sr-users June 2017

sr-users@lists.kamailio.org

79 participants
121 discussions

by Kjeld Flarup

Rumours is that Apple no longer accepts apps which can do persistent connections in the background. To my best knowledge that means that IOS no longer supports SIP incoming calls. The app should now use Push Notifications, but SIP does not support this. Anybody faced this problem? Is the solution to call an external program when processing an Invite? This leaves some issues. 1. Should we wait some seconds before proceeding or wait until the Push notification has been processed, to give the app time to register. We cannot start to send invites, before we have a correct registration. An old registration may change ports when the app wakes up and makes a new register via NAT. 2. How do we know, if a given user should have a push notification? 3. Which information do we need to be able to send the push notification? 4. Does this give a penalty to Android users, because we have to wait for IOS? -- Med venlig hilsen / Best regards Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef Viptel ApS, Hammershusvej 16C, DK-7400 Herning Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk

7 years, 4 months

[mohqueue] MOH service / A case with MOH queue

by Issam Hambouz

Hello! I am exploring mohqueue module and have few questions: - Could it be used to provide MOH service for SIP endpoints? I am aware that its main functionality is to queue calls (along playing a MOH for queued calls) and provides a mechanism to retrieve calls (FIFO), but I had the impression, reading several emails in this group, that it can be used in a way to provide MOH whenever a call is put on-hold -- I tried to use it with the re-INVITE received when a call is put on-hold, but got a log message stating that it works with initial INVITEs only (something like that) - On a test setup (Kamailio 4.2.3 / 2.0.0), I was able to reach the queue / listen to the MOH but encountered a strange case where the call gets automatically removed from the queue after 4-5 seconds (can be seen removed from kamailio.mohqcalls as well) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): mohq_process: Processing INVITE, queue (demo) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): create_call: Added call (<sip:6010@10.110.3.57>;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) to queue (demo) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): first_invite_msg: Making offer for RTP link for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): first_invite_msg: Sent RINGING for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): send_rtp_answer: Answering RTP link for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): start_stream: Starting RTP link for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): send_rtp_answer: Responded to INVITE with RTP for call (< sip:6010@10.110.3.57>;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:35 ERROR: mohqueue [mohq_funcs.c:1243]: invite_cb(): invite_cb: INVITE failed for call (<sip:6010@10.110.3.57 >;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47)! Jun 12 22:58:35 DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): delete_call: Deleting call (<sip:6010@10.110.3.57>;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) from queue (demo) -- route[MOH] is called first at the beginning of request_route() route[MOH] { if (has_totag()) return; if (mohq_process ()) { xlog ("mohqueue: mohq_process"); exit; } } -- Looking at the SIP trace, all seems ok, except receiving a 404 response after BYE (that's me hanging up after few seconds from getting the above error message) Your comments and hints are appreciated! Thank you Issam

7 years, 4 months

Add extra headers on HTTP requests in Kamailio v5

by Joel Serrano

Hi all, Is there a way to add extra headers with http_client_query() from http_client module? In current http_query() from utils module, according to the doc, you can add a custom header: *4. Functions* *4.1. http_query(url, [post-data], [header-data], result)* *...* *If you want to add additional headers to the request, you have to define the “header”-data, that should be submitted in that request as the third parameter. If this parameter is empty, it is not set.* *...* and I have that working with the following: http_query("https://example.com/api", "param1=value1&param2=value2", "Cookie: apikey=super_secret_api_key", "$var(call_info)"); The problem is that http_query() is removed in Kamailio v5, and the http_client_query() only has the ability to add specifically the content-type header: *4. Functions* *4.1. http_connect(connection, url, [content_type, data,] result)* *...* *content_type - Used only when posting data with HTTP POST. An Internet Media type, like "application/json" or "text/plain". Will be added to the HTTP request as a header.* *...* Is there another way to add a custom header to a http request with the http_client module? Any workarounds for this? Thanks, Joel.

7 years, 4 months

remote ip address of sender and receiver

by vuleetu

Hi, all Is there any way to get remote ip address of both sender($si) and receiver(?)? I am configuring kamailio with multiple rtpproxy sets. Basically, rtpproxy set will be choosed based on the country those ip addresses of caller and callee belong to. Lets say if caller and callee are both in US, set 1 will selected. If they are not in the same country, set 2 will be selected. I dont know if there is any way to do that. Best regards Fisher

7 years, 4 months

Segfault on kamailio 5.0.2. Suddenly stops processing traffic, then generates a core when restarting

by Joel Serrano

Hi all, I'm running into a weird situation with our newly deployed Kamailio servers. These servers are only doing SSL offloading and load balancing between another set of servers. Debian v9.0 + Kamailio v5.0.2 (from kamailio50 repository). After some time of running, kamailio suddenly stops responding to any request but the proccess is still alive. When we restart the service, a core is generated. I have opened an issue in GitHub: https://github.com/kamailio/kamailio/issues/1172 I'm attaching the backtrace here too in case anyone can take a look at it, unfortunately I'm lost right now. In dmesg I saw: [138953.744199] kamailio[12052]: segfault at 7efaf32f51b8 ip 00007efcf3b5df69 sp 00007ffe325dc0c0 error 4 in libcrypto.so.1.1[7efcf39f6000+265000] Thanks, Joel.

7 years, 4 months

get online user from api

by 赵国杰

Hello all, I want to know how many users are currently registered to my kamailio server. I understand can be used to achieve this. I'm wondering if there's RESTful or RPC interface that can be requested by external service. Thanks, Jesse

7 years, 4 months

Kamailio 4.3 to 4.4

by Laura

Dear List, I'm planning to upgrade from kamailio 4.3.7 to 4.4.x I am looking on www to see if there are any upgrade or modification to the kamailio db schema but i havenpt found any good details. Anyone here that must did the upgrade and can give me some information or url to check about it ? Thanks for any reply. Laura

7 years, 4 months

Detect network range in UDP traffic

by Emanuele Gambaro

Hi to all I have the following scenario (Kamailio 5 and rtpproxy) *) Bob on a wifi network calls Alice *) During the call, Bob losts his wifi network (and so also changes his IP address) *) The RTP flow is interrupted With asterisk, disabling “Strict RTP” option, the flow does not interrupts: Asterisk detect the new IP address and sends the audio packets to the new IP address. It is possible to setup this behavior also with Kamailio? -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

7 years, 4 months

Question about sip call setup : three-way handshake

by Abdoul Osséni

Hello, During three-way handshake (SIP call setup - INVITE-200 OK - ACK) and before generating or sending the final 200 OK call, is there a way to confirm the caller is still available (no network issue)? I want to detect broken connections (ex. the uac uses 3G network) before generating/sending/forwarding the final 200 OK. Thanks. Regards Abdoul

7 years, 4 months

How to forward DTMF (in RTP package) to conference within SEMS

by Nguyen Tran Nhan

I am using Kamailio + Sems + Rtpproxy for conference call. Once user join to conference, if user send dtmf, sems is detected but it does not forward DTMF (in RTP) to all user in conference. I think rtprelay_dtmf_detection should be configured? Do you have any idea to help? Thanks, Nhan

7 years, 4 months

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sr-users June 2017