Re: [OpenSIPS-Users] Opensips 1.11 permission module problem
2314 From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, November 18, 2015 1:17 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem How many records do you have in the address table ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2015 12:14, dpa wrote: Hello Bogdan, Yes, I see errors “Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:core:db_allocate_rows: no memory left Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_convert_rows: no private memory left Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_store_result: error while converting result Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:core:db_do_query: error while storing result for query [select ip,grp,mask,port,proto,pattern,context_info,id from ast_address ] Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:permissions:reload_address_table: failed to query database” From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, November 18, 2015 1:04 PM To: OpenSIPS users mailling list; denis7...@mail.ru Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem Hi Denis, Do you see any errors in logs when the reload fails ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2015 07:29, dpa wrote: Hello Richard It is about 1 – 1,5 week. No, I do not use subnets, only IP address. “There was a package memory leak that was fixed in the development tree” Where can I read about it? Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Richard Revels Sent: Tuesday, November 17, 2015 8:55 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem Are you using subnets in your address table? How often do you reload the address table? There was a package memory leak that was fixed in the development tree that might explain what you are seeing. On Wed, Nov 11, 2015 at 4:26 AM, dpa <denis7...@mail.ru> wrote: Hello! Is there any assumption about problem? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Wednesday, October 28, 2015 4:12 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem Hello! OpenSIPS (1.11.5-notls (x86_64/linux)) I have a periodic problem with permissions module. I could not make opensipsctl address reload (“400 Trusted table reload failed” received). In the time I have no problem with dialplan or drouting modules. Opensips reload solve the problem. In attachment log from opensips and statistic. Thank you for any help ___ 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 1.11 permission module problem
Hello Bogdan, Yes, I see errors “Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:core:db_allocate_rows: no memory left Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_convert_rows: no private memory left Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_store_result: error while converting result Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:core:db_do_query: error while storing result for query [select ip,grp,mask,port,proto,pattern,context_info,id from ast_address ] Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:permissions:reload_address_table: failed to query database” From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, November 18, 2015 1:04 PM To: OpenSIPS users mailling list; denis7...@mail.ru Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem Hi Denis, Do you see any errors in logs when the reload fails ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2015 07:29, dpa wrote: Hello Richard It is about 1 – 1,5 week. No, I do not use subnets, only IP address. “There was a package memory leak that was fixed in the development tree” Where can I read about it? Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Richard Revels Sent: Tuesday, November 17, 2015 8:55 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem Are you using subnets in your address table? How often do you reload the address table? There was a package memory leak that was fixed in the development tree that might explain what you are seeing. On Wed, Nov 11, 2015 at 4:26 AM, dpa <denis7...@mail.ru> wrote: Hello! Is there any assumption about problem? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Wednesday, October 28, 2015 4:12 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem Hello! OpenSIPS (1.11.5-notls (x86_64/linux)) I have a periodic problem with permissions module. I could not make opensipsctl address reload (“400 Trusted table reload failed” received). In the time I have no problem with dialplan or drouting modules. Opensips reload solve the problem. In attachment log from opensips and statistic. Thank you for any help ___ 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 1.11 permission module problem
32M From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, November 18, 2015 5:24 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem How much pkg memory have you configured in your opensips ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2015 12:24, dpa wrote: 2314 From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, November 18, 2015 1:17 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem How many records do you have in the address table ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2015 12:14, dpa wrote: Hello Bogdan, Yes, I see errors “Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:core:db_allocate_rows: no memory left Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_convert_rows: no private memory left Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_store_result: error while converting result Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:core:db_do_query: error while storing result for query [select ip,grp,mask,port,proto,pattern,context_info,id from ast_address ] Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:permissions:reload_address_table: failed to query database” From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, November 18, 2015 1:04 PM To: OpenSIPS users mailling list; denis7...@mail.ru Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem Hi Denis, Do you see any errors in logs when the reload fails ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2015 07:29, dpa wrote: Hello Richard It is about 1 – 1,5 week. No, I do not use subnets, only IP address. “There was a package memory leak that was fixed in the development tree” Where can I read about it? Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Richard Revels Sent: Tuesday, November 17, 2015 8:55 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem Are you using subnets in your address table? How often do you reload the address table? There was a package memory leak that was fixed in the development tree that might explain what you are seeing. On Wed, Nov 11, 2015 at 4:26 AM, dpa <denis7...@mail.ru> wrote: Hello! Is there any assumption about problem? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Wednesday, October 28, 2015 4:12 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem Hello! OpenSIPS (1.11.5-notls (x86_64/linux)) I have a periodic problem with permissions module. I could not make opensipsctl address reload (“400 Trusted table reload failed” received). In the time I have no problem with dialplan or drouting modules. Opensips reload solve the problem. In attachment log from opensips and statistic. Thank you for any help ___ 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 1.11 permission module problem
Hello Richard It is about 1 – 1,5 week. No, I do not use subnets, only IP address. “There was a package memory leak that was fixed in the development tree” Where can I read about it? Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Richard Revels Sent: Tuesday, November 17, 2015 8:55 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.11 permission module problem Are you using subnets in your address table? How often do you reload the address table? There was a package memory leak that was fixed in the development tree that might explain what you are seeing. On Wed, Nov 11, 2015 at 4:26 AM, dpa <denis7...@mail.ru> wrote: Hello! Is there any assumption about problem? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Wednesday, October 28, 2015 4:12 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem Hello! OpenSIPS (1.11.5-notls (x86_64/linux)) I have a periodic problem with permissions module. I could not make opensipsctl address reload (“400 Trusted table reload failed” received). In the time I have no problem with dialplan or drouting modules. Opensips reload solve the problem. In attachment log from opensips and statistic. Thank you for any help ___ 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 1.11 permission module problem
Hello! Is there any assumption about problem? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Wednesday, October 28, 2015 4:12 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Opensips 1.11 permission module problem Hello! OpenSIPS (1.11.5-notls (x86_64/linux)) I have a periodic problem with permissions module. I could not make opensipsctl address reload ("400 Trusted table reload failed" received). In the time I have no problem with dialplan or drouting modules. Opensips reload solve the problem. In attachment log from opensips and statistic. Thank you for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips and TLS
Hello! I want to ask an advice for making one scheme. There is one softswitch (rather old) which does not support TLS protocol. Can I use Opensips as TLS intermediary between some SIP UA and old softswitch? Scheme: SIP UA ßà Opensips (with TLS) ßà old softswitch. Old softswitch must know SIP UA status (register/unregister) and must make incoming call to SIP UA. Thank you for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.11 permission module problem
Hello! OpenSIPS (1.11.5-notls (x86_64/linux)) I have a periodic problem with permissions module. I could not make opensipsctl address reload ("400 Trusted table reload failed" received). In the time I have no problem with dialplan or drouting modules. Opensips reload solve the problem. In attachment log from opensips and statistic. Thank you for any help Log from opensips Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:core:db_allocate_rows: no memory left Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_convert_rows: no private memory left Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:db_mysql:db_mysql_store_result: error while converting result Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:core:db_do_query: error while storing result for query [select ip,grp,mask,port,proto,pattern,context_info,id from ast_address ] Oct 28 15:55:45 opensips-main /usr/local/opensips1.11/sbin/opensips[7715]: ERROR:permissions:reload_address_table: failed to query database Statistic shmem:total_size:: 536870912 shmem:used_size:: 98591856 shmem:real_used_size:: 163253328 shmem:max_used_size:: 327067912 shmem:free_size:: 373617584 shmem:fragments:: 1246896 pkmem:0-total_size:: 0 pkmem:0-used_size:: 0 pkmem:0-real_used_size:: 0 pkmem:0-max_used_size:: 0 pkmem:0-free_size:: 0 pkmem:0-fragments:: 0 pkmem:1-total_size:: 33554432 pkmem:1-used_size:: 931760 pkmem:1-real_used_size:: 1154672 pkmem:1-max_used_size:: 4667952 pkmem:1-free_size:: 32399760 pkmem:1-fragments:: 32 pkmem:2-total_size:: 33554432 pkmem:2-used_size:: 989496 pkmem:2-real_used_size:: 1208520 pkmem:2-max_used_size:: 4667952 pkmem:2-free_size:: 32345912 pkmem:2-fragments:: 5 pkmem:3-total_size:: 33554432 pkmem:3-used_size:: 23821480 pkmem:3-real_used_size:: 30034288 pkmem:3-max_used_size:: 30460160 pkmem:3-free_size:: 3520144 pkmem:3-fragments:: 1102 pkmem:4-total_size:: 33554432 pkmem:4-used_size:: 950048 pkmem:4-real_used_size:: 1245200 pkmem:4-max_used_size:: 4720208 pkmem:4-free_size:: 32309232 pkmem:4-fragments:: 3171 pkmem:5-total_size:: 33554432 pkmem:5-used_size:: 950352 pkmem:5-real_used_size:: 1245672 pkmem:5-max_used_size:: 4667952 pkmem:5-free_size:: 32308760 pkmem:5-fragments:: 3178 pkmem:6-total_size:: 33554432 pkmem:6-used_size:: 949392 pkmem:6-real_used_size:: 1246152 pkmem:6-max_used_size:: 4667952 pkmem:6-free_size:: 32308280 pkmem:6-fragments:: 3238 pkmem:7-total_size:: 33554432 pkmem:7-used_size:: 950592 pkmem:7-real_used_size:: 1245840 pkmem:7-max_used_size:: 4667952 pkmem:7-free_size:: 32308592 pkmem:7-fragments:: 3175 pkmem:8-total_size:: 33554432 pkmem:8-used_size:: 949192 pkmem:8-real_used_size:: 1245736 pkmem:8-max_used_size:: 4667952 pkmem:8-free_size:: 32308696 pkmem:8-fragments:: 3229 pkmem:9-total_size:: 33554432 pkmem:9-used_size:: 949368 pkmem:9-real_used_size:: 1244328 pkmem:9-max_used_size:: 4667952 pkmem:9-free_size:: 32310104 pkmem:9-fragments:: 3163 pkmem:10-total_size:: 33554432 pkmem:10-used_size:: 949672 pkmem:10-real_used_size:: 1244056 pkmem:10-max_used_size:: 4667952 pkmem:10-free_size:: 32310376 pkmem:10-fragments:: 3139 pkmem:11-total_size:: 33554432 pkmem:11-used_size:: 951784 pkmem:11-real_used_size:: 1247896 pkmem:11-max_used_size:: 4667952 pkmem:11-free_size:: 32306536 pkmem:11-fragments:: 3211 pkmem:12-total_size:: 33554432 pkmem:12-used_size:: 954664 pkmem:12-real_used_size:: 1246744 pkmem:12-max_used_size:: 4667952 pkmem:12-free_size:: 32307688 pkmem:12-fragments:: 3044 pkmem:13-total_size:: 33554432 pkmem:13-used_size:: 953584 pkmem:13-real_used_size:: 1246264 pkmem:13-max_used_size:: 4667952 pkmem:13-free_size:: 32308168 pkmem:13-fragments:: 3069 pkmem:14-total_size:: 33554432 pkmem:14-used_size:: 954288 pkmem:14-real_used_size:: 1246008 pkmem:14-max_used_size:: 4667952 pkmem:14-free_size:: 32308424 pkmem:14-fragments:: 3029 pkmem:15-total_size:: 33554432 pkmem:15-used_size:: 954304 pkmem:15-real_used_size:: 1246096 pkmem:15-max_used_size:: 4667952 pkmem:15-free_size:: 32308336 pkmem:15-fragments:: 3032 pkmem:16-total_size:: 33554432 pkmem:16-used_size:: 954712 pkmem:16-real_used_size:: 1245832 pkmem:16-max_used_size:: 4667952 pkmem:16-free_size:: 32308600 pkmem:16-fragments:: 3004 pkmem:17-total_size:: 33554432 pkmem:17-used_size:: 953728 pkmem:17-real_used_size:: 1244944 pkmem:17-max_used_size:: 4667952 pkmem:17-free_size:: 32309488 pkmem:17-fragments:: 3008 pkmem:18-total_size:: 33554432 pkmem:18-used_size:: 954488 pkmem:18-real_used_size:: 1245800 pkmem:18-max_used_size:: 4667952 pkmem:18-free_size:: 32308632
Re: [OpenSIPS-Users] opensips1.11 dr_routing
Hello Bogdan! Yes, you are right, I want use only carriers in routing. For example. I have for providers #14, #34, #35, #2881. Let`s say that the goal of some prefix is Prefix 9087: #14 - 1$, #34-2$, #35 - 2$, #2881 - 3$ Prefix 8765: #34-1$, #35-2$, #14-3$, #2881 - 3$ In ideal I want to see in routing table: 9087 #14,#34=50,#35=50,#2881 8765 #34,#35,#14=50,#2881=50 First Opensips make serial routing and then parallel. In method you gave I must create carriers: 34=50, 35=50 34=33, 35=33, 2881=33 an so on. And all these carriers I must remember in another system (which forms routing table) and using it during forming routing table. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 4:35 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Denis, So what you want to do is to be able to serial and parallel (combined) having carriers as elements in the list (and not GWs). With the solution I gave you cannot do it as you cannot use carriers in the definition of another carrier (when doing the parallel group), right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 16:14, dpa wrote: No, Bogdan, everything fine. You wright understand my question. I wanted do a combination between serial and parallel selection but method you suggest doesn`t fit to me. The main reason I have many providers and using carrier for sharing doesn`t convenient, because I must have all possible combination of carriers and using (remember) it for making routing table (contents of routing table formed by another system, and I do not want complicate it). Thank you. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 3:52 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Denis, I had the impression that I haven't managed to answer to your question. Maybe because I do not fully understand the question. Could you detail why the solution I gave does not fit you? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 15:35, dpa wrote: Hello Bogdan! I understand and I thought about it before. But it doesn`t fit to me, because I have many prefix which can sharing different carriers. Anyway, thank you for help. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 3:29 PM To: OpenSIPS users mailling list; denis7...@mail.ru Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Hi Denis, If you use weights alg and the GW has no weight, a 0 weight will be assumed, so it will never match (unless after a failover). Using weights, the selection is over all the GW in the set. If you want do a combination between serial and parallel selection , you need to use carriers: Make carrier X with wights flags and set for it destinations 34=50,35=50 and in the rule have the destination set 14,#X (and do not use the W flag anymore) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 12:16, dpa wrote: Hello! There is one question about dr_routing module. Example 1. I have such gw_list #14,#34=50,#35=50,#2881 Example 2. I have such gw_list #14,#34,#35,#2881 I am using d_routing() with W flag. In Example 1 a call first had been sent to sharing between #34 and #35 and only then to #14. In Example 2 a call first had been sent to #14, then #34, then #35 and #2881. Is there any way to tell Opensips in Example 1 first sends calls to #14 and only then to sharing between #34 and #35? Thank you for any help. ___ 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] opensips1.11 dr_routing
Second point seems interesting. I will test it. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Monday, August 24, 2015 12:49 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Hi Denis, Currently you cannot do this, but I see 2 options: 1) allow a carrier definition to contain other carriers (max 3 levels) 2) have a more complex syntax in destination set definition, with brackets, like: #14,(#34=50,#35=50),#2881 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.08.2015 10:42, dpa wrote: Hello Bogdan! Yes, you are right, I want use only carriers in routing. For example. I have for providers #14, #34, #35, #2881. Let`s say that the goal of some prefix is Prefix 9087: #14 - 1$, #34-2$, #35 - 2$, #2881 - 3$ Prefix 8765: #34-1$, #35-2$, #14-3$, #2881 - 3$ In ideal I want to see in routing table: 9087 #14,#34=50,#35=50,#2881 8765 #34,#35,#14=50,#2881=50 First Opensips make serial routing and then parallel. In method you gave I must create carriers: 34=50, 35=50 34=33, 35=33, 2881=33 an so on. And all these carriers I must remember in another system (which forms routing table) and using it during forming routing table. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 4:35 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Denis, So what you want to do is to be able to serial and parallel (combined) having carriers as elements in the list (and not GWs). With the solution I gave you cannot do it as you cannot use carriers in the definition of another carrier (when doing the parallel group), right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 16:14, dpa wrote: No, Bogdan, everything fine. You wright understand my question. I wanted do a combination between serial and parallel selection but method you suggest doesn`t fit to me. The main reason I have many providers and using carrier for sharing doesn`t convenient, because I must have all possible combination of carriers and using (remember) it for making routing table (contents of routing table formed by another system, and I do not want complicate it). Thank you. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 3:52 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Denis, I had the impression that I haven't managed to answer to your question. Maybe because I do not fully understand the question. Could you detail why the solution I gave does not fit you? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 15:35, dpa wrote: Hello Bogdan! I understand and I thought about it before. But it doesn`t fit to me, because I have many prefix which can sharing different carriers. Anyway, thank you for help. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 3:29 PM To: OpenSIPS users mailling list; denis7...@mail.ru Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Hi Denis, If you use weights alg and the GW has no weight, a 0 weight will be assumed, so it will never match (unless after a failover). Using weights, the selection is over all the GW in the set. If you want do a combination between serial and parallel selection , you need to use carriers: Make carrier X with wights flags and set for it destinations 34=50,35=50 and in the rule have the destination set 14,#X (and do not use the W flag anymore) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 12:16, dpa wrote: Hello! There is one question about dr_routing module. Example 1. I have such gw_list #14,#34=50,#35=50,#2881 Example 2. I have such gw_list #14,#34,#35,#2881 I am using d_routing() with W flag. In Example 1 a call first had been sent to sharing between #34 and #35 and only then to #14. In Example 2 a call first had been sent to #14, then #34, then #35 and #2881. Is there any way to tell Opensips in Example 1 first sends calls to #14 and only then to sharing between #34 and #35? Thank you for any help. ___ 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] opensips1.11 dr_routing
