sr-users June 2018

sr-users@lists.kamailio.org

88 participants
117 discussions

Registration of UACs i n a partially encrypted scenario

by mip FKF

Hello, we are pretty new to SIP and kamailio, we do have some questions regarding the following scenario: We have a number of UACs in a small network which are required to communicate without encryption because the are not able to consume certificates. We want to use kamailio (as a proxy?) to establish an encrypted connection to a backend UAS. 1. Is it possible to directly register the UACs with the UAS eventhough communication between kamailio and the UAS is encrypted ? 2. How do we need to configure kamailio in order to make this scenario work ? Thank you for your suppport

5 years, 11 months

Re: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX

by Floimair Florian

Yes it should, but only if “rewrite_contact” is set to “no” for the endpoint (chan_pjsip). With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of "Wilkins, Steve" <swwilkins(a)mitre.org> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Tuesday, 12. June 2018 at 16:27 To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can’t be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn’t Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

5 years, 11 months

Asterisk inviting Kamailio

by Wilkins, Steve

Hello All, I have noticed that sometimes when a call is made from one endpoint to another through Asterisk via Kamailio, Asterisk sends an INVITE to Kamailio even after the call has been established. Sometimes this does not happen. When it does happen, calls drop. Why would an INVITE be sent back to Kamailio? Also, Pan, I am working on the response you requested yesterday. Thanks you All, -Steve

5 years, 11 months

Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio

by Wilkins, Steve

Got it working. Thank you everyone. -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 3:06 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio >>>Email originates from a non-MITRE system. Use caution.<<< Alex, Pan, Daniel,... Could this group => group:BUNDLE audio video in Message Body have anything to do with my Kamailio Video issue. Thank you!! -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 12:14 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Hi Alex, No, I'm not using rtpengine. It must definitely be some sort of Codec issue though, since I can seem to move the problem around and actually finally get video on the softphone. It was a little strange that I lost Audio on the WebRTC client though, since I only disabled VP8. Since Kamailio is only relaying I am so confused why introducing Kamailio is messing up Video. Thank you, -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov Sent: Sunday, June 10, 2018 11:46 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Unless you're using rtpengine, you're just moving the problem around. Kamailio does nothing with SDP unless by way of rtpengine or told in some other way, such as sdpops. On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve" <swwilkins(a)mitre.org> wrote: >I just did a test where I disabled VP8 in Kamailio using SDPOPS and I >now get Video on the softphone however, I lost two-way Audio. Kamailio >seems to be doing something with the codecs but still can't put my >finger on it. The WebRTC Client, who is the caller, needs VP8 for >Video, and apparently Audio. > >I'm not sure if I getting closer or just moving the problem around. > >Thanks All! > >From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf >Of Wilkins, Steve >Sent: Saturday, June 9, 2018 12:47 PM >To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> >Subject: [SR-Users] No Video between WebRTC Client and Softphone when >using Kamailio...works without Kamailio > >>>>Email originates from a non-MITRE system. Use caution.<<< >I am desperately trying to resolve a big issue I have when using >Kamailio 5.2.0 and Asterisk 15.3. > >As soon as I go through Kamailio, I get no Video on either side of the >call; I do get two-way audio and the call stays connected. If I simply >go right through Asterisk, I also get two-way Video. I am having a >tough time determining why Kamailio is messing with the Video portion >of the call. >I have included a full pcap file. As a side note when comparing pcap >traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from >Asterisk and Softphone are exactly the same. > >Thank you, >-Steve -- Alex -- Sent via mobile, please forgive typos and brevity. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

5 years, 11 months

Question on Contact Header on Kamailio 200 OK to PBX

by Wilkins, Steve

Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can't be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn't Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

5 years, 11 months

Facing Problem Registration / Invite pass through one side voice

by Ankit vasava

Hello , I have installed opensips latest version and have free switch. All user is registered with free switch through opensips like registration pass though you can find out my config file into attachments also pcap log. My free switch IP is *192.168.1.147:5060 <http://192.168.1.147:5060>* Opensip IP is : *192.168.1.12:5069 <http://192.168.1.12:5069>* I m sending registration and invite request to public IP and that public IP is bind to open sips. *Problem:* when external user who is registered through public ip when he is dial any extension number who is registered with local ip at that time we are facing one way voice. I need voice(RTP) could be transfer both the side. -- *Regards,Ankit vasava* Sr. VoIP Developer *Phone: *+91-9879491525 *Skype: *ankit.vasava Http://in.linkedin.com/in/ankitvasava

5 years, 11 months

unable to Kamailio IMS 5.1.4?

by vinay kumar

Hi All, Im getting following error when i start kamailio.cfg of icscf. ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] ERROR: ctl [ctl.c:290]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed ERROR: <core> [core/sr_module.c:990]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) can anyone help me on this?? Regards, Vinay

5 years, 11 months

unable to bring up kamailio 5.1.4

by vinay kumar

Hi, Im facing following issue when I tried to bring up kamailio IMS 5.1.4. ERROR: <core> [core/modparam.c:140]: set_mod_param_regex(): parameter <hashing_type> of type <2> not found in module <ims_usrloc_pcscf> CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 354, column 47: Can't set module parameter can anyone share complete installation steps for Kamailio IMS 5.1.4 for volte call?? Regards, Vinay

5 years, 11 months

Problem with TLS

by Arik Halperin

Hello, I’m using TLS After receiving 200OK from kamailio: r2voip.clear2voipdialer I/(NativeSdk_2_0) 1528174138320 PJSIP: (NativeSdk_2_0) 1528174138320 PJSIP:2018-05 07:48:58.319 pjsua_core.c RX 2203 bytes Response msg 200/INVITE/cseq=8107 (rdata0x7a2c56fb38) from TLS 70.36.25.65:443: SIP/2.0 200 OK Via: SIP/2.0/TLS 10.134.232.109:44097;received=109.253.173.146;rport=31373;branch=z9hG4bKPj4MV5llP9SW5ufk-OcFB-Qh78PmIQFrRk;alias Record-Route: <sips:10.168.10.227:5099;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> Record-Route: <sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> From: "number" <sips:972523391991@kamprod.telemessage.com<mailto:972523391991@kamprod.telemessage.com>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO To: <sips:1111111@kamprod.telemessage.com<mailto:1111111@kamprod.telemessage.com>>;tag=64H63g861ajHj Call-ID: Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2 CSeq: 8107 INVITE Contact: <sip:1111111@10.168.10.200:5080;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.6.20+git~20180123T214909Z~987c9b9a2a~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: ti PJSIP responds with: Secure dialog requires SIPS scheme in Contact and Record-Route headers, ending the session What is the reason for this? How can I fix this issue? Thanks, Arik Halperin

5 years, 11 months

Disable basic authentication?

by Skip Morse

Hi All, I couldn't get a clear answer from the documentation. Based on the description of www_authenticate and proxy_authenticate: " The function verifies credentials according to RFC2617. If the credentials are verified successfully then the function will succeed and mark the credentials as authorized (marked credentials can be later used by some other functions). If the function was unable to verify the credentials for some reason then it will fail and the script should call www_challenge which will challenge the user again." The RFC outlines basic and digest auth. Is there a way to disable the ability for a UA to use basic auth? Or maybe these is disabled already? I'm likely understanding this wrong, it seems to me that it's allowed for a UA to auth with basic even if the challenge is for digest. Any clarification would be helpful. Thanks! -Skip

5 years, 11 months

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sr-users June 2018