact: <sip:1794388e3@192.168.208.92:5360;line=0n7jfzih>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 4: * Via: SIP/2.0/UDP 192.168.208.93:5260;branch=z9hG4bK-1ci0qia3yxyl;rport * Contact: <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;reg-id=1;q=1.0 Only client 4 is reachable all the time. The other 3 clients replace other registrations. So after client 1 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;q=1;expires=600;received="sip:1.2.3.4:6411";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=303;received="sip:1.2.3.4:8923" And after client 2 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;q=1;expires=600;received="sip:1.2.3.4:49061";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=301;received="sip:1.2.3.4:8923" So you see, one contact stays the same, the other one gets replaced. Does anybody know what could cause this kind of behavior? I think, it could be GRUU. I see that those three clients all send the same GRUU parameter. From my understanding, this UUID should be different from client to client. I don't know whether this is a bug in the snom firmware always sending the same GRUU UUID or whether som "intelligent" SIP aware router is putting this in the Contact header. However, since we have disabled gruu for the registrar module (see above), I think, Kamailio should ignore it. Ideas? Thanks in advance. Best Regards, Sebastian

2 4

TCP Keepalive for NAT'd Clients
by Colin Morelli 08 Nov '16

08 Nov '16

Hey all, Looking to figure out the best way to allow TCP connections to stay alive for NAT'd clients, however, to protect against people just opening TCP connections to the server, I'm hoping to only keepalive TCP connections for connections that have sent an INVITE and received a 200 OK. Does this type of per-socket connection lifetime setting exist? For a bit more background: this is specifically for clients that are using the TCP connection for an INVITE. They don't register with the server, and as a result there's no active flow timer for their connection. Is there a way to enable this type of behavior without registering? Best, Colin

2 2

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 08 Nov '16

08 Nov '16

Hello I am using Kamailio to collect data about active calls. I write INVITEs to database, and delete it on BYE or any REL. I would like to write source and destination port too, but don't know how to get destination port. I have tried variables: $dp, $rp, $Rp, $op But when I don't have port in URI like INVITE IP:PORT, $rp have default value of 5060. I would like to get real UDP port, that INVITE does to. Is it possible using PSEUDO VARIABLES? Greetings Andrzej

2 1

Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB
by Jurijs Ivolga 08 Nov '16

08 Nov '16

Hi, Our Kamailio server is crashing once per week, with following error: Oct 1 06:25:06 kamailio kernel: [26982632.803789] Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB Core dump was never created, probably it is because of my environment, but I will try to get it. Server constantly eats memory, maybe some kind of memory leak? Any help is highly appreciated! Jurijs

3 25

NER and ASR
by Gholamreza Sabery 07 Nov '16

07 Nov '16

Are there any real data about number of concurrent calls, number of on-line users, NER (Network Effectiveness Ratio) and ASR (Answer Seizure Ratio) of a real VoIP system? I want this data for making comparisons. Can anyone provide a reference, article, etc..

3 4

cdr on remote POSTGRES server
by beer Ll 07 Nov '16

07 Nov '16

Hi all I have a kamailio server configured with accounting on a separate Postgres SQL server by an internal network with acc module and db_postgres module Everything works good but what can I do if I have problem on the internal network or on the postgres server ? Is it possible cache or save the data if the insert data is failed ?

2 1

fnmatch
by Slava Bendersky 07 Nov '16

07 Nov '16

Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava.

3 2

kamailio + farm of Elastix + rtp direct
by Kato 06 Nov '16

06 Nov '16

Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Kamailio DB lookup
by Gonzalo Gasca Meza 06 Nov '16

06 Nov '16

Hi Kamailions, Kamailio 4.3 will receive calls (SIP TLS) for users in this domain: *sip.parzee.io <http://sip.parzee.io>. *Example sip:gonzalo@sip.parzee.io When call is received by Kamailio I need to overwrite R-URI and transport. I should keep a mapping per user. ( gonzalo -> gonzalo58) Example. sip:gonzalo58@test.external.com;transport=tls I started looking into *dbaliases* table in LOCATION. #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif In dbaliases table I see 4 columns. Is the following configuration correct? *alias_username*: gonzalo58 *alias_domain*: test.external.com;transport=tls *username*: gonzalo *domain*: sip.parzee.io Thanks -Gonzalo

4 5

Malformed To header, or?
by Tobias 04 Nov '16

04 Nov '16

Hi! Having some issues with Kamailio 4.3 reporting a malformed header, but I cannot seem to figure out what's wrong. The problem seen is: Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: tm [t_lookup.c:1050]: t_check_msg(): ERROR: reply cannot be parsed Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:1457]: adjust_clen(): error parsing content-length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:2214]: generate_res_buf_from_sip_res(): error while adjusting Content-Length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [forward.c:760]: do_forward_reply(): building failed And the To header seen at this point is: To: +123456789 <sip:123456789@127.0.0.1>;tag=as24ed5606. version: kamailio 4.3.4 (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, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.7.2 I obviously understand Kamailio views this To header as malformed, but what I cannot understand is why? Is it as simple as the number +123456789 is not enclosed with ", i.e. it should be sent as "+123456789"? A similar issue seen, which for some reason doesn't create an ERROR, is when Kamailio receives a reINVITE looking like this which it's supposed to proxy: To: +123456789 <sip:123456789@127.0.0.1>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. which is rewritten into: To: +123456789 <sip:123456789@127*0-)'.:/)&7'...0;user=phone>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. and sent out.. Thanks for your help, /Tobias

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