Ooops and I was very happy some minutes ago:))) Ok, will wait an options. Thank you! From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Monday, August 24, 2015 12:58 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Hold on ! none of the options are available now . I suggested them as options for a fix ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.08.2015 12:55, dpa wrote: Second point seems interesting. I will test it. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Monday, August 24, 2015 12:49 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Hi Denis, Currently you cannot do this, but I see 2 options: 1) allow a carrier definition to contain other carriers (max 3 levels) 2) have a more complex syntax in destination set definition, with brackets, like: #14,(#34=50,#35=50),#2881 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.08.2015 10:42, dpa wrote: Hello Bogdan! Yes, you are right, I want use only carriers in routing. For example. I have for providers #14, #34, #35, #2881. Let`s say that the goal of some prefix is Prefix 9087: #14 - 1$, #34-2$, #35 - 2$, #2881 - 3$ Prefix 8765: #34-1$, #35-2$, #14-3$, #2881 - 3$ In ideal I want to see in routing table: 9087 #14,#34=50,#35=50,#2881 8765 #34,#35,#14=50,#2881=50 First Opensips make serial routing and then parallel. In method you gave I must create carriers: 34=50, 35=50 34=33, 35=33, 2881=33 an so on. And all these carriers I must remember in another system (which forms routing table) and using it during forming routing table. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 4:35 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Denis, So what you want to do is to be able to serial and parallel (combined) having carriers as elements in the list (and not GWs). With the solution I gave you cannot do it as you cannot use carriers in the definition of another carrier (when doing the parallel group), right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 16:14, dpa wrote: No, Bogdan, everything fine. You wright understand my question. I wanted do a combination between serial and parallel selection but method you suggest doesn`t fit to me. The main reason I have many providers and using carrier for sharing doesn`t convenient, because I must have all possible combination of carriers and using (remember) it for making routing table (contents of routing table formed by another system, and I do not want complicate it). Thank you. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 3:52 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Denis, I had the impression that I haven't managed to answer to your question. Maybe because I do not fully understand the question. Could you detail why the solution I gave does not fit you? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 15:35, dpa wrote: Hello Bogdan! I understand and I thought about it before. But it doesn`t fit to me, because I have many prefix which can sharing different carriers. Anyway, thank you for help. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 3:29 PM To: OpenSIPS users mailling list; denis7...@mail.ru Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Hi Denis, If you use weights alg and the GW has no weight, a 0 weight will be assumed, so it will never match (unless after a failover). Using weights, the selection is over all the GW in the set. If you want do a combination between serial and parallel selection , you need to use carriers: Make carrier X with wights flags and set for it destinations 34=50,35=50 and in the rule have the destination set 14,#X (and do not use the W flag anymore) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 12:16, dpa wrote: Hello! There is one question about dr_routing module. Example 1. I have such gw_list #14,#34=50,#35=50,#2881 Example 2. I have such gw_list #14,#34,#35,#2881 I am using d_routing() with W flag. In Example 1 a call first had been sent to sharing between #34 and #35 and only then to #14. In Example 2 a call first had been sent to #14, then #34, then #35 and #2881. Is there any way to tell Opensips in Example 1 first sends calls to #14 and only then to sharing between #34 and #35? Thank you for any help
Re: [OpenSIPS-Users] opensips1.11 dr_routing
Hello Bogdan! I understand and I thought about it before. But it doesn`t fit to me, because I have many prefix which can sharing different carriers. Anyway, thank you for help. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 3:29 PM To: OpenSIPS users mailling list; denis7...@mail.ru Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Hi Denis, If you use weights alg and the GW has no weight, a 0 weight will be assumed, so it will never match (unless after a failover). Using weights, the selection is over all the GW in the set. If you want do a combination between serial and parallel selection , you need to use carriers: Make carrier X with wights flags and set for it destinations 34=50,35=50 and in the rule have the destination set 14,#X (and do not use the W flag anymore) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 12:16, dpa wrote: Hello! There is one question about dr_routing module. Example 1. I have such gw_list #14,#34=50,#35=50,#2881 Example 2. I have such gw_list #14,#34,#35,#2881 I am using d_routing() with W flag. In Example 1 a call first had been sent to sharing between #34 and #35 and only then to #14. In Example 2 a call first had been sent to #14, then #34, then #35 and #2881. Is there any way to tell Opensips in Example 1 first sends calls to #14 and only then to sharing between #34 and #35? Thank you for any help. ___ 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] opensips1.11 dr_routing
No, Bogdan, everything fine. You wright understand my question. I wanted do a combination between serial and parallel selection but method you suggest doesn`t fit to me. The main reason I have many providers and using carrier for sharing doesn`t convenient, because I must have all possible combination of carriers and using (remember) it for making routing table (contents of routing table formed by another system, and I do not want complicate it). Thank you. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 3:52 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Denis, I had the impression that I haven't managed to answer to your question. Maybe because I do not fully understand the question. Could you detail why the solution I gave does not fit you? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 15:35, dpa wrote: Hello Bogdan! I understand and I thought about it before. But it doesn`t fit to me, because I have many prefix which can sharing different carriers. Anyway, thank you for help. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, August 21, 2015 3:29 PM To: OpenSIPS users mailling list; denis7...@mail.ru Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing Hi Denis, If you use weights alg and the GW has no weight, a 0 weight will be assumed, so it will never match (unless after a failover). Using weights, the selection is over all the GW in the set. If you want do a combination between serial and parallel selection , you need to use carriers: Make carrier X with wights flags and set for it destinations 34=50,35=50 and in the rule have the destination set 14,#X (and do not use the W flag anymore) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.08.2015 12:16, dpa wrote: Hello! There is one question about dr_routing module. Example 1. I have such gw_list #14,#34=50,#35=50,#2881 Example 2. I have such gw_list #14,#34,#35,#2881 I am using d_routing() with W flag. In Example 1 a call first had been sent to sharing between #34 and #35 and only then to #14. In Example 2 a call first had been sent to #14, then #34, then #35 and #2881. Is there any way to tell Opensips in Example 1 first sends calls to #14 and only then to sharing between #34 and #35? Thank you for any help. ___ 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] opensips1.11 dr_routing
Hello! There is one question about dr_routing module. Example 1. I have such gw_list #14,#34=50,#35=50,#2881 Example 2. I have such gw_list #14,#34,#35,#2881 I am using d_routing() with W flag. In Example 1 a call first had been sent to sharing between #34 and #35 and only then to #14. In Example 2 a call first had been sent to #14, then #34, then #35 and #2881. Is there any way to tell Opensips in Example 1 first sends calls to #14 and only then to sharing between #34 and #35? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] FW: Opensips 1.10 crash
Hello! Is there any ideas about problem? Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Wednesday, July 15, 2015 4:11 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Opensips 1.10 crash Hello! Opensips1.10.1. Working with rtpproxy2.0. Today Opensips has crashed. Core file was generated. Information from core file is in attachment. The question is why Opensips has crashed? Thank you for any help. gdb /usr/local/opensips1.10.1/sbin/opensips /opensipscore/core GNU gdb (Ubuntu/Linaro 7.4-2012.04-0ubuntu2.1) 7.4-2012.04 Copyright (C) 2012 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as x86_64-linux-gnu. For bug reporting instructions, please see: http://bugs.launchpad.net/gdb-linaro/... Reading symbols from /usr/local/opensips1.10.1/sbin/opensips...done. [New LWP 9791] warning: Can't read pathname for load map: Îøèáêà ââîäà/âûâîäà. [Thread debugging using libthread_db enabled] Using host libthread_db library /lib/x86_64-linux-gnu/libthread_db.so.1. warning: no loadable sections found in added symbol-file system-supplied DSO at 0x7fff78bfe000 Core was generated by `/usr/local/opensips1.10.1/sbin/opensips -P /sock/opensips.pid -u opensips -w /o'. Program terminated with signal 11, Segmentation fault. #0 fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175 175 n-u.nxt_free-prev = pf; (gdb) bt #0 fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175 #1 fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383 #2 0x7fcf685860a5 in shm_malloc_unsafe (size=optimized out) at ../../mem/shm_mem.h:248 #3 shm_malloc (size=optimized out) at ../../mem/shm_mem.h:258 #4 add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367 #5 0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=optimized out, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260 #6 0x7fcf685715a4 in add_rule (rule=optimized out, prefix=optimized out, grplst=optimized out, rdata=optimized out) at dr_load.c:188 #7 dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, drd_table=optimized out, drc_table=optimized out, drr_table=0x7fcf687939f0) at dr_load.c:512 #8 0x7fcf685811c0 in dr_reload_data () at drouting.c:425 #9 dr_reload_cmd (cmd_tree=optimized out, param=optimized out) at drouting.c:813 #10 0x7fcf68baf504 in run_mi_cmd (param=0x25e5aa0, f=optimized out, t=0x0, cmd=optimized out) at ../../mi/mi.h:109 #11 mi_fifo_server (fifo_stream=0x25e8e30) at fifo_fnc.c:490 #12 0x7fcf68bb0bef in fifo_process (rank=optimized out) at mi_fifo.c:213 #13 0x004b9de5 in start_module_procs () at sr_module.c:586 #14 0x0041475a in main_loop () at main.c:840 #15 main (argc=optimized out, argv=optimized out) at main.c:1598 (gdb) bt full #0 fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175 pf = 0x7fcf44a9b640 hash = 2096 #1 fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383 frag = 0x7fcf529e6840 n = optimized out hash = optimized out __FUNCTION__ = fm_malloc #2 0x7fcf685860a5 in shm_malloc_unsafe (size=optimized out) at ../../mem/shm_mem.h:248 p = optimized out #3 shm_malloc (size=optimized out) at ../../mem/shm_mem.h:258 p = 0x2c0 #4 add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367 trg = optimized out rtl_wrp = optimized out rtlw = 0x0 i = optimized out __FUNCTION__ = add_rt_info #5 0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=optimized out, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260 tmp = optimized out res = 0 __FUNCTION__ = add_prefix #6 0x7fcf685715a4 in add_rule (rule=optimized out, prefix=optimized out, grplst=optimized out, rdata=optimized out) at dr_load.c:188 tmp = optimized out t = optimized out ep = 0x2aa79cc ,130,139,142,145,148 n = optimized out #7 dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, drd_table=optimized out, drc_table=optimized out, drr_table=0x7fcf687939f0) at dr_load.c:512 int_vals = {1522874, optimized out, 0, optimized out} str_vals = {0x2aa7948 19,20,23,25,27,31,33,35,37,29,40,44,47,50,12,53,56,59,62,65,66,69,72,75,78,81,84,97,88,91,94,100,103,106,109,112,115,118,121,124,127,130,139,142,145,148, optimized out, optimized out, 0x2aa79ed 13, optimized out, optimized out} tmp = {s = 0x2aa79e1 81061881, len = 8} columns = {0x7fcf68793400, 0x7fcf68793410, 0x7fcf68793420, 0x7fcf68793430
[OpenSIPS-Users] changing $rU number
Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as %20 and -. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing $rU number
Thank you. But what can I use for do it? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Thursday, July 23, 2015 10:40 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as “%20” and “-”. The question is how can I delete these characters from request user? Thank you for any help. ___ 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] changing $rU number
I understand, thank you. I will try. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Trevor Steyn Sent: Thursday, July 23, 2015 12:46 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] changing $rU number Documentation can be found here http://www.opensips.org/html/docs/modules/1.10.x/dialplan.html For egsample to remove all - characters in $rU do the following Add the following into the dialplan table mysql select * from dialplan\G; *** 1. row *** id: 6 dpid: 1 pr: 98 match_op: 0 match_exp: - match_flags: 0 subst_exp: repl_exp: timerec: disabled: 0 attrs: *** 2. row *** Then call dp_translate(1,$rU/$rU); in your script. Regards Trevor Steyn On 23/07/2015 10:51, dpa wrote: And how dialplan helps me to do it, if ,for example, one time I have such characters 8%2089-0987-09, and in another time I have 987-89%20908-1? *From:*users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Trevor Steyn *Sent:* Thursday, July 23, 2015 11:27 AM *To:* users@lists.opensips.org *Subject:* Re: [OpenSIPS-Users] changing $rU number Hi Denis You can use the dialplan module to rewrite variables as below dp_translate(1,$rU/$rU); Then insert you regexp into the dialplan tables, read the dialplan module documentation for more info. Regards Trevor On 23/07/2015 09:49, dpa wrote: Thank you. But what can I use for do it? *From:*users-boun...@lists.opensips.org mailto:users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Schneur Rosenberg *Sent:* Thursday, July 23, 2015 10:40 AM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru mailto:denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as %20 and -. The question is how can I delete these characters from request user? Thank you for any help. ___ 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 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing $rU number
And how dialplan helps me to do it, if ,for example, one time I have such characters 8%2089-0987-09, and in another time I have 987-89%20908-1? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Trevor Steyn Sent: Thursday, July 23, 2015 11:27 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] changing $rU number Hi Denis You can use the dialplan module to rewrite variables as below dp_translate(1,$rU/$rU); Then insert you regexp into the dialplan tables, read the dialplan module documentation for more info. Regards Trevor On 23/07/2015 09:49, dpa wrote: Thank you. But what can I use for do it? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Thursday, July 23, 2015 10:40 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as %20 and -. The question is how can I delete these characters from request user? Thank you for any help. ___ 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] changing $rU number
As I understand regex module just matches string against regexp. but not makes any manipulation. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Thursday, July 23, 2015 11:24 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] changing $rU number http://www.opensips.org/html/docs/modules/1.10.x/regex.html On Thu, Jul 23, 2015 at 10:49 AM, dpa denis7...@mail.ru wrote: Thank you. But what can I use for do it? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Thursday, July 23, 2015 10:40 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as “%20” and “-”. The question is how can I delete these characters from request user? Thank you for any help. ___ 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] Ref: Opensips is not terminating call by “BYE” instead it is sending 404 error
Hello Opensips does not responding to BYE (if you specially didn`t “ask” to do it). It just retransmit BYE to another SIP UA and wait a response from it. Once response had been got Opensips retransmit this response to the first SIP UA (witch initiated BYE). If Opensips doesn`t retransmit BYE than something wrong with BYE message or in opensips.cfg script. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of AMPTEL PTY LTD | RuvixTel Sent: Sunday, July 19, 2015 4:01 PM To: users@lists.opensips.org Cc: rajat kapoor Subject: [OpenSIPS-Users] Ref: Opensips is not terminating call by “BYE” instead it is sending 404 error Hi, Please help us to resolve below issue will be appreciated. Our Opensips server (version1.6) is not responding to “Bye sent from termination server. This is causing the calls to not end properly as it should. By checking sample SIP traces , it shows SIP server is always responding with a 404 not found respond of termination server's Bye message and keeps on sending re-invites request. -- Best Regards + Anup AMPtel PTY LTD, ( RuvixTel ) ABN: 77 162 081 905 http://abr.business.gov.au/SearchByAbn.aspx?SearchText=77+162+081+905 Ph: +61 413 777 075 (Anup) Ph: +61 433 232 177 (Piyali) E-mail: i...@ruvixtel.net http://thedaneshproject.com/wp-content/uploads/2010/04/Green_footers_8.gif ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.10 crash
Hello! Opensips1.10.1. Working with rtpproxy2.0. Today Opensips has crashed. Core file was generated. Information from core file is in attachment. The question is why Opensips has crashed? Thank you for any help. gdb /usr/local/opensips1.10.1/sbin/opensips /opensipscore/core GNU gdb (Ubuntu/Linaro 7.4-2012.04-0ubuntu2.1) 7.4-2012.04 Copyright (C) 2012 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as x86_64-linux-gnu. For bug reporting instructions, please see: http://bugs.launchpad.net/gdb-linaro/... Reading symbols from /usr/local/opensips1.10.1/sbin/opensips...done. [New LWP 9791] warning: Can't read pathname for load map: Îøèáêà ââîäà/âûâîäà. [Thread debugging using libthread_db enabled] Using host libthread_db library /lib/x86_64-linux-gnu/libthread_db.so.1. warning: no loadable sections found in added symbol-file system-supplied DSO at 0x7fff78bfe000 Core was generated by `/usr/local/opensips1.10.1/sbin/opensips -P /sock/opensips.pid -u opensips -w /o'. Program terminated with signal 11, Segmentation fault. #0 fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175 175 n-u.nxt_free-prev = pf; (gdb) bt #0 fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175 #1 fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383 #2 0x7fcf685860a5 in shm_malloc_unsafe (size=optimized out) at ../../mem/shm_mem.h:248 #3 shm_malloc (size=optimized out) at ../../mem/shm_mem.h:258 #4 add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367 #5 0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=optimized out, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260 #6 0x7fcf685715a4 in add_rule (rule=optimized out, prefix=optimized out, grplst=optimized out, rdata=optimized out) at dr_load.c:188 #7 dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, drd_table=optimized out, drc_table=optimized out, drr_table=0x7fcf687939f0) at dr_load.c:512 #8 0x7fcf685811c0 in dr_reload_data () at drouting.c:425 #9 dr_reload_cmd (cmd_tree=optimized out, param=optimized out) at drouting.c:813 #10 0x7fcf68baf504 in run_mi_cmd (param=0x25e5aa0, f=optimized out, t=0x0, cmd=optimized out) at ../../mi/mi.h:109 #11 mi_fifo_server (fifo_stream=0x25e8e30) at fifo_fnc.c:490 #12 0x7fcf68bb0bef in fifo_process (rank=optimized out) at mi_fifo.c:213 #13 0x004b9de5 in start_module_procs () at sr_module.c:586 #14 0x0041475a in main_loop () at main.c:840 #15 main (argc=optimized out, argv=optimized out) at main.c:1598 (gdb) bt full #0 fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175 pf = 0x7fcf44a9b640 hash = 2096 #1 fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383 frag = 0x7fcf529e6840 n = optimized out hash = optimized out __FUNCTION__ = fm_malloc #2 0x7fcf685860a5 in shm_malloc_unsafe (size=optimized out) at ../../mem/shm_mem.h:248 p = optimized out #3 shm_malloc (size=optimized out) at ../../mem/shm_mem.h:258 p = 0x2c0 #4 add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367 trg = optimized out rtl_wrp = optimized out rtlw = 0x0 i = optimized out __FUNCTION__ = add_rt_info #5 0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=optimized out, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260 tmp = optimized out res = 0 __FUNCTION__ = add_prefix #6 0x7fcf685715a4 in add_rule (rule=optimized out, prefix=optimized out, grplst=optimized out, rdata=optimized out) at dr_load.c:188 tmp = optimized out t = optimized out ep = 0x2aa79cc ,130,139,142,145,148 n = optimized out #7 dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, drd_table=optimized out, drc_table=optimized out, drr_table=0x7fcf687939f0) at dr_load.c:512 int_vals = {1522874, optimized out, 0, optimized out} str_vals = {0x2aa7948 19,20,23,25,27,31,33,35,37,29,40,44,47,50,12,53,56,59,62,65,66,69,72,75,78,81,84,97,88,91,94,100,103,106,109,112,115,118,121,124,127,130,139,142,145,148, optimized out, optimized out, 0x2aa79ed 13, optimized out, optimized out} tmp = {s = 0x2aa79e1 81061881, len = 8} columns = {0x7fcf68793400, 0x7fcf68793410, 0x7fcf68793420, 0x7fcf68793430, 0x7fcf68793440, 0x7fcf68793450, 0x7fcf68793460, 0x7fcf68793390} res = 0x7fcf6c24dd48 row = optimized out ri = 0x7fcf4f6bc9f8 rdata = 0x7fcf4955e898 time_rec = optimized out i = 4018 n = 29697 no_rows = optimized out
Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem
Hello Bogdan What I can see in logs during test call ERROR:aaa_radius:send_auth_func: radius authentication message failed with ERROR From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, June 05, 2014 6:20 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Hi, -1 ret code means some internal error - check the opensips logs for any error. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03.06.2014 07:01, dpa wrote: Hello Bogdan Sorry for long answer. I changed opensips.cfg just a little .. radius_send_auth(set1,set2); switch($rc){ case -2: xlog (Case -2 in radius route detected); break; case -1: xlog (Case -1 in radius route detected); break; } .. In log file I see Jun 3 07:19:13 opensips-mirror /usr/local/opensipsdev/sbin/opensips[14956]: Case -1 in radius route detected Meanwhile tcpdump shows that radius server answers Rejected on the request. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, May 29, 2014 11:26 AM To: OpenSIPS users mailling list; Denis Putyato Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Hi, On each case, please an xlog to print something from the script - in this way you can double check which way your script went. Or, you can use the script_trace() function to do that . See: http://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc42 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 29.05.2014 09:22, dpa wrote: Hello! I have such configuration in opensips.cfg .. radius_send_auth(set1,set2); switch($rc){ case -2: exec_msg(echo '$avp(500) $rU $time(%c) rejected' /opensips/alarmradius.txt); break; case -1: exec_msg(echo '$avp(500) $rU $time(%c) radius' /opensips/alarmradius.txt); break; } . In attachment you can see communication of Opensips with radius server during making some test call (1.1.1.1 - Opensips, 2.2.2.2 - radius server) After making a test call i see in alarmradius.txt record with radius word. The question is why do I see radius word (and it means that Opensips detected retcode -1) but not rejected (retcode -2) word? Thank you for any help. ___ 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 1.10.1 and radius problem
Hello Bogdan Yes, I see reject. I will try the patch and let you know about the result From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, June 06, 2014 1:21 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Hi, The libradius returns the error via the rc_auth() function. I notices a bit of an inconsistency in the handling of the retcode of rc_auth() - old stuff :). You say you see a REJECT and the RADIUS level, right ? Could you please try the attached patch and see if the failures are properly reported to the script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 06.06.2014 11:27, dpa wrote: Hello Bogdan What I can see in logs during test call ERROR:aaa_radius:send_auth_func: radius authentication message failed with ERROR From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, June 05, 2014 6:20 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Hi, -1 ret code means some internal error - check the opensips logs for any error. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03.06.2014 07:01, dpa wrote: Hello Bogdan Sorry for long answer. I changed opensips.cfg just a little .. radius_send_auth(set1,set2); switch($rc){ case -2: xlog (Case -2 in radius route detected); break; case -1: xlog (Case -1 in radius route detected); break; } ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem
Unfortunately it doesn`t help. I tried the patch (make command I gave from module/aaa_radius directory). A test call shows Jun 6 13:57:42 opensips-mirror /usr/local/opensipsdev/sbin/opensips[4297]: ERROR:aaa_radius:send_auth_func: radius authentication message failed with ERROR Jun 6 13:57:42 opensips-mirror /usr/local/opensipsdev/sbin/opensips[4297]: Case -1 in radius route detected From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, June 06, 2014 1:21 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Hi, The libradius returns the error via the rc_auth() function. I notices a bit of an inconsistency in the handling of the retcode of rc_auth() - old stuff :). You say you see a REJECT and the RADIUS level, right ? Could you please try the attached patch and see if the failures are properly reported to the script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 06.06.2014 11:27, dpa wrote: Hello Bogdan What I can see in logs during test call ERROR:aaa_radius:send_auth_func: radius authentication message failed with ERROR From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, June 05, 2014 6:20 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Hi, -1 ret code means some internal error - check the opensips logs for any error. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03.06.2014 07:01, dpa wrote: Hello Bogdan Sorry for long answer. I changed opensips.cfg just a little .. radius_send_auth(set1,set2); switch($rc){ case -2: xlog (Case -2 in radius route detected); break; case -1: xlog (Case -1 in radius route detected); break; } ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem
Sorry, Bogdan I missed one moment during starting opensips (I used wrong directory for opensips modules) The patch fixed the main problem. Now I see retcode -2. Thank you very much! From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, June 06, 2014 3:38 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Strange... Could you directly add in your modules/aaa_radius/aaa_radius.c file, after line 331 which is: res = rc_auth(rh, SIP_PORT, send, recv, mess); and extra log: LM_ERR(rc_auth returned %d\n,res); Recompile and run again - let's see exactly what the libradius has to return :). Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 06.06.2014 13:00, dpa wrote: Unfortunately it doesn`t help. I tried the patch (make command I gave from module/aaa_radius directory). A test call shows Jun 6 13:57:42 opensips-mirror /usr/local/opensipsdev/sbin/opensips[4297]: ERROR:aaa_radius:send_auth_func: radius authentication message failed with ERROR Jun 6 13:57:42 opensips-mirror /usr/local/opensipsdev/sbin/opensips[4297]: Case -1 in radius route detected From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, June 06, 2014 1:21 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Hi, The libradius returns the error via the rc_auth() function. I notices a bit of an inconsistency in the handling of the retcode of rc_auth() - old stuff :). You say you see a REJECT and the RADIUS level, right ? Could you please try the attached patch and see if the failures are properly reported to the script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 06.06.2014 11:27, dpa wrote: Hello Bogdan What I can see in logs during test call ERROR:aaa_radius:send_auth_func: radius authentication message failed with ERROR From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, June 05, 2014 6:20 PM To: dpa; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Hi, -1 ret code means some internal error - check the opensips logs for any error. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03.06.2014 07:01, dpa wrote: Hello Bogdan Sorry for long answer. I changed opensips.cfg just a little .. radius_send_auth(set1,set2); switch($rc){ case -2: xlog (Case -2 in radius route detected); break; case -1: xlog (Case -1 in radius route detected); break; } ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem
Hello Bogdan Sorry for long answer. I changed opensips.cfg just a little .. radius_send_auth(set1,set2); switch($rc){ case -2: xlog (Case -2 in radius route detected); break; case -1: xlog (Case -1 in radius route detected); break; } .. In log file I see Jun 3 07:19:13 opensips-mirror /usr/local/opensipsdev/sbin/opensips[14956]: Case -1 in radius route detected Meanwhile tcpdump shows that radius server answers Rejected on the request. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, May 29, 2014 11:26 AM To: OpenSIPS users mailling list; Denis Putyato Subject: Re: [OpenSIPS-Users] Opensips 1.10.1 and radius problem Hi, On each case, please an xlog to print something from the script - in this way you can double check which way your script went. Or, you can use the script_trace() function to do that . See: http://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc42 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 29.05.2014 09:22, dpa wrote: Hello! I have such configuration in opensips.cfg .. radius_send_auth(set1,set2); switch($rc){ case -2: exec_msg(echo '$avp(500) $rU $time(%c) rejected' /opensips/alarmradius.txt); break; case -1: exec_msg(echo '$avp(500) $rU $time(%c) radius' /opensips/alarmradius.txt); break; } . In attachment you can see communication of Opensips with radius server during making some test call (1.1.1.1 - Opensips, 2.2.2.2 - radius server) After making a test call i see in alarmradius.txt record with radius word. The question is why do I see radius word (and it means that Opensips detected retcode -1) but not rejected (retcode -2) word? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.10.1 and radius problem
Hello! I have such configuration in opensips.cfg :. radius_send_auth(set1,set2); switch($rc){ case -2: exec_msg(echo '$avp(500) $rU $time(%c) rejected' /opensips/alarmradius.txt); break; case -1: exec_msg(echo '$avp(500) $rU $time(%c) radius' /opensips/alarmradius.txt); break; } : In attachment you can see communication of Opensips with radius server during making some test call (1.1.1.1 - Opensips, 2.2.2.2 - radius server) After making a test call i see in alarmradius.txt record with radius word. The question is why do I see radius word (and it means that Opensips detected retcode -1) but not rejected (retcode -2) word? Thank you for any help. No. TimeSourceDestination Protocol Length Info 3 3.2684971.1.1.12.2.2.2 RADIUS 113 Access-Request(1) (id=50, l=69) Frame 3: 113 bytes on wire (904 bits), 113 bytes captured (904 bits) Linux cooked capture Internet Protocol Version 4, Src: 1.1.1.1 (1.1.1.1), Dst: 2.2.2.2 (2.2.2.2) User Datagram Protocol, Src Port: 42580 (42580), Dst Port: radius (1812) Radius Protocol Code: Access-Request (1) Packet identifier: 0x32 (50) Length: 69 Authenticator: 9283df0bc7566891b5103fdcb78ae86d [The response to this request is in frame 4] Attribute Value Pairs AVP: l=13 t=Called-Station-Id(30): 89213039240 AVP: l=12 t=Calling-Station-Id(31): 8612033848 AVP: l=12 t=User-Name(1): 8612033848 AVP: l=6 t=NAS-Port(5): 5060 AVP: l=6 t=NAS-IP-Address(4): 1.1.1.1 No. TimeSourceDestination Protocol Length Info 4 4.2755562.2.2.2 1.1.1.1 RADIUS 64 Access-Reject(3) (id=50, l=20) Frame 4: 64 bytes on wire (512 bits), 64 bytes captured (512 bits) Linux cooked capture Internet Protocol Version 4, Src: 2.2.2.2 (2.2.2.2), Dst: 1.1.1.1 (1.1.1.1) User Datagram Protocol, Src Port: radius (1812), Dst Port: 42580 (42580) Radius Protocol Code: Access-Reject (3) Packet identifier: 0x32 (50) Length: 20 Authenticator: a41bae3fb27b20d439161eb531fd74a9 [This is a response to a request in frame 3] [Time from request: 1.007059000 seconds] ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.10.1 stopped to process SIP messages
Hello! Today there was one problem. Opensips stopped to process SIP messages during about 2 minutes. Log analyses of problem`s time period shows some unusual messages ERROR:core:udp_send: sendto(sock,0x7fd068403ad0,35904,0,0x7fff63d1ddd0,16): No buffer space available(105) ERROR:aaa_radius:send_auth_func: radius authentication message failed with TIMEOUT rc_send_server: no reply from RADIUS server unknown:1812 rc_ip_hostname: couldn't look up host by addr: D5AA5098 Opensips during call processing interacts with radius server (and there was some problem with radius server during problem`s time period). It was done for prepaid scheme (although not all clients use prepaid) More During problem`s time period there were a lot of messages BYE has been sent because of dialog lifetime = 10 It`s my own. Xlog (function located in local route of the opensips.cfg) generates this message if Opensips terminates dialog for some reason. lifetime = 10 means that modparam(dialog, default_timeout, 10) expired and there were no ACK on 200 OK. And ngrep shows that 200 OK had not translated to caller by Opensips during problem`s time period. After radius server UP ( and rc_send_server: no reply from RADIUS server unknown:1812 and rc_ip_hostname: couldn't look up host by addr: D5AA5098 disappeared) Opensips became work normally. What can it be? And how can I avoid such problem in future? Thank you for any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.9.1 crash
Hello Bogdan OK, I will try upgrading to 1.10 From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, April 01, 2014 9:42 PM To: OpenSIPS users mailling list; Denis Putyato Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 crash Hello, 1.9 is at the end of its life span - it became unmaintained starting with the 1.11 release (see http://www.opensips.org/About/AvailableVersions). I strongly suggest upgrading to 1.10 , as you may experience problems which were already fixed in the maintained versions. Is this an option for you ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 01.04.2014 07:30, dpa wrote: Hello! I had two times Opensips crashed during one week. On this link http://files.mail.ru/56128EED55BD49219EC1362E7161747A you can find backtrace from core file for the first time. On this link http://files.mail.ru/793A146CF453427893E2A66DF096A41E you can find information from the log file. Unfortunately in this case no core file had been generated. The problem is very critical. Thank you for any help! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.9.1 crash
Hello! I had two times Opensips crashed during one week. On this link http://files.mail.ru/56128EED55BD49219EC1362E7161747A you can find backtrace from core file for the first time. On this link http://files.mail.ru/793A146CF453427893E2A66DF096A41E you can find information from the log file. Unfortunately in this case no core file had been generated. The problem is very critical. Thank you for any help! image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.9.1 problem
Hello Bogdan Unfortunaly there were no more messages in log. Only what I sent. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, March 27, 2014 12:34 PM To: OpenSIPS users mailling list; Denis Putyato Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 problem Hello, Looks like you get HTTP traffic on your SIP/TCP ports - and opensips gives a pars error (as expecting SIP, not HTTP syntax). The critical error from the tm-timer indicates an huge amount of time spent in the TM timer (process 32563) - but I see not previous errors from this process to indicate what is going on there. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 25.03.2014 15:02, dpa wrote: Hello! In attachment there is some messages from log file. What does it mean? These messages appeared in log after Opensips cannot process SIP messages during about 1 minute. Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] An Error in Opensips 1.9.1
Hello Bogdan It will be super, because I spent a much time for finding this:))) P.S. Of course not ($avp(63==1)) but ($avp(63)==1) From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, March 27, 2014 12:37 PM To: OpenSIPS users mailling list; Denis Putyato Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 Hello Denis, Indeed, we need to improve a bit that error message, to make it more human readable and less cryptic :). I will take care of that. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 25.03.2014 14:04, dpa wrote: No problem In opensips.cfg there were one string If ($avp(63)==1) {.. So after I changed the string to f ($avp(63==1)) {... errors disappear from the log From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Tuesday, March 25, 2014 3:55 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 It would be nice if you give us the problem and the solution. On Mar 25, 2014 1:49 PM, dpa denis7...@mail.ru wrote: It seems I found the problem myself. Thank you From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Tuesday, March 25, 2014 7:49 AM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 Hello I am ready. But what type of information do I need provide? To tell the truth I rarely look into opensips log file. Only when some critical case happens (opensips crash for example). Yesterday there was such case. Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of qasimak...@gmail.com Sent: Monday, March 24, 2014 8:29 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 I think a little more information than this would be required if you need help :). Regards, Qasim On Mon, Mar 24, 2014 at 4:36 PM, dpa denis7...@mail.ru wrote: Hello! 1. In log file I see many errors CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!! WARNING:core:do_action: error in expression (l=662) As I understand there is a problem somewhere in opensips.cfg How can I understand where problem is? 2. I have modparam(dialog, db_mode, 1) in opensips.cfg. In normal operation of opensips I can see in statistics, for example, 800 active dialogs, but after restart I see much more active dialogs. Why can it be? Thank you for any help. ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.9.1 accounting
Hello! In attachment you can find ordinary call (not successful) 1.1.1.1- SIP UA 1.1.1.2- Opensips 1.1.1.3- SIP UA In opensips.cfg modparam(acc, early_media, 0) modparam(acc, report_cancels, 1) modparam(acc, detect_direction, 1) modparam(acc, db_flag, 15) modparam(acc, db_missed_flag, 16) modparam(acc, failed_transaction_flag, 17) modparam(acc, db_table_acc, acc) modparam(acc, db_table_missed_calls, acc) modparam(acc, cdr_flag, 22) and before INVITE will be translated to callee SIP UA setflag(15); setflag(16); setflag(17); setflag(22); I see that Opensips tried to insert several entries into acc. The question is, why did Opensips try to insert into acc several entries due to one call? Is this because of db_flag and failed_transaction_flag? Thank you for any help. image001.gifU 2014/03/26 10:49:08.556019 1.1.1.1:53876 - 1.1.1.2:5060 INVITE sip:3364399@1.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B. From: sip:8123364283@1.1.1.1;tag=32C4640-18B3. To: sip:3364399@1.1.1.2. Date: Wed, 26 Mar 2014 06:49:08 GMT. Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2426479216-3018396131-3064463394-2438468884. User-Agent: Cisco-SIPGateway/IOS-12.x. Accept-Language: ru. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 15. Timestamp: 1395816548. Contact: sip:8123364283@1.1.1.1:5060. Expires: 100. Allow-Events: telephone-event. P-Asserted-Identity: sip:8123364283@1.1.1.1. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 310. . v=0. o=CiscoSystemsSIP-GW-UserAgent 4030 1274 IN IP4 1.1.1.1. s=SIP Call. c=IN IP4 1.1.1.1. t=0 0. m=audio 18038 RTP/AVP 8 0 18 101. c=IN IP4 1.1.1.1. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2014/03/26 10:49:08.557516 1.1.1.2:5060 - 1.1.1.1:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B. From: sip:8123364283@1.1.1.1;tag=32C4640-18B3. To: sip:3364399@1.1.1.2. Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1. CSeq: 101 INVITE. Content-Length: 0. . U 2014/03/26 10:49:08.566102 1.1.1.2:5060 - 1.1.1.3:5060 INVITE sip:78123364399@1.1.1.3:5060 SIP/2.0. Record-Route: sip:1.1.1.2;lr;ftag=32C4640-18B3;did=8de.cf20f431. Via: SIP/2.0/UDP 1.1.1.2:5060;branch=z9hG4bKcad3.e39f27b7.0. Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B. From: sip:8123364283@1.1.1.1;tag=32C4640-18B3. To: sip:3364399@1.1.1.2. Date: Wed, 26 Mar 2014 06:49:08 GMT. Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2426479216-3018396131-3064463394-2438468884. User-Agent: Cisco-SIPGateway/IOS-12.x. Accept-Language: ru. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 15. Timestamp: 1395816548. Contact: sip:8123364283@1.1.1.1:5060. Remote-Party-ID:sip:8123364283@1.1.1.1;party=calling;screen=yes;privacy=off. Expires: 100. Allow-Events: telephone-event. P-Asserted-Identity: sip:8123364283@1.1.1.1. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 310. . v=0. o=CiscoSystemsSIP-GW-UserAgent 4030 1274 IN IP4 1.1.1.1. s=SIP Call. c=IN IP4 1.1.1.1. t=0 0. m=audio 18038 RTP/AVP 8 0 18 101. c=IN IP4 1.1.1.1. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2014/03/26 10:49:08.573798 1.1.1.3:5060 - 1.1.1.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.2:5060;branch=z9hG4bKcad3.e39f27b7.0. Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B. From: sip:8123364283@1.1.1.1;tag=32C4640-18B3. To: sip:3364399@1.1.1.2. Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1. CSeq: 101 INVITE. Server: Telphin SoftSwitch. Content-Length: 0. . U 2014/03/26 10:49:08.576295 1.1.1.3:5060 - 1.1.1.2:5060 SIP/2.0 603 Declined. Via: SIP/2.0/UDP 1.1.1.2:5060;branch=z9hG4bKcad3.e39f27b7.0. Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bK448B21124B. From: sip:8123364283@1.1.1.1;tag=32C4640-18B3. To: sip:3364399@1.1.1.2;tag=as26700a4a. Call-ID: 90A25AC0-B3E911E3-B077DBBA-3DF918A8@1.1.1.1. CSeq: 101 INVITE. Server: Telphin MediaServer. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. . U 2014/03/26 10:49:08.576787 1.1.1.2:5060 - 1.1.1.3:5060 ACK sip:78123364399@1.1.1.3:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.2:5060;branch=z9hG4bKcad3.e39f27b7.0. From: sip:8123364283@1.1.1.1;tag=32C4640-18B3. Call-ID:
Re: [OpenSIPS-Users] An Error in Opensips 1.9.1
It seems I found the problem myself. Thank you From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Tuesday, March 25, 2014 7:49 AM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 Hello I am ready. But what type of information do I need provide? To tell the truth I rarely look into opensips log file. Only when some critical case happens (opensips crash for example). Yesterday there was such case. Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of qasimak...@gmail.com Sent: Monday, March 24, 2014 8:29 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 I think a little more information than this would be required if you need help :). Regards, Qasim On Mon, Mar 24, 2014 at 4:36 PM, dpa denis7...@mail.ru wrote: Hello! 1. In log file I see many errors CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!! WARNING:core:do_action: error in expression (l=662) As I understand there is a problem somewhere in opensips.cfg How can I understand where problem is? 2. I have modparam(dialog, db_mode, 1) in opensips.cfg. In normal operation of opensips I can see in statistics, for example, 800 active dialogs, but after restart I see much more active dialogs. Why can it be? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] An Error in Opensips 1.9.1
No problem In opensips.cfg there were one string If ($avp(63)==1) {.. So after I changed the string to f ($avp(63==1)) {... errors disappear from the log From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Tuesday, March 25, 2014 3:55 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 It would be nice if you give us the problem and the solution. On Mar 25, 2014 1:49 PM, dpa denis7...@mail.ru wrote: It seems I found the problem myself. Thank you From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Tuesday, March 25, 2014 7:49 AM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 Hello I am ready. But what type of information do I need provide? To tell the truth I rarely look into opensips log file. Only when some critical case happens (opensips crash for example). Yesterday there was such case. Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of qasimak...@gmail.com Sent: Monday, March 24, 2014 8:29 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 I think a little more information than this would be required if you need help :). Regards, Qasim On Mon, Mar 24, 2014 at 4:36 PM, dpa denis7...@mail.ru wrote: Hello! 1. In log file I see many errors CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!! WARNING:core:do_action: error in expression (l=662) As I understand there is a problem somewhere in opensips.cfg How can I understand where problem is? 2. I have modparam(dialog, db_mode, 1) in opensips.cfg. In normal operation of opensips I can see in statistics, for example, 800 active dialogs, but after restart I see much more active dialogs. Why can it be? Thank you for any help. ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] An Error in Opensips 1.9.1
Hello! 1. In log file I see many errors CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!! WARNING:core:do_action: error in expression (l=662) As I understand there is a problem somewhere in opensips.cfg How can I understand where problem is? 2. I have modparam(dialog, db_mode, 1) in opensips.cfg. In normal operation of opensips I can see in statistics, for example, 800 active dialogs, but after restart I see much more active dialogs. Why can it be? Thank you for any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] An Error in Opensips 1.9.1
Hello I am ready. But what type of information do I need provide? To tell the truth I rarely look into opensips log file. Only when some critical case happens (opensips crash for example). Yesterday there was such case. Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of qasimak...@gmail.com Sent: Monday, March 24, 2014 8:29 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] An Error in Opensips 1.9.1 I think a little more information than this would be required if you need help :). Regards, Qasim On Mon, Mar 24, 2014 at 4:36 PM, dpa denis7...@mail.ru wrote: Hello! 1. In log file I see many errors CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!! WARNING:core:do_action: error in expression (l=662) As I understand there is a problem somewhere in opensips.cfg How can I understand where problem is? 2. I have modparam(dialog, db_mode, 1) in opensips.cfg. In normal operation of opensips I can see in statistics, for example, 800 active dialogs, but after restart I see much more active dialogs. Why can it be? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.9.1 and REGISTER process
Hello Bogdan, Yes E flags is working. But without in doesn`t From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, March 05, 2014 9:56 PM To: OpenSIPS users mailling list; Denis Putyato Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and REGISTER process Hello Denis, Have you tried to use the E option in the save() function? see: http://www.opensips.org/html/docs/modules/1.9.x/registrar.html#id250454 It should have the same effect (setting the max expire), but is per REGISTER bases. Just to see if this works for you. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03.03.2014 15:39, dpa wrote: Hello! There is one question. A little part of opensips.cfg .. modparam(registrar, default_expires, 60) modparam(registrar, max_expires, 60) modparam(registrar, min_expires, 0) ... If I enter register timeout on my SIP UA to 1600, for example, Opensips will return to SIP UA 1600 timeout. In 1.6.4-2 there were no problem with it. If I enter 1600 timeout Opensips returned 60 and after 60 s there was another attempt to register to Opensips. What did I miss? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.9.1 and REGISTER process
Hello! There is one question. A little part of opensips.cfg .. modparam(registrar, default_expires, 60) modparam(registrar, max_expires, 60) modparam(registrar, min_expires, 0) ... If I enter register timeout on my SIP UA to 1600, for example, Opensips will return to SIP UA 1600 timeout. In 1.6.4-2 there were no problem with it. If I enter 1600 timeout Opensips returned 60 and after 60 s there was another attempt to register to Opensips. What did I miss? Thank you for any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.9.1 and dialog process
Hello Early I used opensips1.6.4.-2. There were some fields (hash_entry and hash id) in DB scheme of dialog module. I used this field to end dialog. Now, in 1.9.1, the DB scheme of dialog module has been changed and there are no fields hash_entry and hash id in шею But there is the dlg_id field. The question is, How can I bind dlg_id in mysql with hash entry and hash_id for dialog ending. Thank you for any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.9.1 and opensips_fifo file
Unfortunately I cannot tell which commands leave these entries. Anyway thank you for help. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu Sent: Friday, December 20, 2013 6:03 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and opensips_fifo file Are you able to tell which opensipsctl commands leave undeleted entries in your /tmp directory? In any case, OpenSIPS will work if you manually delete them, although this is not normal :) On 12/20/2013 03:54 PM, dpa wrote: Hello Sorry, but I don`t quite understand to replicate this issue? Where to replicate? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu Sent: Friday, December 20, 2013 5:17 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and opensips_fifo file Hello, That is actually a named pipe and it is opened by opensipsctl itself with each command, in order to receive a reply back from OpenSIPS. Normally, they should be automatically cleaned up. Are you able to replicate this issue? Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 12/20/2013 06:17 AM, dpa wrote: Hello I made a decision to locate all files associated with Opensips to one directory (say /sock/). This package of files includes opensps_fifo and opensips.pid files. I made an appropriate correction in opensips.cfg and opensipsctlrc. After restart Opensips everything works fine. Fifo commands is working. But I found that in /tmp/ directory became appear some files. The name of this files: opensips_receiver_some nubers. These files have 0 size. The question is what is it? If I delete them, Will Opensips still work? Thank you. ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.9.1 and opensips_fifo file
Hello I made a decision to locate all files associated with Opensips to one directory (say /sock/). This package of files includes opensps_fifo and opensips.pid files. I made an appropriate correction in opensips.cfg and opensipsctlrc. After restart Opensips everything works fine. Fifo commands is working. But I found that in /tmp/ directory became appear some files. The name of this files: opensips_receiver_some nubers. These files have 0 size. The question is what is it? If I delete them, Will Opensips still work? Thank you. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT
Hello I understand but in onreply route I make a test: nat_uac_test(55) and only if it successful I make fix_nated_contact(). In my case nat_uac_test(55) must be fail after checking 183 ringing and 200 OK. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey Sent: Friday, November 22, 2013 5:37 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT Hello, The question is not quite clear. In your opensips.cfg you call fix_nated_contact() on both route and reply route. that's why it changes the route. There is no such a thing as direction in opensips unless it's implemented in your logic. A message goes through your route or reply route scripts when it reaches opensips. Regards, Ali On Fri, Nov 22, 2013 at 1:10 AM, dpa denis7...@mail.ru wrote: Hello I have a one question about nat processing of Opensips. There is such scheme UAC1 (softphone behind nat) à Opensips - UAC2 (another softswitch), i.e. UAC1 initiates a call to Opensips and a signaling port = 5068. So UAC2 becomes ringing by sending 183 message. In attachment 183 message from UAC2 and some parts of opensips.cfg After 183 processing by Opensips port in Contact header of 183 (and later 200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact header. So my question is, Why does Opensips changes Contact header? Once Opensips detects nat transaction (setting setflag(21)) is it check all reply messages (and doesn`t matter from which UAC they have been received) or Opensips can detect direction and makes decision about nat process? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users https://contactmonkey.com/api/v1/tracker?cm_session=7aa93c28-7e0d-4b18-9de6 -8fd4319e3154cm_type=opencm_user_email=ali...@gmail.com image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT
OK, but in the current case another question. Is nat_bflag appear in transaction when call goes TO uac behind nat (by using lookup(), for example), or it appears in transaction when call goes FROM uac behind nat? Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey Sent: Monday, November 25, 2013 8:38 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT You also examining isbflagset(10). Isn't that set? Regards, Ali Pey On Mon, Nov 25, 2013 at 3:09 AM, dpa denis7...@mail.ru wrote: Hello I understand but in onreply route I make a test: nat_uac_test(55) and only if it successful I make fix_nated_contact(). In my case nat_uac_test(55) must be fail after checking 183 ringing and 200 OK. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey Sent: Friday, November 22, 2013 5:37 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT Hello, The question is not quite clear. In your opensips.cfg you call fix_nated_contact() on both route and reply route. that's why it changes the route. There is no such a thing as direction in opensips unless it's implemented in your logic. A message goes through your route or reply route scripts when it reaches opensips. Regards, Ali On Fri, Nov 22, 2013 at 1:10 AM, dpa denis7...@mail.ru wrote: Hello I have a one question about nat processing of Opensips. There is such scheme UAC1 (softphone behind nat) à Opensips - UAC2 (another softswitch), i.e. UAC1 initiates a call to Opensips and a signaling port = 5068. So UAC2 becomes ringing by sending 183 message. In attachment 183 message from UAC2 and some parts of opensips.cfg After 183 processing by Opensips port in Contact header of 183 (and later 200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact header. So my question is, Why does Opensips changes Contact header? Once Opensips detects nat transaction (setting setflag(21)) is it check all reply messages (and doesn`t matter from which UAC they have been received) or Opensips can detect direction and makes decision about nat process? Thank you for any help. ___ 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 https://contactmonkey.com/api/v1/tracker?cm_session=33dfe8c1-fb1c-4d59-ae13 -f60c41bb3592cm_type=opencm_user_email=ali...@gmail.com image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT
What is it that you are trying to do? I try to check if message came from behind NAT or not and make some actions depending on result of checking. To speak the truth I tried to include in opensips.cfg logic about direction of the call. What I thought. I thought that nat_bflag activated only after lookup() function processing, i.e. when call goes TO UAC behind NAT and NOT activated when call came FROM UAC behind NAT. So in onreply route I make the test about this flag and waiting (in my current case) that the test of natb_flag will be failed. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey Sent: Tuesday, November 26, 2013 8:16 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT As I said earlier, there is no such a thing as FROM or TO (i.e. direction) unless it's implemented in your logic. When a SIP request reaches the IP address of your opensips server, it goes through your routing script in opensips.cfg and then it's routed out. What you have there, it would apply to the request message. Put yourself in opensips shoes and follow the logic - that's how we debugged code while at uni :) If you have a concept of direction and you need different behavior based on your concept of direction, you need to implement it in your logic. For instance you can examine the source IP or subnet to decide if this message is from internal or external and then apply different logic to it - or whatever else that is specific to your environment. What is it that you are trying to do? Regards, Ali Pey On Mon, Nov 25, 2013 at 10:52 PM, dpa denis7...@mail.ru wrote: OK, but in the current case another question. Is nat_bflag appear in transaction when call goes TO uac behind nat (by using lookup(), for example), or it appears in transaction when call goes FROM uac behind nat? Thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey Sent: Monday, November 25, 2013 8:38 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT You also examining isbflagset(10). Isn't that set? Regards, Ali Pey On Mon, Nov 25, 2013 at 3:09 AM, dpa denis7...@mail.ru wrote: Hello I understand but in onreply route I make a test: nat_uac_test(55) and only if it successful I make fix_nated_contact(). In my case nat_uac_test(55) must be fail after checking 183 ringing and 200 OK. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey Sent: Friday, November 22, 2013 5:37 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT Hello, The question is not quite clear. In your opensips.cfg you call fix_nated_contact() on both route and reply route. that's why it changes the route. There is no such a thing as direction in opensips unless it's implemented in your logic. A message goes through your route or reply route scripts when it reaches opensips. Regards, Ali On Fri, Nov 22, 2013 at 1:10 AM, dpa denis7...@mail.ru wrote: Hello I have a one question about nat processing of Opensips. There is such scheme UAC1 (softphone behind nat) à Opensips - UAC2 (another softswitch), i.e. UAC1 initiates a call to Opensips and a signaling port = 5068. So UAC2 becomes ringing by sending 183 message. In attachment 183 message from UAC2 and some parts of opensips.cfg After 183 processing by Opensips port in Contact header of 183 (and later 200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact header. So my question is, Why does Opensips changes Contact header? Once Opensips detects nat transaction (setting setflag(21)) is it check all reply messages (and doesn`t matter from which UAC they have been received) or Opensips can detect direction and makes decision about nat process? Thank you for any help. ___ 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 https://contactmonkey.com/api/v1/tracker?cm_session=f72377a2-cbd9-408f-98ea -b3dc9fc53d76cm_type=opencm_user_email=ali...@gmail.com image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.9.1 and NAT
Hello I have a one question about nat processing of Opensips. There is such scheme UAC1 (softphone behind nat) à Opensips - UAC2 (another softswitch), i.e. UAC1 initiates a call to Opensips and a signaling port = 5068. So UAC2 becomes ringing by sending 183 message. In attachment 183 message from UAC2 and some parts of opensips.cfg After 183 processing by Opensips port in Contact header of 183 (and later 200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact header. So my question is, Why does Opensips changes Contact header? Once Opensips detects nat transaction (setting setflag(21)) is it check all reply messages (and doesn`t matter from which UAC they have been received) or Opensips can detect direction and makes decision about nat process? Thank you for any help. image001.gifU 2013/11/22 08:56:58.606648 2.2.2.2:5060 - 1.1.1.1:5068 SIP/2.0 183 Progress. Via: SIP/2.0/UDP 1.1.1.1:5068;branch=z9hG4bK8b6b.d794d636.0. Via: SIP/2.0/UDP 192.168.18.150:41314;rport=43238;received=3.3.3.3;branch=z9hG4bK-d8754z-37977264babb1a1c-1---d8754z-. Record-Route: sip:AQEAEBU0TSMfDQMFY3/lbBsPg80DAAQUm+/P@2.2.2.2;lr. Record-Route: sip:AQEAEL8TN9Pcghi+IDj0gLMRnv4DAAR5OlYR@2.2.2.2;lr. Record-Route: sip:1.1.1.1:5068;lr;ftag=9b44a312;did=486.b9e72ff6. From: sip:8172264014@1.1.1.1:5068;transport=UDP;tag=9b44a312. To: sip:89213039240@1.1.1.1:5068;transport=UDP;tag=8maN3eG8u*ECE8XhvLvxAhPEyZHydAAq. Call-ID: YWZlNGYwMTcwNmMxZjE3MGYwNGUxNGQ3NzY4MTEwZTU.. CSeq: 2 INVITE. Contact: sip:79213039240@2.2.2.2:5061;transport=UDP. Server: TS-v4.6.0-09d. Content-Length: 0. 2.2.2.2 - UAC2 1.1.1.1 - Opensips 3.3.3.3 - NAT Opensips.cfg *** modparam(usrloc, nat_bflag, 10) Route [2] { if ($Rp==5068) { if (nat_uac_test(55)) { fix_nated_contact(); setflag(21); } } } onreply_route[1] { . if (isbflagset(10) || (isflagset(21) nat_uac_test(55))) fix_nated_contact(); . } ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] rtpproxy logging shut off
Hello! I am using rtpproxy with Opensips for SIP connections. Now rtpproxy makes a log to syslog during call establishment. The question is may I shut off this process for rtpproxy working without logging? Thank you from any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] rtpproxy logging shut off
I understand, thank you -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of kamika Sent: Wednesday, September 25, 2013 3:48 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpproxy logging shut off you have to change options that you start RTPProxy with. See -d flag e.g.: -d CRIT:LOG_LOCAL5 -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/rtpproxy-logging-shut- off-tp7587893p7587894.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] rtpproxy logging shut off
Hello I made the change (-d INFO:LOG_LOCAL5) but log from rtpproxy is still in syslog.1 file. In config file of syslog there is no mentioned about local7 log_facility. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of kamika Sent: Wednesday, September 25, 2013 3:48 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpproxy logging shut off you have to change options that you start RTPProxy with. See -d flag e.g.: -d CRIT:LOG_LOCAL5 -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/rtpproxy-logging-shut- off-tp7587893p7587894.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
