- sr-users - kamailio.org

presence - Error deleting subscription
by Jan Gaida 03 Jan '15

03 Jan '15

Hello, We are upgrading our servers to Kamailio 4.2 and we see a lot of error messages "Failed to delete subscription from memory". Scenario: This happens every time there is a Notify Timeout (408) and Kamailio tries to delete the subscription. Consequences: I don't know whether the delete fails because the entry did never exist, was already deleted or because the comparison (to_tag, callid or from_tag) is incorrect. It would be nice to at least have some details in the log message. I have not noticed any consequences to general functionality. I suspect the commit (74d73953327ad7702922b27ba37c71e3283659c3) to be responsible for this error because it introduced some extra checks on deletion. Does this happen to anybody else? What may be configured incorrectly on our system? Kind regards *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida(a)grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario.

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t_suspend()/t_continue() problem
by Dmytro Bogovych 02 Jan '15

02 Jan '15

Greetings. I'm trying to adopt kamailio to handle incoming calls / generate push notifications for softphone running on Windows Phone 8. Starting point was this publication http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin… I adopted the script from publication a bit - now notification URI is extracted from REGISTER messages + INVITE transactions are not resuming when unregistering happens. However it does not work for me yet. I see the suspicious output in kamailio log when softphone registers after incoming push notification. Please ignore ERROR: level name - it is just dirty logging call, not real problem. Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: ERROR: <script>: resuming trasaction [40951:63820139] dbogovych (40951:63820139) Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: tm [t_lookup.c:1635]: t_lookup_ident(): DEBUG: t_lookup_ident: transaction found Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:181]: lookup(): contact for [dbogovych] found by address Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 90211984 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91061218 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91229062 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 90750343 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 87612750 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91841578 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: WARNING: tm [t_lookup.c:1537]: t_unref(): WARNING: script writer didn't release transaction Can this "script writer didn't release transaction" be the problem? Thank you :)

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Happy New Year!
by Daniel-Constantin Mierla 01 Jan '15

01 Jan '15

A very dynamic 2014 for Kamailio has reached its end! Thank you everyone for contributing to it! Looking forward to 2015, a lot of new features in Kamailio and new contributors! I wish a healthy and prosperous year to all Kamailio friends, hoping to meet many of you at Kamailio World Conference and other events around the globe! Happy New Year! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Making media packets associated with calls appear unidirectional using rtpproxy and kamailio
by aft 31 Dec '14

31 Dec '14

Hi, I'm trying use two instances of rtpproxy listening to two different public IPs, to handle incoming and outgoing legs. That means i want to implement this network topology: SipClient--------->media from client to proxy--->rtpproxy1 Sipclient<----------media from proxy to client<---rtpproxy2 Where rtpproxy1 listening to IP1 and rtpproxy2 listening to IP2 in kamailio cfg : modparam("rtpproxy", "rtpproxy_sock", " udp:127.0.0.1:22222 udp: 127.0.0.1:22223") if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } rtpproxy_offer("IP1"); } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } rtpproxy_answer("IP2"); } But its not working. Thanks in advance

1 0

kamailio 4.2.1 snmpstats module depends on libsnmp15
by René Bormann 31 Dec '14

31 Dec '14

hello guys, just wanted to add the snmpstats modules to our kamailio 4.2.1 installation. but we are running into a libssnmp15 dependency. searching google showed me that on 14.04 the lib is called libsnmp30 now. does anybody know how to fix this? thanks, rené ------------------- root@kamailio1:~# apt-get install kamailio-snmpstats-modules Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: kamailio-snmpstats-modules : Depends: libsnmp15 (>= 5.4.3~dfsg) but it is not installable E: Unable to correct problems, you have held broken packages. root@kamailio1:~#

2 1

Dialog_end from event route
by Dan-Cristian Bogos 30 Dec '14

30 Dec '14

Hey Guys I need assistance for the following case: I receive dialog_end request from external component in evapi:message-received block and would like to somehow kill the dialog corresponding to h_id and h_entry I get in the message. Is there any way to do that without involving mi_*? Any tip appreciated! Ta, DanB.

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about kamailio
by ThatOther Dev 30 Dec '14

30 Dec '14

is it possible with ypur server project to call to country mobile networks just from a pc, without special equipment?

