sr-users March 2015

sr-users@lists.kamailio.org

88 participants
149 discussions

We’ve got free Tickets for CeBIT 2015!
by Carsten Bock 10 Mar '15

10 Mar '15

Hi, Due to our presentation at CeBIT 2015 at the Open-Source Business Alliance’s Booth, we’ve received a limited number of free tickets for CeBIT 2015! If you want a free ticket, please answer the following question: - What is the key benefit of deploying VoLTE in a mobile network? Send your answer to info(a)ng-voice.com (hint: we’ve got some benefits posted on our website: www.ng-voice.com/volte), I will send the ticket by email to the 5 best answers! Don’t miss my presentation on Kamailio & VoLTE at the booth of the Open-Source Business Alliance (www.osb-alliance.de), Tuesday, March 17th, ~2 PM, Hall 6, Stand H16, (410)! Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-0 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

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WARNING when querying data using db_mongodb
by Mickael Marrache 10 Mar '15

10 Mar '15

Hi, I'm continuously seeing the following WARNING in my logs: WARNING: db_mongodb [mongodb_dbase.c:454]: db_mongodb_get_columns(): unhandled data type column (instance) type id (10), use DB1_STRING as default It looks like the WARNING appears when data is fetched from the location collection. The instance field is always null in my case. Thanks, Mickael

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Regd. Kamailio GPL
by Shankar 10 Mar '15

10 Mar '15

Hello, We are exploring kamailio source for use in our VOIP solution. This is the first time we are looking at open source. We have few doubts in using kamailio for providing our VOIP service. Under GPL any customization (customization mainly with respect to interacting with our proprietary applications e.g. Billing server) we do to kamailio also has to be provided to the end user. Our doubt is whether we can procure commercial rights to our customised code? Kindly clarify on this. Best Regards, Shankar

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Dialog order of operations
by Alex Balashov 10 Mar '15

10 Mar '15

Hi, The dialog module documentation remains unclear about the order of operations with regard to when to call dlg_manage() or set the transaction flag. My impression is that dlg_manage() only registered TM callbacks, so it doesn't matter when you call it, as long as it's before t_relay(). However, the documentation neither confirms nor denies this. So, this raises the questions: 1) Is this okay? set_dlg_profile("caller", "$fU"); dlg_manage(); ... t_relay(); Or do I need to do this? dlg_manage(); set_dlg_profile("caller", "$fU"); ... t_relay(); 2) What about setting dialog-persistent variables? Is this okay? $dlg_var(account_id) = 49555; dlg_manage(); If so, where does the variable go if I never call dlg_manage() because the call is aborted beforehand, e.g. $dlg_var(account_id) = 49555; sl_send_reply("403", "Forbidden"); exit; dlg_manage(); ... t_relay(); 3) Any other gotchas or caveats in relation to the order of operations? I suppose my preference would be to set the dialog profiles in various places throughout call processing and call dlg_manage() at the very end, right before t_relay(). Is this acceptable? Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Announcing rtpproxy v2.0.0
by Maxim Sobolev 09 Mar '15

09 Mar '15

Hi All, I'm happy to announce that we have released rtpproxy v2.0.0. You can review the release notes here: https://github.com/sippy/rtpproxy/releases/tag/v2.0.0 -sobomax

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Enable SNMP causes issue when forking exec_avp request
by Daniel Goepp 09 Mar '15

09 Mar '15

I have an issue that from what I can tell is related to forking. My configuration works just fine, and makes a request out to execute curl to hit an external API: $var(my_query) = "curl -m 2 -s http://x.x.x.x/service/method?fromAddr=" + $var(fu) + "\&toAddr=" + $var(tu); Normally this gives us back a nice JSON response, no problems. All I do is enable snmpstats module, and it fails. Mar 5 16:51:30 server-name /usr/sbin/kamailio[1051]: ERROR: exec [exec.c:291]: exec_avp(): cmd curl -m 2 -s http://x.x.x.x/service/method?fromAddr=me@mydomain.com\&toAddr=someone@mydo… failed. exit_status=-1, errno=10: No child processes I have edited this for security reasons of course. If I disable SNMP, the problem goes away. Thoughts?

