sr-users November 2015

sr-users@lists.kamailio.org

80 participants
125 discussions

Simple User Agent change
by Steven Maddox 24 Nov '15

24 Nov '15

Hi, Sorry to ask something probably very simple but I'm difficulty finding the right kinds of resources to learn from which are succinct enough to not overload my head. We have a PBX which we don't control (and hate) which expects only certain makes/models phones to register (it checks for user agents). I have a phone which unfortunately doesn't support tweaking its user agent, so I'd like to pass it through an extremely SIP proxy to have it's user agent tweaked and all other data to/from the PBX transparently let through. I've managed to knock up something (that isn't a proxy) using Asterisk where I made a trunk and a peer... but obviously since this isn't a proxy... things like BLF and transferring calls will never work. If anyone can point me in the direction of a really simple set up to achieve this, please let me know :) -- Steven Maddox

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kazoo module documentation
by Vik Killa 24 Nov '15

24 Nov '15

Hello, I'm testing kazoo module for use with RabbitMQ I noticed there is a required parameter that is not listed in the documentation: http://kamailio.org/docs/modules/devel/modules/kazoo modparam("kazoo", "pua_mode", 0) The above parameter is required but not documented. Can someone please update the documentation? Thank you. /V

2 1

Problem with resuming suspended transactions.
by Dmytro Bogovych 23 Nov '15

23 Nov '15

Greetings. We build specialized softphone for iOS. Push notifications are used to notify user and wake up softphone app on missed calls and messages. We used Daniel's example config script as base for own configuration file. It has the same parts and does not differ too much. I talk about this example: http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin… Everything was ok for few monthes - with kamailio 4.1.2. Few weeks ago we upgraded to 4.3.3. Now paused calls cannot be resumed properly. Softphone receives 500 Server error (No error 0/SL) in response This is problem part of script: route[RELAYRESUMED] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if ($var(rr_target) != "") { xlog("L_INFO","Relay to $var(rr_ip):$var(rr_port)\n"); $ru = "sip:" + $rU + "@" + $var(rr_ip) + ":" + $var(rr_port) + ";transport=" + $var(rr_proto); xlog("L_INFO","Current request URI is $ru\n"); if (!t_relay()) xlog("L_INFO","Cannot relay PUSHED invite."); sl_reply_error(); } else { xlog("L_WARN","Target proto is not set\n"); sl_reply_error(); } exit; } It produces log ----- Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: resuming trasaction [23927:345343979] 1985987629 (23927:345343979) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Relay to 95.133.110.17:11791 Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Current request URI is sip:1985987629@95.133.110.17:11791;transport=tls Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (0/SL) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Checking FAILED messages for 1985987629: 0 ----- sl_reply_error() sends 500 with No error (0/SL) description. Later 180 & 200 are sent. But client softphone terminates transaction already (as it handles 500 response code) . There is no "Cannot relay PUSHED invite." message in the log however. I will be happy to heary any idea what can be cause... Thank you!

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Kamailio-SEMS Integration
by Prashant Desai 23 Nov '15

23 Nov '15

Hi, We are working on kamailio-4.2.3 and we require B2BUA functionality to be achieved.From kamailio official website we got to know,we can integrate kamailio with SEMS to achieve B2BUA functionality. We have few queries regarding this integration. 1.SEMS-1.6.0(latest version available) is been used,is this version suits kamailio-4.2.3? 2.SEMS-1.6.0 does not support unix based communication with kamailio-4.2.3,as SEMS-1.6.0 doesn't have unix socket interface plugin. Since we need unix socket based communication with SEMS.can we port the unix socket interface plugin to SEMS-1.6.0?If no,please suggest us better approaches. Regards, Prashant.

