Re: [OpenSIPS-Users] Problem with uac_replace_from
Hello, In the current implementation, the display name is current not saved, and obviously, not restored. Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 02/17/2012 11:26 AM, Steven Lam, KeenSystems B.V. wrote: Hi, Thank you for your answer! You are right, this is what I discovered. I think this behavior is wrong, if the function has the feature to overrule the display it should behave the same as any other feature the function has. If there are some good reasons why this is not the case it should be mentioned in the documentation is think. Do you know IF there is a good reason for not storing the overruled display? Steven -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: vrijdag 17 februari 2012 9:26 To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Problem with uac_replace_from Hi, Steven! The dialog module only stores the TO and FROM URIs, not the display names. Also, the uac_replace_from function keeps track only of the URIs. Therefore, the display names are ignored and they will not be automatically restored on any sequential requests. Regards, -- Răzvan Crainea OpenSIPS Developer http://www.opensips-solutions.com On 02/16/2012 07:08 PM, Steven Lam, KeenSystems B.V. wrote: Hi, Replying to myself... Also looking at the opensipsctl fifo dlg_list_ctx output I can see only the changed uri is stored in the dialog. Should the display not be stored? or am I missing something here? Steven -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Steven Lam, KeenSystems B.V. Sent: donderdag 16 februari 2012 17:49 To: users@lists.opensips.org Subject: [OpenSIPS-Users] Problem with uac_replace_from Hi list! When playing with uac_replace_from to replace both display and uri like this: uac_replace_from(anonymous,sip:anonymous@anonymous.invalid); I found that on subsequential (ACK) messages only the uri was changed again and display had the original value. Also when using uac_replace_from to replace only uri and remove display like this: uac_replace_from(,sip:anonymous@anonymous.invalid); On subsequential (ACK) messages the uri is changed as expected but display had the original value. To me this looks wrong. Steven ___ 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 ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Error starting opensips 1.8.0-1-87541
Hi list, I've just install from Debian repository version 1.8.0-1-87541 (trunk) and when I try to start it fails with the next output : Not starting opensips: invalid configuration file! -e Feb 29 12:38:01 [16974] WARNING:core:warn: warning in config file, line 50, column 13-16: tls support not compiled in Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 158, column 20-21: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 161, column 27-28: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 175, column 21-22: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 193, column 18-19: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 202, column 21-22: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 223, column 20-21: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 229, column 19-20: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 233, column 18-19: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 240, column 19-20: unknown command append_hf, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 247, column 18-19: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 253, column 18-19: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 296, column 18-19: unknown command is_method, missing loadmodule? Feb 29 12:38:01 [16974] ERROR:core:main: bad config file (12 errors) Regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with uac_replace_from
Hi Steven, The reason for not storing/restoring the display name is simple: this information has no value in regards to the SIP dialog (it does not affect or interfere with the dialog state, as the FROM URI does) and it is used only in the initial invites - to be displayed when receiving a call ; in the sequential requests, it is not used at all. A good practice (or at least what we do on our systems) is to delete the display name in the sequential requests if the display in initial invite was changed (more for privacy purposes). Regards, Bogdan On 02/17/2012 11:26 AM, Steven Lam, KeenSystems B.V. wrote: I think this behavior is wrong, if the function has the feature to overrule the display it should behave the same as any other feature the function has. If there are some good reasons why this is not the case it should be mentioned in the documentation is think. -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPs-CP -Pro blems with ERROR:mi_fifo:mi_fifo_server:
Hi there. I've just installed OpenSIPs + OpenSIP-CP, but everytime I try to Apply Changes to Server or Refresh Dialog List, I get the following messages. Feb 28 16:27:46 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dlg_list is not available Feb 28 16:27:51 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dp_reload is not available Feb 28 16:27:54 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_list is not available Feb 28 16:27:55 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_reload is not available Feb 28 16:28:00 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command domain_reload is not available Feb 28 16:28:05 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dr_reload is not available Feb 28 16:28:13 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command lb_reload is not available Feb 28 16:28:25 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command nh_reload_rtpp is not available Feb 28 16:28:27 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command sip_trace is not available Any help is welcome. -- Atenciosamente, ALEXANDRE KELLER http://twitter.com/alexandrekeller http://www.facebook.com/alexandre.keller.BR Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho. P Antes de imprimir pense em seu compromisso com o Meio Ambiente. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs-CP -Pro blems with ERROR:mi_fifo:mi_fifo_server:
Hello, In your OpenSIPS script, do you have the dialog, dialplan, dispatcher, domain or drouting modules loaded ? Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 02/29/2012 02:21 PM, Alexandre Keller wrote: Hi there. I've just installed OpenSIPs + OpenSIP-CP, but everytime I try to Apply Changes to Server or Refresh Dialog List, I get the following messages. Feb 28 16:27:46 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dlg_list is not available Feb 28 16:27:51 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dp_reload is not available Feb 28 16:27:54 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_list is not available Feb 28 16:27:55 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_reload is not available Feb 28 16:28:00 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command domain_reload is not available Feb 28 16:28:05 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dr_reload is not available Feb 28 16:28:13 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command lb_reload is not available Feb 28 16:28:25 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command nh_reload_rtpp is not available Feb 28 16:28:27 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command sip_trace is not available Any help is welcome. -- Atenciosamente, ALEXANDRE KELLER http://twitter.com/alexandrekeller http://www.facebook.com/alexandre.keller.BR Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho. * * **P Antes de imprimir pense em seu compromisso com o Meio Ambiente.** ___ 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] Using dispatcher and t_replicate()
Hello, Razvan I have try to use psevdo variable in opensips (1.7.2) script like this: t_replicate($du,0x4); But i catch a bad arguments error: CRITICAL:core:yyerror: parse error in config file, line 197, column 26-27: bad arguments for command t_replicate Is this functional t_replicate can also receive a pseudo-variable as argument. ported to 1.7.2 opensips ? Regards, Alex uiptel.com -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Using-dispatcher-and-t-replicate-tp5592900p7329158.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] Error starting opensips 1.8.0-1-87541
hi, With that it works fine Regards On 29 February 2012 12:58, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Seattle, The warning lets you know that the debs were not compiled with TLS support - and from script, I guess, you are trying to do some TLS stuff. The rest of the errors (about not finding certain functions in script) are related to some recent change on trunk (keep in mind, 1.8.0 is trunk unstable) where the textops module was split into textops and sipmsgops module (following the addition of more sip message oriented functionalities). See: http://lists.opensips.org/pipermail/news/2012-February/000175.html So, just load the sipmsgops module also. Regards, Bogdan On 02/29/2012 01:37 PM, 113 Seattle wrote: Hi list, I've just install from Debian repository version 1.8.0-1-87541 (trunk) and when I try to start it fails with the next output : Not starting opensips: invalid configuration file! -e Feb 29 12:38:01 [16974] WARNING:core:warn: warning in config file, line 50, column 13-16: tls support not compiled in Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 158, column 20-21: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 161, column 27-28: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 175, column 21-22: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 193, column 18-19: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 202, column 21-22: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 223, column 20-21: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 229, column 19-20: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 233, column 18-19: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 240, column 19-20: unknown commandappend_hf, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 247, column 18-19: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 253, column 18-19: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] CRITICAL:core:yyerror: parse error in config file, line 296, column 18-19: unknown commandis_method, missing loadmodule? Feb 29 12:38:01 [16974] ERROR:core:main: bad config file (12 errors) Regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Using dispatcher and t_replicate()