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rtpproxy failed to load
by Tom Braarup Cuykens 09 Mar '15

09 Mar '15

kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I found this guide for an older version of Kamailio, but there seems to be some issues when trying to use it on the latest stable version. When I try to start Kamailio i get this : [....] 0(17153) INFO: tls [tls_init.c:399]: init_tls_compression(): tls: init_tls: disabling compression... 0(17153) ERROR: <core> [cfg.y:3286]: yyparse(): cfg. parser: failed to find command force_rtp_proxy (params 0) 0(17153) : <core> [cfg.y:3426]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 729, column 19: unknown command, missing loadmodule? 0(17153) ERROR: <core> [cfg.y:3286]: yyparse(): cfg. parser: failed to find command force_rtp_proxy (params 0) 0(17153) : <core> [cfg.y:3426]: yyerror_at(): parse error in config fil[FAILc/kamailio/kamailio.cfg, line 806, column 19: unknown command, missing loadmodule? ERROR: bad config file (2 errors) ... It seems it cannot load the rtpproxy module or that force_rtp_proxy command does not exist. For Debian the modules are located in /usr/lib/x86_64-linux-gnu/kamailio/modules/ and it was found there, seems to be loaded in the config file (line 5 and 235). According to the rtpproxy doc the module has this command. Does anyone have an idea what could be the issue ? I am planning to update this guide for the current version once I fixed all the issues. The most easy issues are the db/usernames/password and the path to the modules. -- Med venlig hilsen Tom Braarup Cuykens plusTEL.dk

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lua_run seems to be working in asynchronous way
by José Seabra 09 Mar '15

09 Mar '15

Hi there, I'm using kamailio version 4.2.2 to a short time, and I'm using lua scripts to build some avps with information needed to make calls(lua functions uses redis as source data to build avp data.), what i have noticed today making some tests is that when i make a call my lua function: if(!lua_run("get_prefs","$fU","$fd")) { xdbg("SCRIPT: failed to execute lua function!\n"); } kamailio gets the avp's after send the invite to the destine,(the invite will fail because kamailio needs avp information). it seems that kamailio calls the lua_run in asynchronous way, i never noticed that before in old version. is that a normal behavior or it is an issue in this version? Best Regards -- Cumprimentos José Seabra

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Re: [SR-Users] kamailio asterisk
by Slava Bendersky 08 Mar '15