2 1

ul.dump failing when more than 26 users are registered
by João Vitor Arruda 30 Dec '14

30 Dec '14

Hello, I´m doing a test with the ul.dump output on both Kamailio 4.1.5 or 4.2.1 and it works great if the number of registered users is 26 or lower. Starting with 27 registered users the "kamcmd ul.dump" command returns the error: "error: 500 - Internal error creating aor struct" In Kamailio 4.2.1 the new (and nice) "kamcmd stats.get_statistics usrloc:" returns the correct number of registered_users without this "26 limit". Its possible to check why ul.dump is failing this way? Thanks,

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'from' user != auth username
by Dima T. 30 Dec '14

30 Dec '14

Hello, I need migrate users from the old proprietary softswitch but they have predefined from/to header which is differ from auth username. I read about flags in auth_check but I need strict pairs from user/auth username. How this authentication scheme could be implemented with Kamailio? -- Dmitry

2 1

kamailio configuration problem
by javier falbo 30 Dec '14

30 Dec '14

Hi, I would like some help from a kamailio expert. I have installed on a Debian7 64 kbytes VPS this tutorial: http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour, Problem is that there is no voice in a zrtp connection, but i hear only the ring sound ok. Strange is that on my local wifi works all perfect including the voice, but with external ip no. Any help? I would like to hire a kamailio expert for fixing this and also adding some extras. Write me directly so we could get in contact. regards, javier

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Re: [SR-Users] bug in allow_source_address?
by Henry Fernandes 29 Dec '14

29 Dec '14

Does anyone have feedback about this? -H From: Henry Fernandes <henry(a)usinternet.com> Date: Wednesday, December 17, 2014 at 3:39 PM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: bug in allow_source_address? I¹m using allow_source_address in my Kamailio config and have been for months/years. I¹ve had no issues with it until today. The issue is that allow_source_address("1²) returns ³1" (for $si=123.17.29.201 and $sp=5060) even though we don¹t have this IP address in the address¹ table for group 1. We have defined this IP address for group 2 but not group 1. select * from address order by mask; +----+-----+---------------+------+------+----------+ | id | grp | ip_addr | mask | port | tag | +----+-----+---------------+------+------+----------+ | 20 | 1 | 123.17.0.0 | 23 | 5061 | Testing | | 22 | 1 | 123.17.0.0 | 23 | 5062 | Testing | | 23 | 1 | 123.17.0.0 | 23 | 5063 | Testing | | 24 | 1 | 123.17.0.0 | 23 | 5064 | Testing | | 1 | 1 | 123.17.0.0 | 23 | 5060 | | | 4 | 1 | 20.5.176.0 | 24 | 5060 | | | 21 | 1 | 123.17.1.235 | 32 | 5060 | Override | | 46 | 2 | 123.17.29.201 | 32 | 5060 | customer | +----+-----+---------------+------+------+----------+ Can anyone explain this? I¹m wondering if there¹s a bug in how allow_source_address does its net mask calculations. -H

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about destination blacklisting
by Juha Heinanen 29 Dec '14

29 Dec '14

i made a test where domain name in request uri hostpart has two udp srv records, one with higher priority than the other. when i turn off the higher priority uac, there is about 5 sec delay before the lower priority uac starts to ring. if i make another call within a small number of seconds from the previous one, my sip proxy remembers that the higher priority uac is dead and the lower priority one starts to ring immediately. but if i wait about five seconds or more, my sip proxy again first tries the higher priority uac and there is again the about 5 second delay before the lower priority one starts to ring. i have in config use_dst_blacklist=on dst_blacklist_expire=20 from the docs i get the impression that my sip proxy should remember 20 seconds that the higher priority uac is dead, but it only remembers it about 5 seconds. what could be the explanation? -- juha

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RTPProxy metod UPDATE
by Kalala Alexander 29 Dec '14

29 Dec '14

1 0

What's happening? Follow us on more channels than e-mail!
by Olle E. Johansson 29 Dec '14

29 Dec '14

Friends, The Kamailio project has great mailing lists for discussions, but did you know that we also spread news in various other channels? If you use these channels, please follow us there and spread the word about Kamailio! Facebook https://www.facebook.com/Kamailio Twitter @kamailioproject Google plus has both a Kamailio page and a discussion group - Page https://plus.google.com/u/0/b/114984922071338017631/114984922071338017631/p… - Group https://plus.google.com/communities/104535904708990386927 In addition to the news from the web site there are a few additional tips and tricks coming on these channels. See you there! /O

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SIPURI sending private IP address
by Cibin Paul 28 Dec '14