Yes! Răzvan, Thank you. Alex uiptel.com -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Using-dispatcher-and-t-replicate-tp5592900p7329418.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] OpenSIPs-CP -Pro blems with ERROR:mi_fifo:mi_fifo_server:
I'm sorry for bothering you again. I've trying to add some rule to my DialPlan through OpenSIPs-CP, but, it keeps me showing the following error: Failed to issue query, error message : MDB2 Error: syntax error Any ideas? Thanks again in advance. -- Atenciosamente, ALEXANDRE KELLER http://twitter.com/alexandrekeller http://www.facebook.com/alexandre.keller.BR Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho. P Antes de imprimir pense em seu compromisso com o Meio Ambiente. On 29/02/2012, at 09:43, Vlad Paiu wrote: Hello, In your OpenSIPS script, do you have the dialog, dialplan, dispatcher, domain or drouting modules loaded ? Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 02/29/2012 02:21 PM, Alexandre Keller wrote: Hi there. I've just installed OpenSIPs + OpenSIP-CP, but everytime I try to Apply Changes to Server or Refresh Dialog List, I get the following messages. Feb 28 16:27:46 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dlg_list is not available Feb 28 16:27:51 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dp_reload is not available Feb 28 16:27:54 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_list is not available Feb 28 16:27:55 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_reload is not available Feb 28 16:28:00 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command domain_reload is not available Feb 28 16:28:05 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dr_reload is not available Feb 28 16:28:13 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command lb_reload is not available Feb 28 16:28:25 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command nh_reload_rtpp is not available Feb 28 16:28:27 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command sip_trace is not available Any help is welcome. -- Atenciosamente, ALEXANDRE KELLER http://twitter.com/alexandrekeller http://www.facebook.com/alexandre.keller.BR Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho. P Antes de imprimir pense em seu compromisso com o Meio Ambiente. ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs-CP -Pro blems with ERROR:mi_fifo:mi_fifo_server:
Hi, This is a strange error, considering the fact that I am guessing you already have DB connectivity for CP. Can you paste the rule that you are trying to add ? Maybe we can replicate the error. Regards, Alex On 02/29/2012 04:57 PM, Alexandre Keller wrote: I'm sorry for bothering you again. I've trying to add some rule to my DialPlan through OpenSIPs-CP, but, it keeps me showing the following error: Failed to issue query, error message : MDB2 Error: syntax error Any ideas? Thanks again in advance. -- Atenciosamente, ALEXANDRE KELLER http://twitter.com/alexandrekeller http://www.facebook.com/alexandre.keller.BR Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho. * * **P Antes de imprimir pense em seu compromisso com o Meio Ambiente.** On 29/02/2012, at 09:43, Vlad Paiu wrote: Hello, In your OpenSIPS script, do you have the dialog, dialplan, dispatcher, domain or drouting modules loaded ? Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 02/29/2012 02:21 PM, Alexandre Keller wrote: Hi there. I've just installed OpenSIPs + OpenSIP-CP, but everytime I try to Apply Changes to Server or Refresh Dialog List, I get the following messages. Feb 28 16:27:46 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dlg_list is not available Feb 28 16:27:51 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dp_reload is not available Feb 28 16:27:54 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_list is not available Feb 28 16:27:55 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_reload is not available Feb 28 16:28:00 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command domain_reload is not available Feb 28 16:28:05 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dr_reload is not available Feb 28 16:28:13 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command lb_reload is not available Feb 28 16:28:25 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command nh_reload_rtpp is not available Feb 28 16:28:27 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command sip_trace is not available Any help is welcome. -- Atenciosamente, ALEXANDRE KELLER http://twitter.com/alexandrekeller http://www.facebook.com/alexandre.keller.BR Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho. * * **P Antes de imprimir pense em seu compromisso com o Meio Ambiente.** ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto: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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] help configuring Opensips as proxy