08 Mar '15

Hello Everyone, I set up tcp and udp for right now to test all config and dialog is not established correctly. Call between extensions not working. Call from extesnsion to extension going right a way to voicemail. Any help thank you. Here one call debug. http://fpaste.org/195572/14258553/ Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 3, 2015 9:08:19 AM Subject: Re: [SR-Users] kamailio asterisk Hello Daniel, Is notify should follow subscribe routes ? Here SUBSCRIBE completely look normal to me. <--- SIP read from UDP:10.18.130.46:5060 ---> SUBSCRIBE sip:10102@networklab.ca SIP/2.0 Record-Route: <sip:kamailio_pub_ip:5081;transport=tls;lr=on;ftag=08c99307d3;nat=yes> Accept: application/simple-message-summary Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKdee2.b83c2028dc57c641d2b8cad968347d6a.0;i=2 Via: SIP/2.0/TLS 192.168.0.16:5063;rport=5063;received=client_pub_ip;branch=z9hG4bK0c54032bd8905d262 Max-Forwards: 69 From: "10102" <sips:10102@networklab.ca>;tag=08c99307d3 To: <sips:10102@networklab.ca> Call-ID: cea703eb8c21f709 CSeq: 1096440084 SUBSCRIBE Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Allow-Events: message-summary Authorization: Digest username="10102",realm="networklab.ca",nonce="29ac01f1",uri="sips:10102@networklab.ca",response="0298c2a93bdd4969a3cd0b757d671dbb",algorithm=MD5 Contact: <sips:10102@client_pub_ip:5063>;audio Event: message-summary Expires: 3600 Supported: eventlist, replaces, timer User-Agent: Media5-fone/4.1.6.3283 Android/4.1.2 Content-Length: 0 Path: <sip:outbound@10.18.130.46;lr;received=sip:client_pub_ip:5063%3Btransport%3Dtls> P-hint: outbound Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 3, 2015 1:50:03 AM Subject: Re: [SR-Users] kamailio asterisk Hello, I pushed a fix for it -- the issue was in a LM_DBG() trying to print te address for local socket, but it was not set yet, it was affecting only when running in debug mode with debug=3. I pushed the patch to branch 4.2, so if you want the fix, then you have to install from git branch 4.2. For REGISTER, you need to add the Path header (see path module), otherwise the OPTIONS and INVITE sent out to the phone are not going via Kamailio. You have to check the version of your Asterisk to be one that supports Path. Cheers, Daniel On 27/02/15 14:40, Slava Bendersky wrote: Hello Daniel, Here paste for gdb http://fpaste.org/191338/25043949/ I got REGISTER and SUBSCRIBE start working correctly I see on asterisk correct record routes and sip traffic flow, but when asterisk or client ( soft phone) send OPTIONS or NOTIFY can't get properly relay it. This is SUBSCRIBE route. <--- Transmitting (NAT) to 10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060 Via: SIP/2.0/TLS 192.168.88.252:5062;received=Client public ip;branch=z9hG4bK0bbe1f7d27257bba9;rport=5062 Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes> Record-Route: <sip:PUBLIC_KAMAILIO_IP:5081;transport=tls;r2=on;lr=on;ftag=a185d974ec;nat=yes> From: "Slava Bendersky" <sips:10101@networklab.ca> ;tag=a185d974ec To: <sips:10101@networklab.ca> ;tag=as00757d3e Call-ID: b08adb1ad1804a83 CSeq: 236711034 SUBSCRIBE Server: FPBX-2.11.0(11.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:10101@10.18.130.50:5060> ;expires=3600 Content-Length: 0 Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, February 27, 2015 6:42:33 AM Subject: Re: [SR-Users] kamailio asterisk Hello, I asked for the wrong command, as the bt full was already sent before -- I wanted the output from gdb for: p *tcpconn Daniel On 27/02/15 04:10, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here bt full from back trace. http://fpaste.org/191207/50064491/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, February 26, 2015 9:56:48 PM Subject: Re: [SR-Users] kamailio asterisk Hello Daniel, I tried $rz option on top of request route and that where I see wrong request uri like sip:sips : . And as far I can see it happenes only for SUBSCRIBE INVITE and NOTIFY. if($rz=="sips") { $ru = "sip" + $(ru{s.substr,4,0}); } Slava. Sent from mobile device typos are expected. From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Feb 25, 2015 1:04 PM To: Slava Bendersky Cc: sr-users Subject: Re: [SR-Users] kamailio asterisk Hello, On 25/02/15 17:19, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, substr you suggested didn't worked. See my previous email. your previous email didn't say anything about the results. That's why I asked. Be sure you don't have those spaces that are in the email you wrote. Also, I had more parenthesis in the parameter of the subst_uri(). Or you can try the alternative with: if($rz=="sips") { $ru = "sip" + $(ru{s.substr,4,0}); } I asked for more details from the backtrace to confirm that what I found is the cause for the crash in this case -- see one of my previous emails from today. Cheers, Daniel BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Group call advice: remove calling branch after alias_db_lookup/lookup_branches
by Anthony Messina 08 Mar '15

08 Mar '15

In terms of implementing "group" calling via the append_branch feature of both alias_db_lookup followed by lookup_branches, I'm looking for a reliable way to ensure that if the caller happens to be a member of the group (list of branches), the branch that's created to the original caller is dropped. Originally, I was thinking of comparing $rU and $fU in branch_route, but this would limit the ability for one contact of an AOR to call another contact of the same AOR. Can anyone offer an example of an efficient method to accomplish this? -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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sr-users March 2015