[SR-Users] topos redis -> failed to store

2018-01-05 Thread Abdoul Osséni
Hello list, I activated topos and topos_redis backend. Now, I have the following errors: Jan 6 07:34:11 sd-110402 /usr/local/sbin/kamailio[30288]: {1 102 NOTIFY 645f4b58537f9df53b9ce65f4937d652@163.172.83.169:5064} ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Jan 6

[SR-Users] Kamailio fail to start with db_mongodb and ndb_mongodb modules

2018-01-05 Thread Abdul Basit
Hi Daniel & K-Team, I setup kamailio 5.0.5 from git on vanilla Debian GNU/Linux 8.3 (jessie) # kamailio -v version: kamailio 5.0.5 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC,

Re: [SR-Users] TLS cipher suites

2018-01-05 Thread Steve
Hello, Thank you both for your responses to my query about TLS cipher suites supported by Kamailio 4.3.4. When I used a self-signed certificate generated from an RSA key, the server selected the RSA-AES256-GCM-SHA384 cipher suite for the connection. When I used a self-signed certificate generated

Re: [SR-Users] Kami as NAT traversal + FS as media

2018-01-05 Thread Yu Boot
Thanks for an answer? but what I want to do (already done :)) is totally opposite to your solution. The main thing I wanted, so I don't need to open 5060/UDP on FreeSwitch for (entire world), BUT at the same time АЫ should work with RTP traffic directly, no matter is it NATed UAC or not.

Re: [SR-Users] kamailio-5.0.5 active/passive pcs/cluster with rtpengine - how to fix calls after failover

2018-01-05 Thread Karsten Horsmann
Hello Daniel, hello List, for a better understanding i attached the "pcs status" and "pcs config" output - to show up how things are located and colocated. It maybe also nice for other people that try to do the same. The Services rtpengine_service and kamailio_service running always on the same

Re: [SR-Users] WebRTC proxy for legacy systems

2018-01-05 Thread Karsten Horsmann
Hello, you also be aware of longer callsetups to webrtc clients (13 - 38 seconds) if there are many ICE candidates on the client (like different networking devices). There maybe also fixes in the js SIP libraries for that (I don't touch this area). Am 05.01.2018 9:04 vorm. schrieb

Re: [SR-Users] kamailio-5.0.5 active/passive pcs/cluster with rtpengine - how to fix calls after failover

2018-01-05 Thread Karsten Horsmann
Hi Daniel, Yes, they are. At this point I using only one redis key space for both rtpengines. I just fire it up on the backup machine so it reads the RTP sessions from redis. Both rtpengines had the same configuration. Only one is active. But I found the nice redis key space separated and

Re: [SR-Users] Cnxcc doesn't update amount values in redis

2018-01-05 Thread Daniel-Constantin Mierla
Hello, not a cnxcc user here ... anyhow, few questions to see where this can lead:  - is only the update to redis that doesn't work? are calls terminated/charged as expected?  - when the call is ended, are there any updates? Cheers, Daniel On 20.12.17 11:59, Donat Zenichev wrote: > Hi

Re: [SR-Users] NOTIFY generation for REFER

2018-01-05 Thread Daniel-Constantin Mierla
Hello, On 22.12.17 15:46, Konstantin Tumalevich wrote: > I try to build custom attended transfer logick with kamailio and asterisk. > My backend can do all work (include interconnect between asterisk > servers), so I need to fully handle REFER in kamailio. But I can't > send back NOTIFY message

Re: [SR-Users] BUG: tls [tls_server.c:1229]: tls_read_f(): SSL_ERROR_WANT_READ but data still in the rbio (0x7ffd354fcd30, 8 bytes at 461)

2018-01-05 Thread Daniel-Constantin Mierla
Hello, how often these message appear? Do you have server to server tls connections, or only client (phone) to server? Cheers, Daniel On 23.12.17 16:09, Abdoul Osséni wrote: > Hello Dear, > > I use Kamailio 5.1.0 on Linux server: > > root@sip-africallshop-com:~# kamailio -V > version: kamailio

Re: [SR-Users] WebRTC proxy for legacy systems

2018-01-05 Thread Daniel-Constantin Mierla
Hello, another thing that one should be aware of is that in webrtc/websocket some sip headers (e.g., via, contact) use a random string instead of ip addresses and many old devices will throw parsing error. jssip (and maybe other js sip stacks) has an option to enable using a private ip address