Hi, I am trying to get familiar with Opensips (1.7.1) so I set up two boxes with SIPp, OpenSIPS, and Asterisk: box 1 (ip: 192.168.1.57) running SIPp and Opensips box 2 (ip: 192.168.1.121) running Asterisk I am trying to get OpenSIPS to run as a proxy between SIPp and Asterisk, but this is the problem I have so right now: SIPp OpenSIPS Asterisk ||| | INVITE || ||| || INVITE | ||| ||| || 100 Trying | ||| | 100 Trying | | ||| ||| || 200 OK| ||| ||| |200 OK || ||| ||| | ACK || ||| ||| ||| All the messages look up until SIPp sends an ACK in response to the 200 OK, but instead of sending an ACK to Asterisk, Opensips seems to be sending the ACK back to itself, and goes into a loop. These are the logs from tcpdump for the loopback interface on box 1: 12:11:46.574820 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 567) 192.168.1.57.sip-tls 192.168.1.57.sip: SIP, length: 539 INVITE sip:0119054741990@192.168.1.57:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0 From: sipp sip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sut sip:0119054741990@192.168.1.57:5060 Call-ID: 1-18363@192.168.1.57 CSeq: 1 INVITE Contact: sip:sipp@192.168.1.57:5061 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 133 v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.57 s=- c=IN IP4 192.168.1.57 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/800[|sip] 12:11:46.575332 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 338) 192.168.1.57.sip 192.168.1.57.sip-tls: SIP, length: 310 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0 From: sipp sip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sut sip:0119054741990@192.168.1.57:5060 Call-ID: 1-18363@192.168.1.57 CSeq: 1 INVITE Server: OpenSIPS (1.7.1-notls (x86_64/linux)) Content-Length: 0 12:11:46.577090 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 765) 192.168.1.57.sip 192.168.1.57.sip-tls: SIP, length: 737 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0 Record-Route: sip:192.168.1.57;lr From: sipp sip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sut sip:0119054741990@192.168.1.57:5060;tag=as1642d8ff Call-ID: 1-18363@192.168.1.57 CSeq: 1 INVITE Server: Asterisk PBX 1.8.9.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:0119054741990@192.168.1.121:5060 Content-Type: application/sdp Content-Length: 209 v=0 o=root 1639240398 1639240398 IN IP4 192.168.1.121 s=Asterisk PBX 1.8.9.3 c=IN IP4 192.168.1.121 t=0 0 m=audio 10014 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 12:11:46.577303 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 408) 192.168.1.57.sip-tls 192.168.1.57.sip: SIP, length: 380 ACK sip:0119054741990@192.168.1.57:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-5 From: sipp sip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sut sip:0119054741990@192.168.1.57:5060;tag=as1642d8ff Call-ID: 1-18363@192.168.1.57 CSeq: 1 ACK Contact: sip:sipp@192.168.1.57:5061 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 12:11:46.577613 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 469) 192.168.1.57.sip 192.168.1.57.sip: SIP, length: 441 ACK sip:0119054741990@192.168.1.57:5060 SIP/2.0 Via: SIP/2.0/UDP
Re: [OpenSIPS-Users] help configuring Opensips as proxy
Hi there, My guess is the fault is in the sipp script - the ACK is not properly generated : instead of using the route set information from the 200 OK (the contact and RR URIs), it is simply sent with the same RURI as the INVITE - this is of course bogus. If you want, I can send you an working SIPP UAC file. Regards, Bogdan On 02/29/2012 08:02 PM, dyatsin wrote: Hi, I am trying to get familiar with Opensips (1.7.1) so I set up two boxes with SIPp, OpenSIPS, and Asterisk: box 1 (ip: 192.168.1.57) running SIPp and Opensips box 2 (ip: 192.168.1.121) running Asterisk I am trying to get OpenSIPS to run as a proxy between SIPp and Asterisk, but this is the problem I have so right now: SIPp OpenSIPS Asterisk ||| | INVITE || ||| ||INVITE | | || ||| ||100 Trying | ||| | 100 Trying | | ||| ||| || 200 OK| ||| | || |200 OK || ||| ||| | ACK || ||| ||| ||| All the messages look up until SIPp sends an ACK in response to the 200 OK, but instead of sending an ACK to Asterisk, Opensips seems to be sending the ACK back to itself, and goes into a loop. These are the logs from tcpdump for the loopback interface on box 1: 12:11:46.574820 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 567) 192.168.1.57.sip-tls 192.168.1.57.sip: SIP, length: 539 INVITE