28 Dec '14

Hi, I have a kamailio 4.1 as a gateway and registrar behind asterisk servers. On asterisk, I save the IP address of the originating call parsing the SIPURI or SIPCHANNEL(recvip). In both cases, I am receiving the private ip of the user agent registered with kamailio. Do I need to change anything in kamailio to receive the actual IP address. Please suggest. Regards Cibin

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Meddling with password during authentication
by Olli Heiskanen 27 Dec '14

27 Dec '14

Hello all, During authentication, is there any way to affect the password user is sending? I do suspect not as it is a clear security matter, but won't hurt to ask. I use auth_db module with calculate_ha1 parameter set to 1. For reasons in integrating Kamailio into my system architecture there is a need to store a password in some other format than for example md5('555:domain.com:password)') while not allowing any passwords to be stored as plaintext. For example: md5('555:domain.com:md5('password')') but this would require me to hash the password before authentication, in Kamailio script as I can't do it in the clients. Reason for this question is to have my users in a separate database, and these users could have 0-n sip peers assigned to them, and have users authenticate to my software and the sip peers using the same password. cheers, Olli

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Authorization Issue kamailio 4.2
by Ahmed Salem 27 Dec '14

27 Dec '14

Hi, I'm using kamailio 4.2 some of my clients are getting 401 unauthorized after a couple of days of usage without any problems, their usernames and passwords are still in the database server unchanged. Any idea what's going on? Thanks Ahmed

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Merry Christmas!
by Daniel-Constantin Mierla 25 Dec '14

25 Dec '14

Another year getting slowly to the end, an excellent opportunity to thank to all old and new friends of Kamailio project, from contributing code to using it, helping on mailing lists and other places online, or advertising it across the world! Merry Christmas and Happy Winter Holidays! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Media trouble with kamailio/rtpengine
by Marc Soda 24 Dec '14

24 Dec '14

I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not getting audio back to my browser. From a packet capture I can see media from the browser to rtpengine, and then bi-directional RTP back and forth from my asterisk server, but rtpengine is not sending the media on to the browser, i.e.: browser ---------> kamailio/rtpengine <---------> asterisk This is the output from rtpengine: https://gist.github.com/marcantonio/bfe72644306b205cc7e1 Thanks.

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Dispatcher weight nor working
by Yuriy Gorlichenko 24 Dec '14

24 Dec '14

Hello. We use kamailio 4.3 and dispatcher with 9 algorithm. We use db instanse for dispatcher and at attrs column for our backend servers set WEIGHT=40 and 60 for first and second server, but packets sended only at first server ignoring weight

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Re: [SR-Users] Mediaproxy do not work well with Kamailio SIP server issue
by Alfred C.M. Lu 24 Dec '14