[OpenSIPS-Users] Problem with DB migration from 1.8.2 to 1.9.1
Hello! Now I am using Opensips 1.6.4-2 but decide to upgrade ones to 1.9.1 First of all I try to migrate MySQL DB. Migrate from 1.6.4.2 to 1.7.0 - OK Migrate from 1.7.0 to 1.8.2 - OK Migrate from 1.8.2 to 1.9.1 - OK except one table dr_carriers During migration process from 1.8 to 1.9 i see such error ERROR: failed to migrate opensips8.dr_carriers to opensips9.dr_carriers!!! Skip it and continue (y/n)? Other tables had been migrated successfully Thank you for any help image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Database Record
acc_db_request inserts information in a table that you mentioned in second parameter of the function with some comments mentioned in first parameter. For example, acc_db_request(some information, acc). The function works only in request and failure route. If you want put some information in acc table before call hang up then you should make acc_db_request(some information, acc) in any request route where INVITE processed. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab Sent: Wednesday, May 08, 2013 2:27 PM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Database Record version: opensips 1.8.2-notls (x86_64/linux) flags: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:9954M @(#) $Id: main.c 9791 2013-02-15 10:15:25Z bogdan_iancu $ main.c compiled on 15:29:28 Apr 18 2013 with gcc 4.4.5 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Wednesday, May 08, 2013 7:48 AM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Database Record The version of Opensips? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab Sent: Tuesday, May 07, 2013 4:52 PM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Database Record Hello, Please can you make it more clear or give me an example for the function acc_log_request(Some comment, Some table); onreply_route[2] { if (is_method(INVITE) t_check_status(200) ) { acc_log_request(coonect time ,acc); ??? } From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Tuesday, May 07, 2013 1:06 PM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Database Record Hello Try http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id293991 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab Sent: Tuesday, May 07, 2013 2:01 PM To: users@lists.opensips.org Subject: [OpenSIPS-Users] Database Record Dears, I am using acc module and I am inserting two rows for each call (INVITE,BYE) , I have a problem such I want to update the record in acc table before the calls ends which have a method = INVITE, and now that cant be done now since opensips Does not insert the invite record before the calls ends How to let the invite record inserted in database before calls ends Please advice Khaled Chehab Senior NGN Engineer Description: icucall Operations Office - Lebanon Office: +961 1 515155 ext 300 Mobile : +961 3 045212 E-mail: kche...@icucall.com MSN ID :khalidche...@hotmail.com Skype: k_chehab Web Site: http://www.icucall.com http://www.allohi.com image001.png___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Database Record
The version of Opensips? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab Sent: Tuesday, May 07, 2013 4:52 PM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Database Record Hello, Please can you make it more clear or give me an example for the function acc_log_request(Some comment, Some table); onreply_route[2] { if (is_method(INVITE) t_check_status(200) ) { acc_log_request(coonect time ,acc); ??? } From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Tuesday, May 07, 2013 1:06 PM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Database Record Hello Try http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id293991 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab Sent: Tuesday, May 07, 2013 2:01 PM To: users@lists.opensips.org Subject: [OpenSIPS-Users] Database Record Dears, I am using acc module and I am inserting two rows for each call (INVITE,BYE) , I have a problem such I want to update the record in acc table before the calls ends which have a method = INVITE, and now that cant be done now since opensips Does not insert the invite record before the calls ends How to let the invite record inserted in database before calls ends Please advice Khaled Chehab Senior NGN Engineer Description: icucall Operations Office - Lebanon Office: +961 1 515155 ext 300 Mobile : +961 3 045212 E-mail: kche...@icucall.com MSN ID :khalidche...@hotmail.com Skype: k_chehab Web Site: http://www.icucall.com http://www.allohi.com image001.png___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.6.4-2 call handling problem
Hello! There is Opensips 1.6.4-2. Today I found a problem that Opensips hasn`t been responded to any SIP message that has been sent to it for a sometime. I began to learn log file and found such messages: rc_ip_hostname: couldn't look up host by addr: xx May 6 09:50:17 opensips-main /usr/local/opensips1.6.4-2/sbin/opensips[2146]: rc_send_server: no reply from RADIUS server unknown:1812 May 6 09:50:17 opensips-main /usr/local/opensips1.6.4-2/sbin/opensips[2146]: ERROR:aaa_radius:send_auth_func: radius authentication message failed with TIMEOUT These messages appeared in log file somewhere about at that time while Opensips hasn`t been responded to SIP messages. The fact is I am using radius for prepaid scheme which I apply to some subscribers (but not for all). May be it is just coincidence but my question is. May radius problem affect normal Opensips working? Thank you for any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg
Hello Yes the problem was there. In local route I had exec function to do dlg_end_dlg. Thank you very much. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, December 20, 2012 10:40 AM To: dpa Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg yes, it is, as when you do dlg_end_dlg, opensips will generate and send 2 BYEs (one to caller, one to callee); and each time opensips sends a locally generated request, local route is triggered. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/20/2012 10:34 AM, dpa wrote: Bogdan, as I understand when I am doing fifo dlg_end_dlg through FIFO command from console the config of local route is activated? From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, December 20, 2012 9:57 AM To: dpa; users@lists.opensips.org Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Hi, You are doing something wrong there - based on the script, it seems you do dlg_end_dlg, which triggers sending BYE - local route - you do an exec there to do again dlg_end_dlg. So you try to do nested FIFO commands which does not work - FIFO engine will not read a new command until the current is not finished. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/20/2012 07:36 AM, dpa wrote: Hello Bogdan The gdb information in attachment. Unfortunately for some unknown reason I cannot see logs. I will check the problem and write later. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 11:17 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Try to set debug=4 and see the logs from the MI fifo process when running the dlg_end_dlg command. Also, try the get a backtrace from the MO FIFO proc after blocking. For that, before the dlg command, do opensipsctl fifo ps to find out the PID of the FIFO process. After running the dlg command and blocking, use gdb /path/to/opensips FIFO_PID and run bt inside - these will give some ideas on where the process blocked. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 10:07 AM, dpa wrote: So only the MI fifo part blocks? As I understand, yes. do you see any errors in the logs? I see only INFO:core:buf_init: initializing... in log file also, does it recover after some time (like minutes) ? To speak the truth I didn`t wait recovering. I used Ctrl+Z to stop process of dlg_end_dlg and tried another fifo command. ps -A show me that opensipsctl process appears. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 9:44 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg So only the MI fifo part blocks ? do you see any errors in the logs? also, does it recover after some time (like minutes) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 09:42 AM, dpa wrote: Hello Bogdan it simply blocks and does not answer I can make calls. But no fifo command available. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 9:39 AM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Hi, Does your opensips actually crashes when you run the dlg_end_dlg command ? or it simply blocks and does not answer ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 07:37 AM, dpa wrote: Hello! I discovered a problem with using dlg_end_dlg fifo command. When I try to make it, dialog hasn't been ended and I cannot see in console that command has been finished success. In /tmp directory (where opensips_fifo is located) I found some new files opensips_receiver_x, where x - some numbers. Ls -la shows that files were generated when I make dlg_end_dlg command. Moreover after this I cannot make any fifo command. After opensips restart I can make any fifo command but until dlg_end_dlg. There were no problems with it but recently I decided to make opensips generates core file if it crashes. I use this documentation http://www.opensips.org/Resources/DocsTsCrash. I used -w option and make changes in /proc/sys/kernel/core_pattern. May be some problems with it? Thank you for any help! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg
Bogdan, as I understand when I am doing fifo dlg_end_dlg through FIFO command from console the config of local route is activated? From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, December 20, 2012 9:57 AM To: dpa; users@lists.opensips.org Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Hi, You are doing something wrong there - based on the script, it seems you do dlg_end_dlg, which triggers sending BYE - local route - you do an exec there to do again dlg_end_dlg. So you try to do nested FIFO commands which does not work - FIFO engine will not read a new command until the current is not finished. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/20/2012 07:36 AM, dpa wrote: Hello Bogdan The gdb information in attachment. Unfortunately for some unknown reason I cannot see logs. I will check the problem and write later. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 11:17 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Try to set debug=4 and see the logs from the MI fifo process when running the dlg_end_dlg command. Also, try the get a backtrace from the MO FIFO proc after blocking. For that, before the dlg command, do opensipsctl fifo ps to find out the PID of the FIFO process. After running the dlg command and blocking, use gdb /path/to/opensips FIFO_PID and run bt inside - these will give some ideas on where the process blocked. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 10:07 AM, dpa wrote: So only the MI fifo part blocks? As I understand, yes. do you see any errors in the logs? I see only INFO:core:buf_init: initializing... in log file also, does it recover after some time (like minutes) ? To speak the truth I didn`t wait recovering. I used Ctrl+Z to stop process of dlg_end_dlg and tried another fifo command. ps -A show me that opensipsctl process appears. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 9:44 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg So only the MI fifo part blocks ? do you see any errors in the logs? also, does it recover after some time (like minutes) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 09:42 AM, dpa wrote: Hello Bogdan it simply blocks and does not answer I can make calls. But no fifo command available. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 9:39 AM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Hi, Does your opensips actually crashes when you run the dlg_end_dlg command ? or it simply blocks and does not answer ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 07:37 AM, dpa wrote: Hello! I discovered a problem with using dlg_end_dlg fifo command. When I try to make it, dialog hasn't been ended and I cannot see in console that command has been finished success. In /tmp directory (where opensips_fifo is located) I found some new files opensips_receiver_x, where x - some numbers. Ls -la shows that files were generated when I make dlg_end_dlg command. Moreover after this I cannot make any fifo command. After opensips restart I can make any fifo command but until dlg_end_dlg. There were no problems with it but recently I decided to make opensips generates core file if it crashes. I use this documentation http://www.opensips.org/Resources/DocsTsCrash. I used -w option and make changes in /proc/sys/kernel/core_pattern. May be some problems with it? Thank you for any help! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg
Hello! I discovered a problem with using dlg_end_dlg fifo command. When I try to make it, dialog hasn't been ended and I cannot see in console that command has been finished success. In /tmp directory (where opensips_fifo is located) I found some new files opensips_receiver_x, where x - some numbers. Ls -la shows that files were generated when I make dlg_end_dlg command. Moreover after this I cannot make any fifo command. After opensips restart I can make any fifo command but until dlg_end_dlg. There were no problems with it but recently I decided to make opensips generates core file if it crashes. I use this documentation http://www.opensips.org/Resources/DocsTsCrash. I used -w option and make changes in /proc/sys/kernel/core_pattern. May be some problems with it? Thank you for any help! image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg
And one more, in log file I can see INFO:core:buf_init: initializing... while I make dlg_end_dlg command From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Wednesday, December 19, 2012 7:38 AM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Hello! I discovered a problem with using dlg_end_dlg fifo command. When I try to make it, dialog hasn't been ended and I cannot see in console that command has been finished success. In /tmp directory (where opensips_fifo is located) I found some new files opensips_receiver_x, where x - some numbers. Ls -la shows that files were generated when I make dlg_end_dlg command. Moreover after this I cannot make any fifo command. After opensips restart I can make any fifo command but until dlg_end_dlg. There were no problems with it but recently I decided to make opensips generates core file if it crashes. I use this documentation http://www.opensips.org/Resources/DocsTsCrash. I used -w option and make changes in /proc/sys/kernel/core_pattern. May be some problems with it? Thank you for any help! image002.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg
Hello Bogdan it simply blocks and does not answer I can make calls. But no fifo command available. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 9:39 AM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Hi, Does your opensips actually crashes when you run the dlg_end_dlg command ? or it simply blocks and does not answer ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 07:37 AM, dpa wrote: Hello! I discovered a problem with using dlg_end_dlg fifo command. When I try to make it, dialog hasn't been ended and I cannot see in console that command has been finished success. In /tmp directory (where opensips_fifo is located) I found some new files opensips_receiver_x, where x - some numbers. Ls -la shows that files were generated when I make dlg_end_dlg command. Moreover after this I cannot make any fifo command. After opensips restart I can make any fifo command but until dlg_end_dlg. There were no problems with it but recently I decided to make opensips generates core file if it crashes. I use this documentation http://www.opensips.org/Resources/DocsTsCrash. I used -w option and make changes in /proc/sys/kernel/core_pattern. May be some problems with it? Thank you for any help! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg
So only the MI fifo part blocks? As I understand, yes. do you see any errors in the logs? I see only INFO:core:buf_init: initializing... in log file also, does it recover after some time (like minutes) ? To speak the truth I didn`t wait recovering. I used Ctrl+Z to stop process of dlg_end_dlg and tried another fifo command. ps -A show me that opensipsctl process appears. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 9:44 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg So only the MI fifo part blocks ? do you see any errors in the logs? also, does it recover after some time (like minutes) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 09:42 AM, dpa wrote: Hello Bogdan it simply blocks and does not answer I can make calls. But no fifo command available. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 9:39 AM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Hi, Does your opensips actually crashes when you run the dlg_end_dlg command ? or it simply blocks and does not answer ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 07:37 AM, dpa wrote: Hello! I discovered a problem with using dlg_end_dlg fifo command. When I try to make it, dialog hasn't been ended and I cannot see in console that command has been finished success. In /tmp directory (where opensips_fifo is located) I found some new files opensips_receiver_x, where x - some numbers. Ls -la shows that files were generated when I make dlg_end_dlg command. Moreover after this I cannot make any fifo command. After opensips restart I can make any fifo command but until dlg_end_dlg. There were no problems with it but recently I decided to make opensips generates core file if it crashes. I use this documentation http://www.opensips.org/Resources/DocsTsCrash. I used -w option and make changes in /proc/sys/kernel/core_pattern. May be some problems with it? Thank you for any help! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg
ОК, i will try tomorrow morning. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 11:17 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Try to set debug=4 and see the logs from the MI fifo process when running the dlg_end_dlg command. Also, try the get a backtrace from the MO FIFO proc after blocking. For that, before the dlg command, do opensipsctl fifo ps to find out the PID of the FIFO process. After running the dlg command and blocking, use gdb /path/to/opensips FIFO_PID and run bt inside - these will give some ideas on where the process blocked. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 10:07 AM, dpa wrote: So only the MI fifo part blocks? As I understand, yes. do you see any errors in the logs? I see only INFO:core:buf_init: initializing... in log file also, does it recover after some time (like minutes) ? To speak the truth I didn`t wait recovering. I used Ctrl+Z to stop process of dlg_end_dlg and tried another fifo command. ps -A show me that opensipsctl process appears. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 9:44 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg So only the MI fifo part blocks ? do you see any errors in the logs? also, does it recover after some time (like minutes) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 09:42 AM, dpa wrote: Hello Bogdan it simply blocks and does not answer I can make calls. But no fifo command available. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 19, 2012 9:39 AM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] opensips 1.6.4-2 and dlg_end_dlg Hi, Does your opensips actually crashes when you run the dlg_end_dlg command ? or it simply blocks and does not answer ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/19/2012 07:37 AM, dpa wrote: Hello! I discovered a problem with using dlg_end_dlg fifo command. When I try to make it, dialog hasn't been ended and I cannot see in console that command has been finished success. In /tmp directory (where opensips_fifo is located) I found some new files opensips_receiver_x, where x - some numbers. Ls -la shows that files were generated when I make dlg_end_dlg command. Moreover after this I cannot make any fifo command. After opensips restart I can make any fifo command but until dlg_end_dlg. There were no problems with it but recently I decided to make opensips generates core file if it crashes. I use this documentation http://www.opensips.org/Resources/DocsTsCrash. I used -w option and make changes in /proc/sys/kernel/core_pattern. May be some problems with it? Thank you for any help! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with timeout_avp.
The version of opensips? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Dragomir Haralambiev Sent: Thursday, December 13, 2012 12:46 PM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Problem with timeout_avp. Hello, I try to use timeout_avp. Here is part of my opensips.cfg: modparam(dialog, timeout_avp, $avp(maxtime)) ... create_dialog(); $avp(maxtime)=10; The call is connected and I see record in dialog tabble. After 10 sek opensips delete record from dialog table but not stop the call. Where is problem? Thanks in advance ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with timeout_avp.