sip:0119054741990@192.168.1.57:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0 From: sippsip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sutsip:0119054741990@192.168.1.57:5060 Call-ID: 1-18363@192.168.1.57 CSeq: 1 INVITE Contact: sip:sipp@192.168.1.57:5061 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 133 v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.57 s=- c=IN IP4 192.168.1.57 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/800[|sip] 12:11:46.575332 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 338) 192.168.1.57.sip 192.168.1.57.sip-tls: SIP, length: 310 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0 From: sippsip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sutsip:0119054741990@192.168.1.57:5060 Call-ID: 1-18363@192.168.1.57 CSeq: 1 INVITE Server: OpenSIPS (1.7.1-notls (x86_64/linux)) Content-Length: 0 12:11:46.577090 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 765) 192.168.1.57.sip 192.168.1.57.sip-tls: SIP, length: 737 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0 Record-Route:sip:192.168.1.57;lr From: sippsip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sutsip:0119054741990@192.168.1.57:5060;tag=as1642d8ff Call-ID: 1-18363@192.168.1.57 CSeq: 1 INVITE Server: Asterisk PBX 1.8.9.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact:sip:0119054741990@192.168.1.121:5060 Content-Type: application/sdp Content-Length: 209 v=0 o=root 1639240398 1639240398 IN IP4 192.168.1.121 s=Asterisk PBX 1.8.9.3 c=IN IP4 192.168.1.121 t=0 0 m=audio 10014 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 12:11:46.577303 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 408) 192.168.1.57.sip-tls 192.168.1.57.sip: SIP, length: 380 ACK sip:0119054741990@192.168.1.57:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-5 From: sippsip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sutsip:0119054741990@192.168.1.57:5060;tag=as1642d8ff Call-ID:
Re: [OpenSIPS-Users] help configuring Opensips as proxy
Hi David, Here it is. Regards, Bogdan On 02/29/2012 08:23 PM, David Yat Sin wrote: Hi Bogdan, Thanks for looking into this. Could you send me your SIPP UAC file. Regards, David On 12-02-29 1:20 PM, Bogdan-Andrei Iancubog...@opensips.org wrote: Hi there, My guess is the fault is in the sipp script - the ACK is not properly generated : instead of using the route set information from the 200 OK (the contact and RR URIs), it is simply sent with the same RURI as the INVITE - this is of course bogus. If you want, I can send you an working SIPP UAC file. Regards, Bogdan On 02/29/2012 08:02 PM, dyatsin wrote: Hi, I am trying to get familiar with Opensips (1.7.1) so I set up two boxes with SIPp, OpenSIPS, and Asterisk: box 1 (ip: 192.168.1.57) running SIPp and Opensips box 2 (ip: 192.168.1.121) running Asterisk I am trying to get OpenSIPS to run as a proxy between SIPp and Asterisk, but this is the problem I have so right now: SIPp OpenSIPS Asterisk ||| | INVITE || ||| || INVITE | | || ||| || 100 Trying | ||| | 100 Trying | | ||| ||| || 200 OK| ||| | || |200 OK || ||| ||| |ACK || ||| ||| ||| All the messages look up until SIPp sends an ACK in response to the 200 OK, but instead of sending an ACK to Asterisk, Opensips seems to be sending the ACK back to itself, and goes into a loop. These are the logs from tcpdump for the loopback interface on box 1: 12:11:46.574820 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 567) 192.168.1.57.sip-tls 192.168.1.57.sip: SIP, length: 539 INVITE sip:0119054741990@192.168.1.57:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0 From: sippsip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sutsip:0119054741990@192.168.1.57:5060 Call-ID: 1-18363@192.168.1.57 CSeq: 1 INVITE Contact: sip:sipp@192.168.1.57:5061 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 133 v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.57 s=- c=IN IP4 192.168.1.57 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/800[|sip] 12:11:46.575332 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 338) 192.168.1.57.sip 192.168.1.57.sip-tls: SIP, length: 310 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0 From: sippsip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sutsip:0119054741990@192.168.1.57:5060 Call-ID: 1-18363@192.168.1.57 CSeq: 1 INVITE Server: OpenSIPS (1.7.1-notls (x86_64/linux)) Content-Length: 0 12:11:46.577090 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 765) 192.168.1.57.sip 192.168.1.57.sip-tls: SIP, length: 737 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.57:5061;branch=z9hG4bK-18363-1-0 Record-Route:sip:192.168.1.57;lr From: sippsip:sipp@192.168.1.57:5061;tag=18363SIPpTag001 To: sutsip:0119054741990@192.168.1.57:5060;tag=as1642d8ff Call-ID: 1-18363@192.168.1.57 CSeq: 1 INVITE Server: Asterisk PBX 1.8.9.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact:sip:0119054741990@192.168.1.121:5060 Content-Type: application/sdp Content-Length: 209 v=0 o=root 1639240398 1639240398 IN IP4 192.168.1.121 s=Asterisk PBX 1.8.9.3 c=IN IP4 192.168.1.121 t=0 0 m=audio 10014 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 12:11:46.577303 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 408)