24 Dec '14

|---------+-----------------------------> | | dong-ming.li(a)mail.| | | foxconn.com | | | | | | 2014/12/24 上午 | | | 11:54 | |---------+-----------------------------> >--------------------------------------------------------------------------------------------------------------------------------------------------| | | | To: alfred.lu(a)foxconn.com | | cc: | | Subject: Mediaproxy do not work well with Kamailio SIP server issue | >--------------------------------------------------------------------------------------------------------------------------------------------------| Secret Level: |------------------------| | ( ) 1.Non-Confidential | | ( ) 2.Confidential | | ( ) 3.Secret | | ( ) 4.Top Secret | |------------------------| Priority Level: |--------------------| | ( ) 1.Normal | | ( ) 2.Urgent | | ( ) 3.Extra Urgent | |--------------------| ------------------------------------------------------------------------- Hi, sir, We try to setup Mediaproxy on CentOS 6.6 with kamailio-4.2.1 SIP server. We use Linphone android version as SIP client to test. attached kamailio.cfg.txt is our kamailio-4.2.1 SIP server config file. pls help to rename kamailio.cfg.txt to kamailio.cfg. kamailio-4.2.1 SIP server is run on public IP server,and Linphone SIP client is run behind NAT. The problem is that only callee can hear caller's rtp audio data. caller can not hear callee's audio data. And the call will auto disconnect after about 30s time out. Below's our modification regarding the kamailio.cfg file,pls help to check if this is OK or not ? #foxconn gallice add start #### MediaProxy module loadmodule "mediaproxy.so" modparam("mediaproxy", "disable", 0) modparam("mediaproxy", "ice_candidate", "high-priority") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") modparam("mediaproxy", "mediaproxy_timeout", 500) modparam("mediaproxy", "signaling_ip_avp", "$avp(nat_ip)") modparam("mediaproxy", "media_relay_avp", "$avp(media_relay)") #foxconn gallice add end # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting use_media_proxy(); #foxconn gallice add } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { #foxconn gallice add #route(NATMANAGE); #foxconn gallice delete use_media_proxy(); #foxconn gallice add } #foxconn gallice add } For mediaproxy,we can see the media related processes are already run. 31212 ? SL 0:01 python ./media-dispatcher start 31218 ? SL 0:16 python ./media-relay start ***************************************************** dong-ming / 李東明 Foxconn Electronics Inc. HH Precision Ind. Co., Ltd. CNSBG NSDI TIPBU R&D Dept. LH Office 地址:深圳市龍華新區龍華東環二路二號富士康科技集團分機 : +86-0755-2812-9588 #26109 Mobile : 13510123670 Mail : dong-ming.li(a)mail.foxconn.com, ***************************************************** 溫馨提醒﹕入廠人員須遵守富士康出入管制,請勿攜帶手提電腦、各種攝像、U盤、MP3 等存儲設備本電子郵件及附件所載信息均為保密信息，受合同保護或依法不得洩漏。其內容僅供指定收件人按限定範圍或特殊目的使用。未經授權者收到此信息均無權閱讀、使用、複製、洩漏或散佈。若您因為誤傳而收到本郵件或者非本郵件之指定收件人，請即刻回覆郵件或致電Super Notes郵件客服熱線 560-104，並永久刪除此郵件及其附件和銷毀所有複印件。謝謝您的合作！ This e-mail message together with any attachments thereto (if any) is confidential, protected under an enforceable non-disclosure agreement, intended only for the use of the named recipient(s) above and may contain information that is privileged, belonging to professional work products or exempt from disclosure under applicable laws.Any unauthorized review, use, copying, disclosure, or distribution of any information contained in or attached to this transmission is STRICTLY PROHIBITED and may be against the laws. If you have received this message in error, or are not the named recipient(s), please immediately notify the sender by e-mail or telephone at Super Notes support hotline 560-104 and delete this e-mail message and any attached documentation from your computer. Receipt by anyone other than the intended recipient(s) is not a waiver of any attorney-client or work product privilege. Thank you! mail from ip-->192.168.2.2 mail from pc-->MICROSOF-700F48 Version: Super Notes 1.6.4.5B (See attached file: kamailio.cfg.txt)

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TLS enable false.
by Thanh Truong 23 Dec '14

23 Dec '14

Hi all, I have tried several configure TLS in kamailio but no luck. Please give me some suggestion that I can make it work correctly. This is my configure in TLS module. modparam("tls", "tls_method", "SSLv23") modparam("tls", "private_key", "/usr/local/etc/kamailio/ca/privkey.pem") modparam("tls", "certificate", "/usr/local/etc/kamailio/ca/kamailio1_cert.pem") modparam("tls", "ca_list", "/usr/local/etc/kamailio/ca/calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) I am only getting issue with verify_certifiate = 1, it i let it to 0, my client can register correctly. When I set this flag, i got message from server as: Dec 18 10:26:31 17237 /usr/local/sbin/kamailio[12655]: ERROR: <core> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: tls [tls_server.c:1193]: tls_read_f(): TLS accept:error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: <core> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading I Cant add any pem file into client, i have used Blink phone but no luck. Thank all in advance. ThanhTruong.

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[BOUNTY] Configs for Video calls between Linphone Android App and WebRTC clients (JsSIP, SIPML)
by Andrey Utkin 23 Dec '14

23 Dec '14

I hope the subject line says it all. I need to set up a backend to establish calls between browser clients and usual SIP clients, and the abovementioned applications are of the highest interest. (Of course, calls between same UAs must work, too.) If Linphone is not going to work due to its internal issues, then other established open source mobile apps for iOS and Android will be fine. Currently I have configured latest kamailio+rtpengine with configs from here https://github.com/caruizdiaz/kamailio-ws The calls pass this way currently: jssip -> android: no sound from phone to the browser, i see that jssip sends "sendonly" attribute for audio in INVITE's SDP. Audio from browser to phone, and both video streams appear immediately, everything is fine with them. android -> jssip: video from browser to the phone appears in 1-2 _minutes_ after the call is answered. All other media streams are fine. The above results are the same in such browsers [IP-] [ ] www-client/firefox-bin-31.3.0:0 [IP-] [ ~] www-client/google-chrome-unstable-41.0.2243.0_p1:0 With sipml currently i have no stable results, so it's hard to describe what happens. Please contact me if you have configs to make the needed things work, or if you have experience of such things working stable, and can configure it quickly. -- Andrey Utkin