Make change to opensips.cfg as Federico wrote. create_dialog(B); From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Dragomir Haralambiev Sent: Thursday, December 13, 2012 12:51 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Problem with timeout_avp. Opensips 1.8.2 revision 9513. 2012/12/13 dpa denis7...@mail.ru The version of opensips? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Dragomir Haralambiev Sent: Thursday, December 13, 2012 12:46 PM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Problem with timeout_avp. Hello, I try to use timeout_avp. Here is part of my opensips.cfg: modparam(dialog, timeout_avp, $avp(maxtime)) ... create_dialog(); $avp(maxtime)=10; The call is connected and I see record in dialog tabble. After 10 sek opensips delete record from dialog table but not stop the call. Where is problem? Thanks in advance ___ 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] Timer process opensips1.6.4-2
Hello Bogdan As I understand I should start opensips using opensips -w path to core file. Now I am using opensipsctl script for starting opensips. Can -w option be used with the script? And If cannot how can I stop opensips if early I started with opensips -w path to core file? I tried to stop it using opensipsctl stop but got error No PID file found. Thank you for any help. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, December 07, 2012 4:15 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 See more on: http://www.opensips.org/Resources/DocsTsCrash Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/07/2012 12:24 PM, Bogdan-Andrei Iancu wrote: Hi, maybe you do not have permissions to generate core. Be sure you do ulimits -c unlimited before starting OpenSIPS - this will allow opensips to write a core file if crashing. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/07/2012 05:45 AM, dpa wrote: Hello Bogdan Sorry I was wrong at the beginning. In log during crash I see INFO:core:handle_sigs: core was not generated From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, December 06, 2012 3:08 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 In your logs, when crashing, after the line with: INFO:core:handle_sigs: child process 2405 exited by a signal 11 Do you have any line with core was generated or similar. ? Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/06/2012 12:57 PM, dpa wrote: ls -la /core* doesn't show any file From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, December 06, 2012 2:48 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, look into the root of you file system: ls -la /core* Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/06/2012 06:36 AM, dpa wrote: Hello Thank you for consultation, but I cannot find any core* file in opensips directory (/usr/local/opensips1.6.4-2/). Can you tell me how can I get it? From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, December 04, 2012 6:37 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, First locate the core file - usually in the working directory of OpenSIPS (typically the / dir - try ls -la /core* ). Second, start gdb : gdb /path/to/opensips /path/to/core Run the bt command in gdb: bt and bt full Post the output here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/04/2012 03:12 PM, dpa wrote: Hello Sorry Bogdan but can you tell me how can I use gdb? I have not made it before. I noticed crash while the radius server (which I use for call limit duration request) has been failed. I use radius_send_auth() to get call limit duration from Billing. When radius has been failing opensips didn't receive any reply from it. Crashes have been stopped when I commented this function in opensips.cfg. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, December 04, 2012 4:49 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, In opensips there are multiple timer processes - different modules may ask for separate timer processes to avoid interferences with modules (in terms of load on the timer routines). Nothing unusual here. Regarding the crash, do you have a core file resulted ? if so, please use gdb to extract a backtrace from it. Thanks and Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/03/2012 08:35 AM, dpa wrote: Hello When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many processes. There are two of them have TYPE=timer. What does it mean? Today I have a twice problem with this process. INFO:core:handle_sigs: child process 2405 exited by a signal 11 Where 2405 was a timer process. Thank you for any help. ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Timer process opensips1.6.4-2
I understand, thank you. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, December 12, 2012 4:33 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, If you use opensipsctl start/stop, you can add extra parameters to opensips via the STARTOPTIONS variable in opensipsctlrc file. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/12/2012 02:04 PM, dpa wrote: Hello Bogdan As I understand I should start opensips using opensips -w path to core file. Now I am using opensipsctl script for starting opensips. Can -w option be used with the script? And If cannot how can I stop opensips if early I started with opensips -w path to core file? I tried to stop it using opensipsctl stop but got error No PID file found. Thank you for any help. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, December 07, 2012 4:15 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 See more on: http://www.opensips.org/Resources/DocsTsCrash Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/07/2012 12:24 PM, Bogdan-Andrei Iancu wrote: Hi, maybe you do not have permissions to generate core. Be sure you do ulimits -c unlimited before starting OpenSIPS - this will allow opensips to write a core file if crashing. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/07/2012 05:45 AM, dpa wrote: Hello Bogdan Sorry I was wrong at the beginning. In log during crash I see INFO:core:handle_sigs: core was not generated From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, December 06, 2012 3:08 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 In your logs, when crashing, after the line with: INFO:core:handle_sigs: child process 2405 exited by a signal 11 Do you have any line with core was generated or similar. ? Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/06/2012 12:57 PM, dpa wrote: ls -la /core* doesn't show any file From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, December 06, 2012 2:48 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, look into the root of you file system: ls -la /core* Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/06/2012 06:36 AM, dpa wrote: Hello Thank you for consultation, but I cannot find any core* file in opensips directory (/usr/local/opensips1.6.4-2/). Can you tell me how can I get it? From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, December 04, 2012 6:37 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, First locate the core file - usually in the working directory of OpenSIPS (typically the / dir - try ls -la /core* ). Second, start gdb : gdb /path/to/opensips /path/to/core Run the bt command in gdb: bt and bt full Post the output here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/04/2012 03:12 PM, dpa wrote: Hello Sorry Bogdan but can you tell me how can I use gdb? I have not made it before. I noticed crash while the radius server (which I use for call limit duration request) has been failed. I use radius_send_auth() to get call limit duration from Billing. When radius has been failing opensips didn't receive any reply from it. Crashes have been stopped when I commented this function in opensips.cfg. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, December 04, 2012 4:49 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, In opensips there are multiple timer processes - different modules may ask for separate timer processes to avoid interferences with modules (in terms of load on the timer routines). Nothing unusual here. Regarding the crash, do you have a core file resulted ? if so, please use gdb to extract a backtrace from it. Thanks and Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/03/2012 08:35 AM, dpa wrote: Hello When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many processes. There are two of them have TYPE=timer. What does it mean? Today I have a twice problem with this process. INFO:core:handle_sigs: child process 2405 exited by a signal 11 Where 2405 was a timer process. Thank you for any help. ___ Users mailing
Re: [OpenSIPS-Users] Timer process opensips1.6.4-2
Thank you Bogdan I will learn the Docs. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, December 07, 2012 4:15 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 See more on: http://www.opensips.org/Resources/DocsTsCrash Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/07/2012 12:24 PM, Bogdan-Andrei Iancu wrote: Hi, maybe you do not have permissions to generate core. Be sure you do ulimits -c unlimited before starting OpenSIPS - this will allow opensips to write a core file if crashing. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/07/2012 05:45 AM, dpa wrote: Hello Bogdan Sorry I was wrong at the beginning. In log during crash I see INFO:core:handle_sigs: core was not generated From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, December 06, 2012 3:08 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 In your logs, when crashing, after the line with: INFO:core:handle_sigs: child process 2405 exited by a signal 11 Do you have any line with core was generated or similar. ? Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/06/2012 12:57 PM, dpa wrote: ls -la /core* doesn't show any file From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, December 06, 2012 2:48 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, look into the root of you file system: ls -la /core* Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/06/2012 06:36 AM, dpa wrote: Hello Thank you for consultation, but I cannot find any core* file in opensips directory (/usr/local/opensips1.6.4-2/). Can you tell me how can I get it? From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, December 04, 2012 6:37 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, First locate the core file - usually in the working directory of OpenSIPS (typically the / dir - try ls -la /core* ). Second, start gdb : gdb /path/to/opensips /path/to/core Run the bt command in gdb: bt and bt full Post the output here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/04/2012 03:12 PM, dpa wrote: Hello Sorry Bogdan but can you tell me how can I use gdb? I have not made it before. I noticed crash while the radius server (which I use for call limit duration request) has been failed. I use radius_send_auth() to get call limit duration from Billing. When radius has been failing opensips didn't receive any reply from it. Crashes have been stopped when I commented this function in opensips.cfg. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, December 04, 2012 4:49 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, In opensips there are multiple timer processes - different modules may ask for separate timer processes to avoid interferences with modules (in terms of load on the timer routines). Nothing unusual here. Regarding the crash, do you have a core file resulted ? if so, please use gdb to extract a backtrace from it. Thanks and Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/03/2012 08:35 AM, dpa wrote: Hello When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many processes. There are two of them have TYPE=timer. What does it mean? Today I have a twice problem with this process. INFO:core:handle_sigs: child process 2405 exited by a signal 11 Where 2405 was a timer process. Thank you for any help. ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Timer process opensips1.6.4-2
Hello Thank you for consultation, but I cannot find any core* file in opensips directory (/usr/local/opensips1.6.4-2/). Can you tell me how can I get it? From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, December 04, 2012 6:37 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, First locate the core file - usually in the working directory of OpenSIPS (typically the / dir - try ls -la /core* ). Second, start gdb : gdb /path/to/opensips /path/to/core Run the bt command in gdb: bt and bt full Post the output here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/04/2012 03:12 PM, dpa wrote: Hello Sorry Bogdan but can you tell me how can I use gdb? I have not made it before. I noticed crash while the radius server (which I use for call limit duration request) has been failed. I use radius_send_auth() to get call limit duration from Billing. When radius has been failing opensips didn't receive any reply from it. Crashes have been stopped when I commented this function in opensips.cfg. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, December 04, 2012 4:49 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, In opensips there are multiple timer processes - different modules may ask for separate timer processes to avoid interferences with modules (in terms of load on the timer routines). Nothing unusual here. Regarding the crash, do you have a core file resulted ? if so, please use gdb to extract a backtrace from it. Thanks and Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/03/2012 08:35 AM, dpa wrote: Hello When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many processes. There are two of them have TYPE=timer. What does it mean? Today I have a twice problem with this process. INFO:core:handle_sigs: child process 2405 exited by a signal 11 Where 2405 was a timer process. Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Install opensips 1.8.2
Thank you, I will check 2 point. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Muhammad Shahzad Sent: Tuesday, December 04, 2012 4:48 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Install opensips 1.8.2 Sorry i think two separate issues are missed in this thread and I answered the second one instead of first one. Anyhow, here is the answer for your original problem. 1. Make sure you have libmysqlclient-dev installed. If you have mysql libraries installed on some custom location then make sure to add them using CFLAGS and LDFLAGS directive. 2. Look at Makefile.conf in opensips sources and make sure db_mysql is NOT in exclude_modules list. 3. Do make and make install. Thank you. On Tue, Dec 4, 2012 at 5:18 AM, dpa denis7...@mail.ru wrote: I can compile and install opensips with db_mysql only make “make prefix=/usr/local/opensips1.8.2/ all include_modules=db_mysql make prefix=/usr/local/opensips1.8.2/ install include_modules=db_mysql” From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Monday, December 03, 2012 1:23 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Install opensips 1.8.2 Hello There is a desire to install opensips 1.8.2 from src. I download it from web, change Makefile in “exclude_modules” area by excluding db_mysql from the list of “exclude_modules”. Do “make prefix=/usr/local/opensips1.8.2/ all”. After that I look at modules/db_mysql directory and cannot see any db_mysql.so. So if I make “make prefix=/usr/local/opensips1.8.2/ install” i cannot see any db_mysql.so in “/usr/local/opensips1.8.2/lib/opensips/modules” I tried to do “make prefix=/usr/local/opensips1.8.2/ install include_modules=modules/mysql” but problem is still exists. The only way to compile db_mysql is to do “make” from “modules/db_mysql” directory. My goal is to migrate DB from 1.7.0 to 1.8.2 and as I understand I need to compile db_mysql from general directory but not from module directory to make opensipsctl support mysql. Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: mailto:shari_78...@hotmail.com shari_78...@hotmail.com Email: mailto:shaherya...@googlemail.com shaherya...@googlemail.com image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Timer process opensips1.6.4-2
Hello Sorry Bogdan but can you tell me how can I use gdb? I have not made it before. I noticed crash while the radius server (which I use for call limit duration request) has been failed. I use radius_send_auth() to get call limit duration from Billing. When radius has been failing opensips didn't receive any reply from it. Crashes have been stopped when I commented this function in opensips.cfg. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, December 04, 2012 4:49 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] Timer process opensips1.6.4-2 Hi, In opensips there are multiple timer processes - different modules may ask for separate timer processes to avoid interferences with modules (in terms of load on the timer routines). Nothing unusual here. Regarding the crash, do you have a core file resulted ? if so, please use gdb to extract a backtrace from it. Thanks and Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12/03/2012 08:35 AM, dpa wrote: Hello When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many processes. There are two of them have TYPE=timer. What does it mean? Today I have a twice problem with this process. INFO:core:handle_sigs: child process 2405 exited by a signal 11 Where 2405 was a timer process. Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Install opensips 1.8.2
Hello There is a desire to install opensips 1.8.2 from src. I download it from web, change Makefile in exclude_modules area by excluding db_mysql from the list of exclude_modules. Do make prefix=/usr/local/opensips1.8.2/ all. After that I look at modules/db_mysql directory and cannot see any db_mysql.so. So if I make make prefix=/usr/local/opensips1.8.2/ install i cannot see any db_mysql.so in /usr/local/opensips1.8.2/lib/opensips/modules I tried to do make prefix=/usr/local/opensips1.8.2/ install include_modules=modules/mysql but problem is still exists. The only way to compile db_mysql is to do make from modules/db_mysql directory. My goal is to migrate DB from 1.7.0 to 1.8.2 and as I understand I need to compile db_mysql from general directory but not from module directory to make opensipsctl support mysql. Thank you for any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Install opensips 1.8.2
I can compile and install opensips with db_mysql only make make prefix=/usr/local/opensips1.8.2/ all include_modules=db_mysql make prefix=/usr/local/opensips1.8.2/ install include_modules=db_mysql From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of dpa Sent: Monday, December 03, 2012 1:23 PM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Install opensips 1.8.2 Hello There is a desire to install opensips 1.8.2 from src. I download it from web, change Makefile in exclude_modules area by excluding db_mysql from the list of exclude_modules. Do make prefix=/usr/local/opensips1.8.2/ all. After that I look at modules/db_mysql directory and cannot see any db_mysql.so. So if I make make prefix=/usr/local/opensips1.8.2/ install i cannot see any db_mysql.so in /usr/local/opensips1.8.2/lib/opensips/modules I tried to do make prefix=/usr/local/opensips1.8.2/ install include_modules=modules/mysql but problem is still exists. The only way to compile db_mysql is to do make from modules/db_mysql directory. My goal is to migrate DB from 1.7.0 to 1.8.2 and as I understand I need to compile db_mysql from general directory but not from module directory to make opensipsctl support mysql. Thank you for any help. image002.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Timer process opensips1.6.4-2
Hello When I use /usr/local/opensips1.6.4-2/sbin/opensipsctl fifo ps i see many processes. There are two of them have TYPE=timer. What does it mean? Today I have a twice problem with this process. INFO:core:handle_sigs: child process 2405 exited by a signal 11 Where 2405 was a timer process. Thank you for any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0