Re: [OpenSIPS-Users] OpenSIPs-CP -Pro blems with ERROR:mi_fifo:mi_fifo_server:
I might be doing something wrong. I'm kind of new on OpenSIPs, I've only worked with Asterisk for the past 6 years. Here is my Rule, it's a testing rule. Dialplan ID: 1 Rule Priority: 1 Matching Operator: REGEX Matching Regular Expression: 123* Matching String Length: * Substitution Regular Expression: BLANK Replacement Expression: BLANK Attributes: NONE When a push ADD button: Failed to issue query, error message : MDB2 Error: syntax error Another strange thing is when I add a Gateway on PERMISSIONS page. I fill the netmask field with 255.255.255.0, but it does not save the informations. I tried filling with 24 (bits), and it saved just fine. Is it correct? When I APPLY CHANGES TO SERVER the following message appears on syslogd: Feb 29 14:33:25 vm-opensips /usr/sbin/opensips[4281]: WARNING:core:mk_net: invalid network address/netmask combination fixed... As I said before, I must be doing something wrong. Thanks again. -- Atenciosamente, ALEXANDRE KELLER http://twitter.com/alexandrekeller http://www.facebook.com/alexandre.keller.BR Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho. P Antes de imprimir pense em seu compromisso com o Meio Ambiente. On 29/02/2012, at 14:09, Alex Ionescu wrote: Hi, This is a strange error, considering the fact that I am guessing you already have DB connectivity for CP. Can you paste the rule that you are trying to add ? Maybe we can replicate the error. Regards, Alex On 02/29/2012 04:57 PM, Alexandre Keller wrote: I'm sorry for bothering you again. I've trying to add some rule to my DialPlan through OpenSIPs-CP, but, it keeps me showing the following error: Failed to issue query, error message : MDB2 Error: syntax error Any ideas? Thanks again in advance. -- Atenciosamente, ALEXANDRE KELLER http://twitter.com/alexandrekeller http://www.facebook.com/alexandre.keller.BR Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho. P Antes de imprimir pense em seu compromisso com o Meio Ambiente. On 29/02/2012, at 09:43, Vlad Paiu wrote: Hello, In your OpenSIPS script, do you have the dialog, dialplan, dispatcher, domain or drouting modules loaded ? Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 02/29/2012 02:21 PM, Alexandre Keller wrote: Hi there. I've just installed OpenSIPs + OpenSIP-CP, but everytime I try to Apply Changes to Server or Refresh Dialog List, I get the following messages. Feb 28 16:27:46 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dlg_list is not available Feb 28 16:27:51 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dp_reload is not available Feb 28 16:27:54 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_list is not available Feb 28 16:27:55 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command ds_reload is not available Feb 28 16:28:00 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command domain_reload is not available Feb 28 16:28:05 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command dr_reload is not available Feb 28 16:28:13 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command lb_reload is not available Feb 28 16:28:25 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command nh_reload_rtpp is not available Feb 28 16:28:27 vm-opensips /usr/sbin/opensips[5079]: ERROR:mi_fifo:mi_fifo_server: command sip_trace is not available Any help is welcome. -- Atenciosamente, ALEXANDRE KELLER http://twitter.com/alexandrekeller http://www.facebook.com/alexandre.keller.BR Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho. P Antes de imprimir pense em seu compromisso com o Meio Ambiente. ___ 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 ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Subject: ClueCon 2012 - Call For Speaking Proposals
Greetings! The ClueCon team is gearing up for this summer's event, which will be held August 7-9, 2012 at the Wyndham in downtown Chicago. We want to hear your proposals for presentations to be given at ClueCon '12. Here's what we're looking for: - Discussions on technology or software that relate directly or indirectly to open source telephony - Working title or very brief description of the talk - A two or three sentence abstract giving more details - Speaker's name, company, and biographical information - All presentations should be 30 minutes in length, including question and answer time - ClueCon sponsors who wish to have a presentation always get first priority when it comes to scheduling Please send all proposals to market...@cluecon.com. If you spoke at ClueCon last year and your bio has not changed then please make note of this fact. We look forward to hearing from you! The ClueCon Team http://www.cluecon.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Can you drop a 487 Request Terminated?