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websocket ws_handle_handshake() error
by jaflong jaflong 23 Dec '14

23 Dec '14

Hi list, I am getting this error "ERROR: websocket [ws_handshake.c:153]: ws_handle_handshake(): retrieving connection" Any idea why? This is the debug log 12(24113) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... 12(24113) DEBUG: xhttp [xhttp_mod.c:358]: xhttp_handler(): new fake msg created (537 bytes): <GET /ws HTTP/1.1 Via: SIP/2.0/TLS 10.1.1.1:45968 Host: 10.1.1.2:10443 User-Agent: Mozilla/5.0 (X11; Linux x86_64; rv:34.0) Gecko/20100101 Firefox/34.0 Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 Accept-Language: en-US,en;q=0.5 Accept-Encoding: gzip, deflate Sec-WebSocket-Version: 13 Origin: http://mydomain.co.uk Sec-WebSocket-Protocol: sip Sec-WebSocket-Key: GRbV+ovG+DKIBJtwb4pRiQ== Connection: keep-alive, Upgrade Pragma: no-cache Cache-Control: no-cache Upgrade: websocket > 12(24113) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: 12(24113) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <GET> 12(24113) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: </ws> 12(24113) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <HTTP/1.1> 12(24113) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 12(24113) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 12(24113) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via 12(24113) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 12(24113) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header 12(24113) DEBUG: <core> [msg_translator.c:205]: check_via_address(): check_via_address(10.1.1.1, 10.1.1.1, 0) 12(24113) DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (24113 (12)), reusing fd 12(24113) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... 12(24113) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fc08162ad68 n=421 fd=7 12(24113) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf= �/�*�޹��!��S��zo�e�/D��V�Gs��s�� ^cd��S"Ŏ�K��<�c7��4lx�<Z��#�c(AJ�8V��d� ��4bF��Wu�(�w�a� f��n��s�KjQv�݈�q|Z��64{Wk��Z\ױ��M�,[�Arٵ��Q�_��u�d�Gza�f��I�fI��z��Š9=��Ħ3�ȀJPx�}��S?��eVe��\id��y�wP�r(kZs��Z�"��4`�N_MR��z�!��Eb}��+�\;�8�3��V��l��M�|�1�2��VVK�@%�a�B&�P��k 12(24113) INFO: <script>: Received HTTP request to /ws from [10.1.1.1:45968] with protocol tls 14(24115) DEBUG: <core> [tcp_main.c:3565]: handle_ser_child(): handle_ser_child: read response= 7fc08162ad68, -1, fd -1 from 12 (24113) 14(24115) DEBUG: tls [tls_server.c:604]: tls_h_close(): Closing SSL connection 0x7fc081626e20 12(24113) ERROR: websocket [ws_handshake.c:153]: ws_handle_handshake(): retrieving connection 12(24113) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 12(24113) DEBUG: <core> [msg_translator.c:205]: check_via_address(): check_via_address(10.1.1.1, 10.1.1.1, 0) 12(24113) ERROR: sl [../../forward.h:247]: msg_send(): tcp_send failed 12(24113) ERROR: websocket [ws_handshake.c:113]: ws_send_reply(): sending reply 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) 12(24113) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up 12(24113) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0xa31f40, 7, -1, 0x10) fd_no=2 called 12(24113) DEBUG: <core> [tcp_read.c:1434]: release_tcpconn(): releasing con 0x7fc08162ad68, state -2, fd=7, id=3 12(24113) DEBUG: <core> [tcp_read.c:1435]: release_tcpconn(): extra_data 0x7fc081626e20 14(24115) DEBUG: <core> [tcp_main.c:3331]: handle_tcp_child(): handle_tcp_child: reader response= 7fc08162ad68, -2 from 2 9(24110) DEBUG: websocket [ws_conn.c:459]: wsconn_get_list(): wsconn_get_list

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Rpc dialog variable
by Pars3c 22 Dec '14

22 Dec '14

Hi How can i do to see the dialog variables like $dlg_var with RPC ? It's possible? Thanks

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