Yes, problem was there. Thank you very much. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, November 30, 2012 1:05 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0 Assuming that you have other files in that dir (/usr/local/opensips1.7.0//lib/opensips/opensipsctl/), I suppose you did not include mysql module in the list of modules to be compiled + installed - the mysql support in opensipsctl was not installed too. So do make install include_modules=modules/mysql. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/30/2012 06:19 AM, dpa wrote: Hello, Bogdan No, there is no opensipsdbctl.mysql file in /usr/local/opensips1.7.0//lib/opensips/opensipsctl/ I should make it myself? From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, November 29, 2012 9:36 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0 Hi, and have you checked if the file /usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql really exists on your system ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/28/2012 10:46 AM, dpa wrote: Hello! I want to upgrade opensips from 1.6.4-2 to latest version (1.8). As I understand first I should migrate my DB from 1.6.4 to 1.7.0 and then migrate 1.7.0 to 1.8 I have installed opensips 1.7.0. Make ./opensipsdbctl migrate opensips5 opensips51 where opensips5 - existing DB. and receive such errors ERROR: could not load the script in /usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0
Hello, Bogdan No, there is no opensipsdbctl.mysql file in /usr/local/opensips1.7.0//lib/opensips/opensipsctl/ I should make it myself? From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, November 29, 2012 9:36 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0 Hi, and have you checked if the file /usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql really exists on your system ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11/28/2012 10:46 AM, dpa wrote: Hello! I want to upgrade opensips from 1.6.4-2 to latest version (1.8). As I understand first I should migrate my DB from 1.6.4 to 1.7.0 and then migrate 1.7.0 to 1.8 I have installed opensips 1.7.0. Make ./opensipsdbctl migrate opensips5 opensips51 where opensips5 - existing DB. and receive such errors ERROR: could not load the script in /usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] DB migration from 1.6.4 to 1.7.0
Hello! I want to upgrade opensips from 1.6.4-2 to latest version (1.8). As I understand first I should migrate my DB from 1.6.4 to 1.7.0 and then migrate 1.7.0 to 1.8 I have installed opensips 1.7.0. Make ./opensipsdbctl migrate opensips5 opensips51 where opensips5 - existing DB. and receive such errors ERROR: could not load the script in /usr/local/opensips1.7.0//lib/opensips/opensipsctl/opensipsdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' Thank you for any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] extra duration of a call in 1.6.4-2
Hello I am using “bye_on_timeout_flag” for each transaction in Opensips. The default timeout – 10 s. “Real timeout” rewriting after loose_routing() when routing ACK request on 200 ОК and different for clients. The max timeout in Opensips for dialogs is s. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Binan AL Halabi Sent: Monday, November 12, 2012 1:31 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] extra duration of a call in 1.6.4-2 Hi Denis, The client must send BYE to end the dialog. but in case of timeout you can generate the local BYE to end the dialog: create_dialog(B); The string B activate the BYE on timeout behavior. In OpenSIPS 1.6 use bye_on_timeout_flag. // Binan _ Från: dpa denis7...@mail.ru Till: 'OpenSIPS users mailling list' users@lists.opensips.org Skickat: måndag, 12 november 2012 7:30 Ämne: [OpenSIPS-Users] extra duration of a call in 1.6.4-2 Hello, I have detected one call which has duration 28 seconds. I found this call in ngrep log (you can get it from here http://files.mail.ru/CU4GGS). 1.1.1.1– One UA 2.2.2.2 – Opensips 1.6.4-2 3.3.3.3 – Another UA After analyze the log I cannot find a reason why this call has so big duration. In addition I control a duration of any calls and from this client the call shouldn`t be alive more than 1800 seconds. As you can see in log there is no local BYE for the session. Thank you for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] extra duration of a call in 1.6.4-2
Hello, I have detected one call which has duration 28 seconds. I found this call in ngrep log (you can get it from here http://files.mail.ru/CU4GGS). 1.1.1.1- One UA 2.2.2.2 - Opensips 1.6.4-2 3.3.3.3 - Another UA After analyze the log I cannot find a reason why this call has so big duration. In addition I control a duration of any calls and from this client the call shouldn`t be alive more than 1800 seconds. As you can see in log there is no local BYE for the session. Thank you for any help image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips upgrade
Hello! Sorry if it were but anyway. I am using opensips 1.6.4-2 and have a wish to upgrade to 1.8 I have directory /usr/local/opensips1.6.4-2/ where opensips 1.6.4-2 has been installed at one time. If I will make tar archive of this directory and install to this directory opensips 1.8? Would opensips 1.8 work after this? (of course I will make change to opensips.cfg and sql database) And if for some reason I will need change back opensips to 1.6.4-2? May I make same procedure using tar archive which I made in previous step? Thank you for any help. P.S. It is important to using directory /usr/local/opensips1.6.4-2/ after upgrade to 1.8. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Rtpproxy connection
You should first start opensips, it creates unix socket and then rtpproxy. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of spady Sent: Thursday, September 27, 2012 4:27 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Rtpproxy connection I have some news but issue still present. I used unix socket, instead of udp, but RTPPROXY crashes at each call. Below traces of opensips side and rtpproxy side: OPENSIPS: *Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26026]: ERROR:rtpproxy:send_rtpp_command: can't read reply from a RTP proxy Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26026]: ERROR:rtpproxy:send_rtpp_command: proxy unix:/var/run/rtpproxy.sock does not respond, disable it Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26026]: ERROR:rtpproxy:force_rtp_proxy_body: no available proxies Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26020]: ERROR:rtpproxy:force_rtp_proxy: Unable to parse body Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26021]: ERROR:rtpproxy:force_rtp_proxy: Unable to parse body Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26024]: ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26022]: ERROR:rtpproxy:force_rtp_proxy: Unable to parse body Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26022]: ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26022]: ERROR:rtpproxy:engage_close_callback: cannot unforce rtp proxy Sep 27 13:02:02 opensips /usr/local/opensips_proxy/sbin/opensips[26025]: ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies Sep 27 13:02:03 opensips /usr/local/opensips_proxy/sbin/opensips[26026]: ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies Sep 27 13:02:05 opensips /usr/local/opensips_proxy/sbin/opensips[26024]: ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies * RTPPROXY: *Sep 27 13:01:46 opensips rtpproxy[25993]: DBUG:handle_command: received command VF 20071116 Sep 27 13:01:57 opensips rtpproxy[25993]: DBUG:handle_command: received command UIER172.16.52.121c0,8,18,101 ZGY2Yjk5ZGJiNDQ0ZWI0MGVhZmMzODAxNDc3YzgwMjI 151.x.x.19 60242 671d9c3d;1 * Any idea??? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-tp 7581935p7581952.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] Rtpproxy connection
Hello Instead of udp socket for control try to use unix socket. Maybe it will help. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of spady Sent: Thursday, September 27, 2012 2:41 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Rtpproxy connection - No!, after opensips module sends command to rtpproxy this one shut down ( i cannot see anymore with ps command ) - in rtpproxy logs i cannot see any specific error. Below example of code that I get from rtpproxy: *Sep 27 00:31:01 opensips rtpproxy[19991]: DBUG:handle_command: received command 20126_5 UIER172.16.52.121c0,8,18,101 M2EzYzNmMmRmYzg0MGNiY2M1YjhjMTUzMWFkOWNiYTg 37.103.117.107 60234 cfd2be4f;1 Sep 27 00:31:01 opensips rtpproxy[19991]: INFO:handle_command: new session M2EzYzNmMmRmYzg0MGNiY2M1YjhjMTUzMWFkOWNiYTg, tag cfd2be4f;1 requested, type strong* As you can see, no error!!! P.S. I started rtpproxy like below: *rtpproxy -F -s udp:127.0.0.1:10177 -l 10.9.23.41/151.x.x.201 -u root -d DBUG:LOG_LOCAL2* -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-tp 7581935p7581940.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
[OpenSIPS-Users] Opensips and rtpproxy
Hello! There is one question about Opensips and rtpproxy interaction. Documentation says that I can use more than one rtpproxy set. And to choose which rtpproxy to use I should write to opensips.cfg something like modparam(nathelper, rtpproxy_sock, 1 == /sock/rtpproxy.sock) modparam(nathelper, rtpproxy_sock, 2 == /sock/rtpproxy1.sock) And what about rtpp_notify_socket parameter? Can I write it to opensips.cfg the same way or multiple rtpproxy can use similar unix socket for timeout notification? Opensips 1.6.4-2. Thank you for any help image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips and rtpproxy
Hello! And I cannot use one unix socket (timeout notification) for several rtpproxy set? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Monday, August 13, 2012 4:46 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Opensips and rtpproxy Hello! No, currently OpenSIPS doesn't support multiple notification sockets. Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 08/13/2012 01:25 PM, dpa wrote: Hello! There is one question about Opensips and rtpproxy interaction. Documentation says that I can use more than one rtpproxy set. And to choose which rtpproxy to use I should write to opensips.cfg something like “modparam(nathelper, rtpproxy_sock, 1 == /sock/rtpproxy.sock) modparam(nathelper, rtpproxy_sock, 2 == /sock/rtpproxy1.sock)” And what about “rtpp_notify_socket” parameter? Can I write it to opensips.cfg the same way or multiple rtpproxy can use similar unix socket for timeout notification? Opensips 1.6.4-2. Thank you for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips and rtpproxy
I understand, thank you. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Monday, August 13, 2012 5:06 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Opensips and rtpproxy Hello! You can use for several sets, but only a single socket for all of them. Regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 08/13/2012 03:56 PM, dpa wrote: Hello! And I cannot use one unix socket (timeout notification) for several rtpproxy set? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Monday, August 13, 2012 4:46 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Opensips and rtpproxy Hello! No, currently OpenSIPS doesn't support multiple notification sockets. Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 08/13/2012 01:25 PM, dpa wrote: Hello! There is one question about Opensips and rtpproxy interaction. Documentation says that I can use more than one rtpproxy set. And to choose which rtpproxy to use I should write to opensips.cfg something like “modparam(nathelper, rtpproxy_sock, 1 == /sock/rtpproxy.sock) modparam(nathelper, rtpproxy_sock, 2 == /sock/rtpproxy1.sock)” And what about “rtpp_notify_socket” parameter? Can I write it to opensips.cfg the same way or multiple rtpproxy can use similar unix socket for timeout notification? Opensips 1.6.4-2. Thank you for any help ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] rtpproxy
Hello Bogdan http://files.mail.ru/7HLCXK - here you can get faxcall and rtpproxy communication. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, July 25, 2012 8:40 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] rtpproxy Denis, if you are connecting to rtpproxy via an UDP connector, simply use ngrep (on the configured port). If using a unixsock connector, switch to a UDP one (rtpproxy_sock = udp:localhost:8899 and do a similar change in the rtpproxy command line). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/23/2012 12:02 PM, dpa wrote: Hello To speak the truth I do not know how to do it. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Monday, July 23, 2012 12:54 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] rtpproxy Hi Denis, could you make a capture to see the comunication between opensips and rtproxy ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/20/2012 01:11 PM, dpa wrote: Hello Bogdan I am very sorry, last time I gave wrong information about the problem. Only with r flag there is no fax stream through rtpproxy (last email I said that everything is OK, but I made wrong test). Here http://files.mail.ru/XF4J0L Is tcpdump of unsuccessful call. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, July 17, 2012 11:44 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] rtpproxy Hi , Try only with r then. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/17/2012 06:26 AM, dpa wrote: Hello Bogdan I applied r and a flags and problem with fax disappeared, but I got another problem. After I applied ra clients behind nat cannot hear caller or callee (never mind who makes call). Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but there is no rtp stream. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, July 12, 2012 7:01 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] rtpproxy Hi, According to your trace the RTPproxy is properly inserted during the re-INVITE. Try to put the r and a flags when using RTPproxy from script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 04:32 PM, dpa wrote: Hello Bogdan Thank you for answering. My case is 2). First I have audio stream which is replaced later by another fax stream. Here http://files.mail.ru/8BYVCQ is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t proxy fax stream. Rtpproxy (as an Opensips) is located on 213.170.100.150 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Wednesday, July 11, 2012 4:43 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpproxy Hi, You may have 2 cases, not sure which is yours: 1) the 2 media sessions are in parallel in the same calls (you can have 2 -or more- RTP streams in the same SIP session). In this case, RTPproxy will take care automatically of each RTP stream) 2) you have first one audio stream which is replaced by another FAX stream, via a re-INVITE - in this case, if you the normal offer / answer for RTPproxy also for the re-INVITE, it should be ok. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 11:42 AM, dpa wrote: Описание: cid:part1.00030104.06050802@opensips.org Hello There is one question about rtpproxy working. If some UA using 2 different UDP ports for sending RTP and t38 fax packets during one call session, whether rtpproxy is working (proxy both UDP streams) in such case? Thank you for any help ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.6.4-2
Hello I have one question I had such opensips.cfg route { .. create_dialog(); //INVITE processing store_dlg_value(error,yes); . } onreply_route[1] { .. if (status=~200||18[0,3] $rm==INVITE) { .. store_dlg_value(error,no); .. } . } failure_route [1] { . fetch_dlg_value(error,$avp(i:42)); xlog(L_INFO, FAIL_route:$rs was received (IP=$si, CALLID=$ci, FROMTAG=$ft, TOTAG=$tt, AVP(i:42)=$avp(i:42))); .. } In attachment there is log of a call. 2.2.2.2 - Opensips. 1.1.1.1 and 3.3.3.3 - some UA. As you can see there is no 200 or 183 replies in this call. But in log file I see Jul 24 15:52:20 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[11522]: FAIL_route:null was received (IP=1.1.1.1, CALLID=676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1, FROMTAG=67000fc7a4, TOTAG=null, AVP(i:42)=no) The question is why does AVP(i:42) has meaning no? Thank you for any help. image001.gifU 2012/07/24 15:52:20.968054 1.1.1.1:5060 - 2.2.2.2:5060 INVITE sip:735603@2.2.2.2 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216. From: sip:4862442964@1.1.1.1;tag=67000fc7a4. To: sip:735603@2.2.2.2. Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1. CSeq: 216 INVITE. Supported: replaces, timer, 100rel, early-session. Min-SE: 1800. Date: Sun, 11 Jan 2010 22:42:47 GMT. Session-Expires: 1800. User-Agent: AddPac SIP Gateway. Contact: sip:4862442964@1.1.1.1. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 270. Max-Forwards: 70. . v=0. o=4862442964 945767 945767 IN IP4 1.1.1.1. s=AddPac Gateway SDP. c=IN IP4 1.1.1.1. t=0 0. m=audio 23002 RTP/AVP 18 8 0 101. a=ptime:80. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 2012/07/24 15:52:20.968307 2.2.2.2:5060 - 1.1.1.1:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216. From: sip:4862442964@1.1.1.1;tag=67000fc7a4. To: sip:735603@2.2.2.2. Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1. CSeq: 216 INVITE. Content-Length: 0. . U 2012/07/24 15:52:20.970247 2.2.2.2:5060 - 3.3.3.3:5060 INVITE sip:735603@3.3.3.3:5060 SIP/2.0. Record-Route: sip:2.2.2.2;lr=on;ftag=67000fc7a4;did=4b6.ab160851. Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK8f52.5ec4773.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216. From: sip:4862442964@1.1.1.1;tag=67000fc7a4. To: sip:735603@2.2.2.2. Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1. CSeq: 216 INVITE. Supported: replaces, timer, 100rel, early-session. Min-SE: 1800. Date: Sun, 11 Jan 2010 22:42:47 GMT. Session-Expires: 1800. User-Agent: AddPac SIP Gateway. Contact: sip:4862442964@1.1.1.1. P-Asserted-Identity: sip:4862442964@1.1.1.1. Remote-Party-ID:sip:4862442964@1.1.1.1;party=calling;screen=yes;privacy=off. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 294. Max-Forwards: 70. . v=0. o=4862442964 945767 945767 IN IP4 2.2.2.2. s=AddPac Gateway SDP. c=IN IP4 2.2.2.2. t=0 0. m=audio 40834 RTP/AVP 18 8 0 101. a=ptime:80. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=nortpproxy:yes. U 2012/07/24 15:52:20.994730 3.3.3.3:5060 - 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK8f52.5ec4773.0,SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216. From: sip:4862442964@1.1.1.1;tag=67000fc7a4. To: sip:735603@2.2.2.2. Date: Tue, 24 Jul 2012 11:56:20 GMT. Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1. CSeq: 216 INVITE. Allow-Events: telephone-event. Server: Cisco-SIPGateway/IOS-12.x. Content-Length: 0. . U 2012/07/24 15:52:21.981394 3.3.3.3:5060 - 2.2.2.2:5060 SIP/2.0 486 Busy here. Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK8f52.5ec4773.0,SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216. From: sip:4862442964@1.1.1.1;tag=67000fc7a4. To: sip:735603@2.2.2.2;tag=E9560618-197A. Date: Tue, 24 Jul 2012 11:56:20 GMT. Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1. CSeq: 216 INVITE. Allow-Events: telephone-event. Server: Cisco-SIPGateway/IOS-12.x. Reason: Q.850;cause=17. Content-Length: 0. . U 2012/07/24 15:52:21.981549 2.2.2.2:5060 - 3.3.3.3:5060 ACK sip:735603@3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK8f52.5ec4773.0. From: sip:4862442964@1.1.1.1;tag=67000fc7a4. Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1. To: sip:735603@2.2.2.2;tag=E9560618-197A. CSeq: 216 ACK. Max-Forwards: 70. Content-Length: 0. . U 2012/07/24 15:52:22.011214 2.2.2.2:5060 - 1.1.1.1:5060 SIP/2.0 486 Busy here. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK67000fc7a4216. From: sip:4862442964@1.1.1.1;tag=67000fc7a4. To: sip:735603@2.2.2.2;tag=E9560618-197A. Date: Tue, 24 Jul 2012 11:56:20 GMT. Call-ID: 676e0e00-ac77-0f22-80c7-0002a4085a16@1.1.1.1. CSeq: 216 INVITE.