When I setup a call then cancel the call I am getting a 487 from my gateway
that is relayed to the client.
I don't wish to show the 487 to the client. Is it possible to drop the reply
for the 487?
I am hoping someone could let me know if this is possible.
I am running opensips 1.7 Here is a copy of my config as well.
debug=4
log_stderror=no
log_facility=LOG_LOCAL0
fork=yes
children=4
disable_tcp=yes
auto_aliases=yes
sip_warning=yes
listen=udp:10.8.1.139:5060
group=nobody
user=nobody
server_header=ZZZ
server_signature = off
user_agent_header=User-Agent: ZZZ
### Modules Section
#set module path
mpath=/usr/local/lib64/opensips/modules/
loadmodule db_text.so
loadmodule signaling.so
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule textops.so
loadmodule mi_fifo.so
loadmodule uri.so
loadmodule domain.so
loadmodule permissions.so
loadmodule userblacklist.so
loadmodule dialog.so
modparam(domain|userblacklist|dialog|permissions,
db_url,text:///zxa/server/opensipsDNC)
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
modparam(rr, append_fromtag, 0)
modparam(uri, use_uri_table, 0)
modparam(domain, db_mode, 1) # Use caching
### Routing Logic
# main request routing logic
route{
xlog(=== TOP ===);
xlog( tu= $tu | fu= $fu | od= $od | ReceivedINT: $Ri | SourceIP: $si
%%%);
if ( $si == 10.8.1.139) ( $Ri == 10.8.1.139)
{
exit;
}
if (!check_address(1,$si,$sp,$proto))
{
xlog(=== ACCESS FAILED ===);
xlog(=== $si| $sp | $proto ===);
sl_send_reply(403,Forbidden);
exit;
}
if (!mf_process_maxfwd_header(10))
{
xlog(=== TOO MANY HOPS ===);
sl_send_reply(483,Too Many Hops);
exit;
}
#CANCEL processing
if (is_method(CANCEL))
{
xlog(=== CANCEL 76 ===);
if (t_check_trans())
t_relay();
exit;
}
if (is_method(PUBLISH))
{
sl_send_reply(503, Service Unavailable);
exit;
}
if (has_totag())
{
xlog(=== HAS TO TAG ===);
if (loose_route())
{
xlog(=== HAS LOOSE ===);
if (is_method(BYE))
{
xlog(=== LOOSE BYE ===);
route(3);
exit;
} else if (is_method(INVITE)) {
xlog(=== RE-INVITE LOOSE ===);
record_route();
}
xlog(In loouse going to Route 1);
route(1);
} else {
if ( is_method(ACK) )
{
xlog(=== LOOSE ELSE ACK ===);
if ( t_check_trans() )
{
t_relay();
exit;
} else {
xlog(=== LOOSE Discard ACK ===);
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
t_check_trans();
if (loose_route())
{
xlog(L_ERR,Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
{
sl_send_reply(403,Preload Route denied);
exit;
}
}
# record routing
if (!is_method(REGISTER|MESSAGE))
{
record_route();
}
if (!is_uri_host_local())
{
xlog(Not local so lets just see what happens $rd);
route(1);
}
if ($rU==NULL)
{
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
if (is_method(INVITE) (!has_totag()) )
{
xlog(|| NEW CALL ||);
}
route(2);
send_reply(420, Invalid Extension);
exit;
}
route[1] {
# RTP Proxy handling ---#
xlog(=== ROUTE 1 ===);
rewritehostport(10.8.1.44:5060);
if (is_method(INVITE)) {
t_on_reply(1);
t_on_failure(1);
}
if (!t_relay())
{
sl_reply_error();
}
exit;
}
route[2] {
xlog(=== ROUTE 2 ===);
if (!check_blacklist(userblacklist))
{
xlog('~~~ BLACKLISTED DID Forbidden ~~~');
sl_send_reply(403, DID Forbidden);
exit;
}
route(1);
}
route[3] {
xlog(=== ROUTE 3 ===);
t_on_reply(1);
t_on_failure(1);
t_relay(udp:10.8.1.44:5060);
}
branch_route[1] {
xlog(new branch at $ru\n);
}
onreply_route[1] {
xlog(=== ON REPLY ROUTE 2 rs= $rs | fu= $fu | si= $si | Ri= $Ri ===);
if (t_check_status(487))
{
xlog(487 at reply route);
t_cancel_branch();
drop;
}
}
failure_route[1] {
xlog(=== FAIL ROUTE ===);
xlog(=== FAIL ROUTE 2 fu= $fu | od=
Re: [OpenSIPS-Users] Packet Loss and its Solution