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Kamailio, rtpproxy, websocket and 488 error
by Alfredo Pastor 23 Nov '15

23 Nov '15

Hello, I was doing some tests with Elastix MT that use version 4.1.6 of Kamailio and by default uses rtpproxy, I had the problem of error 488 and unable to make calls from websockets (Incompatible SDP). I could see that Peter Dunkley in a video explains that the problem is solved by rewriting the SDP, but he uses the mediaproxy not rtpproxy. What are your recommendations? I hope you can help me. Chreers, Alfredo.

1 0

rtpengine via-branch=extra question
by Juha Heinanen 22 Nov '15

22 Nov '15

rtpengine readme tells: via-branch=... - Include the “branch” value of one of the “Via” headers in the request to the RTP proxy. Possible values are: “1” - use the first “Via” header; “2” - use the second “Via” header; “auto” - use the first “Via” header if this is a request, or the second one if this is a reply; “extra” - don't take the value from a header, but instead use the value of the “extra_id_pv” variable. This can be used to create one media session per branch on the RTP proxy. When sending a subsequent “delete” command to the RTP proxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the “rtpengine_delete”, or stop all sessions for a call when not passing one of those two flags there. if via-branch=extra is used in offers of branched, should it also be used in answers and deletes? -- juha

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Functionality of rtpproxy
by kai.ohnacker＠cbc.de 20 Nov '15

20 Nov '15

Hello, I have installed a kamailio server with rtpproxy. The server is located between 2 networks and bridge these networks. My question: Does handle rtpproxy only 1 call at the same time? Because my configuration works, but only for one call from the first to the second network. The second call is not working. But if I make internal or external call, I can do more than one call. Do I need the multiple rtpproxy usage configuration? I thought not, because I don't use different IPs for the second call... Cheers, Kai

2 1

403 Forbidden for INVITE in kamailio for ims call
by Priyaranjan Nayak 20 Nov '15

20 Nov '15

Hi All, I have installed kamailio and using as a i-cscf,s-cscf and p-cscf server. I have taken FHoSS(OpenImscore) for hss server Now I am able to do register and unregister. When I am sending INVITE for call, the calling-party and called-party were registerd but I am getting "403 Forbidden - Dialog not found on S-CSCF or Terminating user not suitable for unregistered services". Could you please tell me what is the root cause of above error message ? Best regards, Priyaranjan

1 0

RFC: rtpproxy and a=nortpproxy:yes
by Daniel-Constantin Mierla 19 Nov '15

19 Nov '15

Hello, I am looking to get some feedback on some issues I noticed more and more lately. Apparently some SIP devices (media servers, phones, ...) are keeping the "a=nortpproxy:yes" line in SDP when replying to an INVITE that contains such line. [Alice] ------> [Kamailio+RTPProxy] ------> [Bob] The 200ok response from Bob has "a=nortpproxy:yes" in SDP. By default, that line in SDP makes the rtpproxy not to engage itself anymore in rtp relaying, and as a result things like no audio or one way audio happens. Anyone else encountering such situations? If yes, what are the devices with such behaviour? So far I noticed with some FreeSwitch and Snom -- none of them I can control, so there might be a specific configuration of those devices, not something by default there. The solution is to set: modparam("rtpproxy", "nortpproxy_str", "") and use flag 'r' for rtpproxy_manage() if the IP in SDP is not a private address. I already updated the default config for master to use flag 'r' if the SDP media IP is not private, wondering if nortpproxy_str should be set to empty in kamailio.cfg (or made empty as default in config). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 0

rtpproxy database structure for kamailio 4.2.3
by Jan Hazenberg 19 Nov '15

19 Nov '15

Hi all, I'm trying to load the rtpproxy url's from database but get a error when starting kamailio (4.2.3): ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'set_name' in 'field list' Looks like the field set_name is missing in the database structure, but i can't find it in the docs. If i add the field kamailio starts, but what is the field set_name used for. Seems it needs to be a int value: ERROR: rtpproxy [rtpproxy.c:528]: get_rtpp_set(): Invalid set name 'test' - must be integer Thanks, Jan

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sr-users November 2015