Re: [OpenSIPS-Users] rtpproxy
Hello To speak the truth I do not know how to do it. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Monday, July 23, 2012 12:54 PM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] rtpproxy Hi Denis, could you make a capture to see the comunication between opensips and rtproxy ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/20/2012 01:11 PM, dpa wrote: Hello Bogdan I am very sorry, last time I gave wrong information about the problem. Only with r flag there is no fax stream through rtpproxy (last email I said that everything is OK, but I made wrong test). Here http://files.mail.ru/XF4J0L Is tcpdump of unsuccessful call. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, July 17, 2012 11:44 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] rtpproxy Hi , Try only with r then. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/17/2012 06:26 AM, dpa wrote: Hello Bogdan I applied r and a flags and problem with fax disappeared, but I got another problem. After I applied ra clients behind nat cannot hear caller or callee (never mind who makes call). Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but there is no rtp stream. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, July 12, 2012 7:01 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] rtpproxy Hi, According to your trace the RTPproxy is properly inserted during the re-INVITE. Try to put the r and a flags when using RTPproxy from script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 04:32 PM, dpa wrote: Hello Bogdan Thank you for answering. My case is 2). First I have audio stream which is replaced later by another fax stream. Here http://files.mail.ru/8BYVCQ is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t proxy fax stream. Rtpproxy (as an Opensips) is located on 213.170.100.150 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Wednesday, July 11, 2012 4:43 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpproxy Hi, You may have 2 cases, not sure which is yours: 1) the 2 media sessions are in parallel in the same calls (you can have 2 -or more- RTP streams in the same SIP session). In this case, RTPproxy will take care automatically of each RTP stream) 2) you have first one audio stream which is replaced by another FAX stream, via a re-INVITE - in this case, if you the normal offer / answer for RTPproxy also for the re-INVITE, it should be ok. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 11:42 AM, dpa wrote: Описание: cid:part1.00030104.06050802@opensips.org Hello There is one question about rtpproxy working. If some UA using 2 different UDP ports for sending RTP and t38 fax packets during one call session, whether rtpproxy is working (proxy both UDP streams) in such case? Thank you for any help ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] rtpproxy
Hello Bogdan I am very sorry, last time I gave wrong information about the problem. Only with r flag there is no fax stream through rtpproxy (last email I said that everything is OK, but I made wrong test). Here http://files.mail.ru/XF4J0L Is tcpdump of unsuccessful call. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, July 17, 2012 11:44 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] rtpproxy Hi , Try only with r then. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/17/2012 06:26 AM, dpa wrote: Hello Bogdan I applied r and a flags and problem with fax disappeared, but I got another problem. After I applied ra clients behind nat cannot hear caller or callee (never mind who makes call). Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but there is no rtp stream. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, July 12, 2012 7:01 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] rtpproxy Hi, According to your trace the RTPproxy is properly inserted during the re-INVITE. Try to put the r and a flags when using RTPproxy from script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 04:32 PM, dpa wrote: Hello Bogdan Thank you for answering. My case is 2). First I have audio stream which is replaced later by another fax stream. Here http://files.mail.ru/8BYVCQ is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t proxy fax stream. Rtpproxy (as an Opensips) is located on 213.170.100.150 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Wednesday, July 11, 2012 4:43 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpproxy Hi, You may have 2 cases, not sure which is yours: 1) the 2 media sessions are in parallel in the same calls (you can have 2 -or more- RTP streams in the same SIP session). In this case, RTPproxy will take care automatically of each RTP stream) 2) you have first one audio stream which is replaced by another FAX stream, via a re-INVITE - in this case, if you the normal offer / answer for RTPproxy also for the re-INVITE, it should be ok. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 11:42 AM, dpa wrote: Описание: cid:part1.00030104.06050802@opensips.org Hello There is one question about rtpproxy working. If some UA using 2 different UDP ports for sending RTP and t38 fax packets during one call session, whether rtpproxy is working (proxy both UDP streams) in such case? Thank you for any help ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] rtpproxy
Hello Bogdan Now it's working properly, both fax and rtp. Thank you very much. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Tuesday, July 17, 2012 11:44 AM To: dpa Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] rtpproxy Hi , Try only with r then. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/17/2012 06:26 AM, dpa wrote: Hello Bogdan I applied r and a flags and problem with fax disappeared, but I got another problem. After I applied ra clients behind nat cannot hear caller or callee (never mind who makes call). Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but there is no rtp stream. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, July 12, 2012 7:01 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] rtpproxy Hi, According to your trace the RTPproxy is properly inserted during the re-INVITE. Try to put the r and a flags when using RTPproxy from script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 04:32 PM, dpa wrote: Hello Bogdan Thank you for answering. My case is 2). First I have audio stream which is replaced later by another fax stream. Here http://files.mail.ru/8BYVCQ is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t proxy fax stream. Rtpproxy (as an Opensips) is located on 213.170.100.150 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Wednesday, July 11, 2012 4:43 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpproxy Hi, You may have 2 cases, not sure which is yours: 1) the 2 media sessions are in parallel in the same calls (you can have 2 -or more- RTP streams in the same SIP session). In this case, RTPproxy will take care automatically of each RTP stream) 2) you have first one audio stream which is replaced by another FAX stream, via a re-INVITE - in this case, if you the normal offer / answer for RTPproxy also for the re-INVITE, it should be ok. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 11:42 AM, dpa wrote: Описание: cid:part1.00030104.06050802@opensips.org Hello There is one question about rtpproxy working. If some UA using 2 different UDP ports for sending RTP and t38 fax packets during one call session, whether rtpproxy is working (proxy both UDP streams) in such case? Thank you for any help ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] rtpproxy
Hello Bogdan I applied r and a flags and problem with fax disappeared, but I got another problem. After I applied ra clients behind nat cannot hear caller or callee (never mind who makes call). Rtpproxy is working, i.e. it changes SDP body of INVITE and replies, but there is no rtp stream. From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, July 12, 2012 7:01 PM To: OpenSIPS users mailling list Cc: dpa Subject: Re: [OpenSIPS-Users] rtpproxy Hi, According to your trace the RTPproxy is properly inserted during the re-INVITE. Try to put the r and a flags when using RTPproxy from script. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 04:32 PM, dpa wrote: Hello Bogdan Thank you for answering. My case is 2). First I have audio stream which is replaced later by another fax stream. Here http://files.mail.ru/8BYVCQ is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t proxy fax stream. Rtpproxy (as an Opensips) is located on 213.170.100.150 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Wednesday, July 11, 2012 4:43 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpproxy Hi, You may have 2 cases, not sure which is yours: 1) the 2 media sessions are in parallel in the same calls (you can have 2 -or more- RTP streams in the same SIP session). In this case, RTPproxy will take care automatically of each RTP stream) 2) you have first one audio stream which is replaced by another FAX stream, via a re-INVITE - in this case, if you the normal offer / answer for RTPproxy also for the re-INVITE, it should be ok. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 11:42 AM, dpa wrote: Описание: cid:part1.00030104.06050802@opensips.org Hello There is one question about rtpproxy working. If some UA using 2 different UDP ports for sending RTP and t38 fax packets during one call session, whether rtpproxy is working (proxy both UDP streams) in such case? Thank you for any help ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] rtpproxy
Hello There is one question about rtpproxy working. If some UA using 2 different UDP ports for sending RTP and t38 fax packets during one call session, whether rtpproxy is working (proxy both UDP streams) in such case? Thank you for any help image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] rtpproxy
Hello Bogdan Thank you for answering. My case is 2). First I have audio stream which is replaced later by another fax stream. Here http://files.mail.ru/8BYVCQ is a tcpdump of unsuccessful call where you can see that rtpproxy doesn`t proxy fax stream. Rtpproxy (as an Opensips) is located on 213.170.100.150 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Wednesday, July 11, 2012 4:43 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpproxy Hi, You may have 2 cases, not sure which is yours: 1) the 2 media sessions are in parallel in the same calls (you can have 2 -or more- RTP streams in the same SIP session). In this case, RTPproxy will take care automatically of each RTP stream) 2) you have first one audio stream which is replaced by another FAX stream, via a re-INVITE - in this case, if you the normal offer / answer for RTPproxy also for the re-INVITE, it should be ok. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07/11/2012 11:42 AM, dpa wrote: Описание: cid:part1.00030104.06050802@opensips.org Hello There is one question about rtpproxy working. If some UA using 2 different UDP ports for sending RTP and t38 fax packets during one call session, whether rtpproxy is working (proxy both UDP streams) in such case? Thank you for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] FW: Opensips memory problem
Hello! Opensips 1.6.4-2. When doing dp_reload or dr_reload there are such errors: Jun 27 17:12:53 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Jun 27 17:12:53 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: ERROR:core:db_allocate_rows: no memory left Jun 27 17:12:53 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: ERROR:db_mysql:db_mysql_fetch_result: no memory left Jun 27 17:12:53 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: ERROR:dialplan:dp_load_db: failed to fetch Jun 27 17:12:53 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: ERROR:dialplan:mi_reload_rules: failed to reload database data Jun 27 17:12:53 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: ERROR:mi_fifo:mi_fifo_server: command () processing failed Jun 27 17:14:43 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: INFO:drouting:dr_reload_cmd: dr_reload MI command received! Jun 27 17:14:43 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Jun 27 17:14:43 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: ERROR:core:db_allocate_rows: no memory left Jun 27 17:14:43 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: ERROR:db_mysql:db_mysql_fetch_result: no memory left Jun 27 17:14:43 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: ERROR:drouting:dr_load_routing_info: Error fetching rows Jun 27 17:14:43 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: CRITICAL:drouting:dr_reload_data: failed to load routing info Jun 27 17:14:43 opensips-mirror /usr/local/opensips1.6.4-2/sbin/opensips[21801]: CRITICAL:drouting:dr_reload_cmd: failed to load routing data When, for example, doing address reload there is no problem. There is no problem with call proceeding. opensips.cfg ... modparam(drouting, fetch_rows, 1000) .. modparam(dialplan, fetch_rows, 1000) Opensips memory shmem:total_size = 104857600 shmem:used_size = 41207560 shmem:real_used_size = 46998880 shmem:max_used_size = 53773824 shmem:free_size = 57858720 shmem:fragments = 54557 pkmem:0-total_size = 20971520 pkmem:0-used_size = 710496 pkmem:0-real_used_size = 857016 pkmem:0-max_used_size = 1036592 pkmem:0-free_size = 20114504 pkmem:0-fragments = 3 Other process has nearly same values. Thank you for any help image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips and rtpproxy
Hello! Opensips 1.6.4-2 There is one question. rtpp_notify_socket may be only unix socket? Or I can use UPD or TCP too? Thank you for any help image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips and rtpproxy
Hello Răzvan Thank you for replay. But one more question In man for rtpproxy I read -n timeout_socket.. The socket should be created by another application, preferably before starting rtpproxy.. So as I understand rtpp_notify_socket should be like modparam(nathelper, rtpp_notify_socket, tcp:localhost:) and after this rtpproxy should be started with -n tcp:1.1.1.1: where 1.1.1.1 - ip of opensips serever. Am I right? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Wednesday, May 23, 2012 5:47 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips and rtpproxy Hi, Denis! The socket can be UNIX or TCP according to the documentation[1]. [1] http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id250454 Regards, -- Răzvan Crainea OpenSIPS Developer http://www.opensips-solutions.com On 05/23/2012 04:11 PM, dpa wrote: Hello! Opensips 1.6.4-2 There is one question. rtpp_notify_socket may be only unix socket? Or I can use UPD or TCP too? Thank you for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips and rtpproxy
Result after timeout detection May 23 18:30:10 kam rtpproxy[5925]: ERR:reconnect_timeout_handler: can't connect to timeout socket: Connection refused May 23 18:30:10 kam rtpproxy[5925]: ERR:do_timeout_notification: unable to send timeout notification From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Wednesday, May 23, 2012 6:23 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips and rtpproxy Hi, Denis! Try to use the same socket for both rtpproxy module and application: modparam(nathelper, rtpp_notify_socket, tcp:127.0.0.1:) rtpproxy -n tcp:127.0.0.1: Regards, -- Răzvan Crainea OpenSIPS Developer http://www.opensips-solutions.com On 05/23/2012 05:13 PM, dpa wrote: Hello Răzvan Thank you for replay. But one more question In man for rtpproxy I read -n timeout_socket.. The socket should be created by another application, preferably before starting rtpproxy.. So as I understand rtpp_notify_socket should be like modparam(nathelper, rtpp_notify_socket, tcp:localhost:) and after this rtpproxy should be started with -n tcp:1.1.1.1: where 1.1.1.1 - ip of opensips serever. Am I right? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Wednesday, May 23, 2012 5:47 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips and rtpproxy Hi, Denis! The socket can be UNIX or TCP according to the documentation[1]. [1] http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id250454 Regards, -- Răzvan Crainea OpenSIPS Developer http://www.opensips-solutions.com On 05/23/2012 04:11 PM, dpa wrote: Hello! Opensips 1.6.4-2 There is one question. rtpp_notify_socket may be only unix socket? Or I can use UPD or TCP too? Thank you for any help ___ 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 image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy and RTP timeout
I try to test it and after 180 s. (I didn`t change default value) dialog was being in active state. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Saul Ibarra Corretge Sent: Monday, May 14, 2012 12:21 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Mediaproxy and RTP timeout On May 14, 2012, at 9:23 AM, dpa wrote: I understand but how mediaproxy should notice dialog module to terminate call? And can I configure timeout for RTP missing? MediaProxy does that internally through the management interface. Look on the sample config.ini file for directions on how to configure the timeout (by default is 180 seconds). Regards, -- Saúl Ibarra Corretgé AG Projects ___ 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] Mediaproxy and RTP timeout
I understand but how mediaproxy should notice dialog module to terminate call? And can I configure timeout for RTP missing? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Georgescu Sent: Sunday, May 13, 2012 11:19 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Mediaproxy and RTP timeout There is nothing to set in mediaproxy. The dialog module takes care of terminating the dialog. Adrian On May 12, 2012, at 8:30 AM, dpa wrote: Thank you for your reply What should I do to make it working? I have tested the feature by emulating situation when there were no RTP streams neither from caller nor from callee and there was nothing, dialog was in active state. Thank toy From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Georgescu Sent: Friday, May 11, 2012 9:38 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Mediaproxy and RTP timeout Yes, sending BYEs on RTP timeout is supported. Adrian On May 11, 2012, at 12:04 PM, dpa wrote: Hello! In rtpproxy there is a feature which allows controlling absence of RTP stream (in both directions) and signaling to Opensips to disconnect the call. Is there such feature in mediaproxy? Thank you for any help. ___ Users mailing list mailto:Users@lists.opensips.org Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users 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] Mediaproxy and RTP timeout
Thank you for your reply What should I do to make it working? I have tested the feature by emulating situation when there were no RTP streams neither from caller nor from callee and there was nothing, dialog was in active state. Thank toy From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Georgescu Sent: Friday, May 11, 2012 9:38 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Mediaproxy and RTP timeout Yes, sending BYEs on RTP timeout is supported. Adrian On May 11, 2012, at 12:04 PM, dpa wrote: Hello! In rtpproxy there is a feature which allows controlling absence of RTP stream (in both directions) and signaling to Opensips to disconnect the call. Is there such feature in mediaproxy? Thank you for any help. ___ Users mailing list mailto:Users@lists.opensips.org Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users 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] Mediaproxy and RTP timeout
Hello! In rtpproxy there is a feature which allows controlling absence of RTP stream (in both directions) and signaling to Opensips to disconnect the call. Is there such feature in mediaproxy? Thank you for any help. image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.6.4.2 and some problem
Hello! Once I found that Opensips does not response to any message. Rather it response but with great delay (debug log is in attachment. Debug log has been done using ngrep utility). 1.1.1.1- Some UAC 1.1.1.2- Opensips 1.1.1.3- Some UAS During this problem I found errors in opensips log (debug level = 3) CRITICAL:tm:set_timer: set_timer for 1 list called on a detached timer -- ignoring: 0x7ffd073d5968 And the part 0x7ffd073d5968 is different in each message. And one time I saw ERROR:dialog:dlg_onroute: failed to update dialog lifetime. And many messages CRITICAL:dialog:log_next_state_dlg: bogus event 7 in state 1 for dlg 0x7ffd08c2f1c8 [3221:2137264810] with clid '21009543-8F6F11E1-A7309929-19A1016D' and tags '221D86F4-1EA3' 'NULL' Besides, during problem I saw that Opensips disconnects some calls (see attachment) using default dialog timeout (5 sec.) The problem disappeared without anything action from me. Thank you for any help. image001.gifU 2012/04/27 10:59:18.690929 1.1.1.1:62160 - 1.1.1.2:5060 INVITE sip:74997033315@1.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bKC4DEF61640. From: sip:929530@1.1.1.1;tag=4202CB20-9D. To: sip:74997033315@1.1.1.2. Date: Sun, 07 Sep 2003 03:33:12 GMT. Call-ID: D8179B0E-E01A11D7-BE26E851-360FA078@1.1.1.1. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 3625385702-3759804887-3051749410-2438464902. User-Agent: Cisco-SIPGateway/IOS-12.x. Accept-Language: ru. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 15. Timestamp: 1062905592. Contact: sip:929530@1.1.1.1:5060. Expires: 60. Allow-Events: telephone-event. P-Asserted-Identity: sip:929530@1.1.1.1. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 310. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1053 1891 IN IP4 1.1.1.1. s=SIP Call. c=IN IP4 1.1.1.1. t=0 0. m=audio 17534 RTP/AVP 8 18 0 101. c=IN IP4 1.1.1.1. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2012/04/27 10:59:19.188543 1.1.1.1:62160 - 1.1.1.2:5060 INVITE sip:74997033315@1.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bKC4DEF61640. From: sip:929530@1.1.1.1;tag=4202CB20-9D. To: sip:74997033315@1.1.1.2. Date: Sun, 07 Sep 2003 03:33:13 GMT. Call-ID: D8179B0E-E01A11D7-BE26E851-360FA078@1.1.1.1. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 3625385702-3759804887-3051749410-2438464902. User-Agent: Cisco-SIPGateway/IOS-12.x. Accept-Language: ru. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 15. Timestamp: 1062905593. Contact: sip:929530@1.1.1.1:5060. Expires: 60. Allow-Events: telephone-event. P-Asserted-Identity: sip:929530@1.1.1.1. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 310. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1053 1891 IN IP4 1.1.1.1. s=SIP Call. c=IN IP4 1.1.1.1. t=0 0. m=audio 17534 RTP/AVP 8 18 0 101. c=IN IP4 1.1.1.1. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2012/04/27 10:59:20.188422 1.1.1.1:62160 - 1.1.1.2:5060 INVITE sip:74997033315@1.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bKC4DEF61640. From: sip:929530@1.1.1.1;tag=4202CB20-9D. To: sip:74997033315@1.1.1.2. Date: Sun, 07 Sep 2003 03:33:14 GMT. Call-ID: D8179B0E-E01A11D7-BE26E851-360FA078@1.1.1.1. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 3625385702-3759804887-3051749410-2438464902. User-Agent: Cisco-SIPGateway/IOS-12.x. Accept-Language: ru. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 15. Timestamp: 1062905594. Contact: sip:929530@1.1.1.1:5060. Expires: 60. Allow-Events: telephone-event. P-Asserted-Identity: sip:929530@1.1.1.1. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 310. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1053 1891 IN IP4 1.1.1.1. s=SIP Call. c=IN IP4 1.1.1.1. t=0 0. m=audio 17534 RTP/AVP 8 18 0 101. c=IN IP4 1.1.1.1. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2012/04/27 10:59:26.188959 1.1.1.1:62160 - 1.1.1.2:5060 INVITE sip:74997033315@1.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;x-route-tag=tgrp:TFOP;branch=z9hG4bKC4DEF61640. From: sip:929530@1.1.1.1;tag=4202CB20-9D. To: sip:74997033315@1.1.1.2. Date: Sun, 07 Sep 2003 03:33:20 GMT. Call-ID:
Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 1.8.0 - a new major release is out
Pay attention on “curses.h:31:19: error: curses.h: No such file or directory” I think You should install libncurses-dev package first From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of goup2010 Sent: Friday, March 23, 2012 11:52 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 1.8.0 - a new major release is out Hello, I try to install 1.8.0. When run make install I receive follow errors: make[1]: Entering directory `/root/opensips_1_8/menuconfig' rm -f configure rm -f cfg.o curses.o items.o commands.o menus.o parser.o main.o make[1]: Leaving directory `/root/opensips_1_8/menuconfig' make[1]: Entering directory `/root/opensips_1_8/menuconfig' gcc -g -Wall -DMENUCONFIG_CFG_PATH=\/usr/local/share/opensips//menuconfig_templ ates/\ -DMENUCONFIG_GEN_PATH=\/etc/opensips/\ -DMENUCONFIG_HAVE_SOURCES=0 - c -o cfg.o cfg.c In file included from main.h:33, from cfg.c:30: curses.h:31:19: error: curses.h: No such file or directory In file included from cfg.c:30: main.h:35: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ to ken main.h:39: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ to ken make[1]: *** [cfg.o] Error 1 make[1]: Leaving directory `/root/opensips_1_8/menuconfig' make: *** [opensipsmc] Error 2 Where is problem? Best regards, PlayMen ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users