Hi Sammy, Sure I will paste the stats here but will you please tell me some important steps to be performed on software side. Here I am talking about the OS, kernel upgrade advanced system administration level stuff that can reduce the packet loss immediately. Through the final conclusions I have become to know that its not the OpenSIPS problem but it regards with physical, software network upgrades. Regards, Faisal Rehman From: Sammy Govind govoi...@gmail.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Tuesday, February 28, 2012 10:38 AM Subject: Re: [OpenSIPS-Users] Packet Loss and its Solution Hi Faisal, Can you copy/paste the stats of ifconfig ethN on which traffic is terminating. I just wanted to see the error and dropped packets on physical interface. Hardware and physical connectivity plays major role in packet losses.OnAsterisk server Jitter options might help you but this is media-proxy, I assume from the interface you are viewing, the packets are shown as lost. So could it be heavy media traffic flowing through the interface and media proxy is unable to use much CPU processing power to process all the RTPs ? Before troubleshooting the software application I suggest start digging the networking interfaces and tweak the ethN and related properties of kernel to maximize the throughput. This would be just how I'd go with this kind of problem. Regards. Sammy On Tue, Feb 28, 2012 at 3:45 AM, Muhammad Danish Moosa danishmo...@gmail.com wrote: Helo Bogdan only signalling packets can be lost on opensips? But rtp streams are faster and frequent and have high impact on voice quality. He seems to ask packet losses on rtp packets. Even if the problem is identified what are the clues to solve the problem? On Tue, Feb 28, 2012 at 3:14 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Faisal, Let me comment a bit on the loss at software - packages can be discarded at TCP/IP stack level (by kernel) if no application is reading the data (or no reading as fast as the data comes). You can check on this (if opensips is able to process all incoming traffic, without having the kernel to discard data because of full buffering on sockets) via some statistics from the NET class : http://www.opensips.org/Resources/DocsCoreStats17#toc17 An overall idea over the load in opensips (if you have idle processes or not) can be monitored via the LOAD stats: http://www.opensips.org/Resources/DocsCoreStats17#toc14 Regards, Bogdan On 02/27/2012 04:02 PM, Faisal Rehman wrote: Hi Everyone, I am facing huge packet loss in my server, so I am here to share with you some of the output of packets losses that you can see in attached image. Secondly I have few questions that I want to discuss with: 1. How can we reduce the packet loss to a minimum in an asterisk server, I mean I just want to know more detailed info about packet loss reduction. 2. I am calculating packet loss following that link http://www.linuxjournal.com/article/9398 where there is written that 20% loss is acceptable, but if you see the attached image what will be your conclusions about packet loss here? 3. At last but not least I just want to know the responsibilities of the software the network, I mean how much software or network is responsible for packet loss? 4. What are the best possible ways to reduce the packet loss to a minimum extent? Regards, Faisal Rehman ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Muhammad Danish Moosa The core of mans' spirit comes from new experiences. ___ Christopher McCandless ___ 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___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
