[OpenSIPS-Users] Anybody is interested to develop this scenario using OpenSIPS?
Hello All, If anybody is interested to develop this below written scenario using OpenSIPs, please let me know. Global SIP Users ---> OpenSIPS ---> Asterisk media server1 --- | |-- VoIP provider for PSTN calls | | ---> Asterisk media server2 --- *Assumptions:* OpenSIPS public IP address (eth0): 104.131.65.66 OpenSIPS private IP address (eth1): 10.10.10.1 Asterisk media server1 private IP address (eth1): 10.10.10.2 Asterisk media server2 private IP address (eth1): 10.10.10.3 MySQL DB server private IP address (eth1): 10.10.10.4 VoIP provider public ip address: 123.456.789.111 1) All servers are hosted in Digital ocean and in private network 2) All SIP users, voice mail users, dial rules will be stores in MySQL database 3) I must give OpenSIPS proxy server public ip address in to my VoIP provider. My provider will allow incoming/out going traffic through this IP address only. But, call should go through our media servers only. Because, dial rules will be stored in MySQL database. 4) SIP users will connects to openSIPs proxy server from globally 5) I will provide you the whole environment with the installed OpenSIPs (Ubuntu), installed Asterisk (CentOS) servers, and installed MySQL database tables. 6) I will configure Asterisk in real time and data base. *Task:* You need to provide me OpenSIPs working configuration file to fulfill the below needs for the above environment: 1) Nat traversal 2) SIP registrations through proxy (As I said, we store all sip users details in MySQL database table) 3) Load balancing (We will give two media servers) with fail over 4) PSTN inbound/outbound calling through media servers by using MySQL data base tables (Because, we store users dial rules in db table). But, we give our Proxy server ip address to our VoIP provider for authentication purpose. Please do not reply me, if you are a learner. Only experienced professional with OpenSIPS are welcome. Thank you. Chandra. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Lawful solution with opensips/rtp proxy
Hi. any idea or pointer to some blog regarding lawful intercept solution. On Apr 13, 2015 5:30 PM, "Abdul Basit" wrote: > Hi, > > I am searching for LI solution that can integrate with opensips and rtp > proxy for LEA. > > Can anyone suggest for a model for HI1, HI2 and HI3 interfaces for > monitoring and interception according to ETSI standards? > > Is there any ready made product available in market. > > -- > regards, > > abdul basit > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] not trigger on_failure_route[0] when use mi interface
Hi： I make a mistake. I use tcp with async that fix problem. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/not-trigger-on-failure-route-0-when-use-mi-interface-tp7596480p7596574.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP/2.0 477 Send failed (477/TM) - Route - SOLVED
Thanks a lot Vlad!!!, I guess the TCP/UDP was the problem. I followed your advise and now it works. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-2-0-477-Send-failed-477-TM-Route-tp7595929p7596568.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] not trigger on_failure_route[0] when use mi interface
Hello,
Not sure I understand what is taking more than 10 minutes ?
Why are you trying to change the maximum number of milliseconds of
OpenSIPS trying to retransmit a message ?
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 16.04.2015 07:04, chow wrote:
Hi：
It work for me with use $du that redirect request to Main route.
have another problem about retransmission
eg：
I followed the order sent 50 messages. 40 of message will be fast
store into db.
the remaining message (40-50) store into db that Spend more than 10
minute.
I modify:
modparam("tm", "fr_timer", 5)
modparam("tm", "fr_inv_timer", 30)
modparam("tm", "T2_timer", 1000)
but it always spend long time to do it.
why T2_timer no work with a loss value?
have some timer like TIMER B or TIME F for non-invite message?
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/not-trigger-on-failure-route-0-when-use-mi-interface-tp7596480p7596551.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP/2.0 477 Send failed (477/TM) - Route
Hello, Try to have an OpenSIPS localhost UDP listener, and do force_send_socket(udp:127.0.0.1:5060) before attempting to relay to 127.0.0.1:9 In your case it fails since you're trying to relay via TCP, which fails since nobody listens over there - you won't have those issues with UDP. Best Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 15.04.2015 23:04, leo wrote: Enabling more detailed logs i think the error would be "$du = "sip:127.0.0.1:9";" Wouldn't be $ru instead of $du? And i receive ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR [server=127.0.0.1:9] (111) Connection refused Thanks one more time, Leo -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-2-0-477-Send-failed-477-TM-Route-tp7595929p7596548.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] command get_statistics is not available
Hello, The accepted syntax’s are either opensipsctl fifo get_statistics all in order to fetch all statistics, or opensipsctl fifo get_statistics core: in order to fetch all statistics form a certain module, or opensipsctl fifo get_statistics core rcv_requests in order to fetch a single statistic. Best Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 15.04.2015 17:42, Dale Harris wrote: Hi, I have an OpenSIPS system that isn't allowing me to run opensipctl fifo get_statistics. Log show an error of: opensips[51324]: ERROR:mi_fifo:mi_fifo_server: command get_statistics is not available Any suggestions on how to fix this? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] regex in if statement
Thanks Aron! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
