Re: [OpenSIPS-Users] Question about error 500 only via WIFI
Hi. I found the error cause. But I still don't know why I have such issue. When I use my Internet Link (WIFI in my home office), the SIP register message is sent correctly. Like this: Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:54.233.189.46:5060;transport=UDP SIP/2.0 Method: REGISTER Request-URI: sip:54.233.189.46:5060;transport=UDP [Resent Packet: False] Message Header Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-524287-1---6dbfa766cffddeee;rport Max-Forwards: 70 Contact: To: ;transport=UDP> From: ;tag=98bfc34c Call-ID: H1E0jkwiMniiyT5az1BT7g.. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper v2.10.18.1-mod Allow-Events: presence, kpml, talk Content-Length: 0 Opensips got the message above. However, when I use the GSM mobile network (from VIVO) , some service changes the content of the SIP Register message. Like this: Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:[64:ff9b::36e9:bd2e]:5060;transport=UDP SIP/2.0 Method: REGISTER Request-URI: sip:[64:ff9b::36e9:bd2e]:5060;transport=UDP [Resent Packet: False] Message Header Via: SIP/2.0/UDP [64:ff9b::c000:4];branch=z9hG4bK-524287-1---0a8189adf6c3449a Max-Forwards: 70 Contact: To: From: ;tag=f19aea4d Call-ID: VICBinZsDk5_ZhpHGd__CQ.. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper v2.10.18.1-mod Allow-Events: presence, kpml, talk Content-Length: 0 That is why Opensips returns error 500. I guess some service changed IPv4 to something IPv6. Could it be caused by the GSM operator (VIVO) ? What should I investigage to solve this problem? Any hint will be very helpful ! Thanks alot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 De: Users em nome de Daniel Zanutti Enviado: quinta-feira, 12 de maio de 2022 11:57 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] Question about error 500 only via WIFI Olá Rodrigo, tudo bem? Saudações de São Paulo! Opensips doesn't differentiate the network, it will look just to the sip packet. I recommend you sniff through your packets and check what's different. Probably there's somenthing on opensips log you didn't get yet, recommend you take a look there first btw. About push, I think you're enable push notifications on your device, take a look: https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy Regards On Wed, May 11, 2022 at 4:13 PM Rodrigo Pimenta Carvalho mailto:pime...@inatel.br>> wrote: Hi. My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and today I turned it on again for some tests. I usually use my home office local WIFI to connect my softphones to the network and it can be all connected (online) to this SIP proxy. However, if I use the mobile network (LTE/4G) to connect the softphones to the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500 Server error occurred (7/TM)". One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If I use it, the problem is solved even while using the mobile network. What is a proxy PUSH? Why OpenSIPs return error in a case, but not in the other one? What could I do to avoid using a 'proxy PUSH'? Local WIFI and mobile network come from different carriers. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 ___ 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
Re: [OpenSIPS-Users] Question about error 500 only via WIFI
Olá Daniel. Thank you ! I will take a look. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 De: Users em nome de Daniel Zanutti Enviado: quinta-feira, 12 de maio de 2022 11:57 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] Question about error 500 only via WIFI Olá Rodrigo, tudo bem? Saudações de São Paulo! Opensips doesn't differentiate the network, it will look just to the sip packet. I recommend you sniff through your packets and check what's different. Probably there's somenthing on opensips log you didn't get yet, recommend you take a look there first btw. About push, I think you're enable push notifications on your device, take a look: https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy Regards On Wed, May 11, 2022 at 4:13 PM Rodrigo Pimenta Carvalho mailto:pime...@inatel.br>> wrote: Hi. My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and today I turned it on again for some tests. I usually use my home office local WIFI to connect my softphones to the network and it can be all connected (online) to this SIP proxy. However, if I use the mobile network (LTE/4G) to connect the softphones to the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500 Server error occurred (7/TM)". One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If I use it, the problem is solved even while using the mobile network. What is a proxy PUSH? Why OpenSIPs return error in a case, but not in the other one? What could I do to avoid using a 'proxy PUSH'? Local WIFI and mobile network come from different carriers. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 ___ 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
[OpenSIPS-Users] Question about error 500 only via WIFI
Hi. My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and today I turned it on again for some tests. I usually use my home office local WIFI to connect my softphones to the network and it can be all connected (online) to this SIP proxy. However, if I use the mobile network (LTE/4G) to connect the softphones to the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500 Server error occurred (7/TM)". One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If I use it, the problem is solved even while using the mobile network. What is a proxy PUSH? Why OpenSIPs return error in a case, but not in the other one? What could I do to avoid using a 'proxy PUSH'? Local WIFI and mobile network come from different carriers. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] What happens when the broker is suddenly interrupted (Ex: power supply interruption)?
Hi. Today a directory used by the broker was corrupted and I had to remove it to solve a problem. Before removing such directory the broker failed to start. The directory is: /var/lib/rabbitmq/mnesia/ The question is: What could cause this kind of corruption problem in a directory used by the broker? Could a interruption of power supply in my hardware cause such problem? If not, even old versions of the broker can avoid this issue? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
Ok. Thank you very much! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 17:10 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Yes, when the ACK is lost there will be retransmissions of the 200 OK. But if the ACK is being misrouted or the connectivity issue persists for too long then the ACK will never be received. Now the endpoint that did not receive the ACK *should* then send a BYE to disconnect. However, not all endpoints operate as they should at all times and we have seen this sometimes does not occur. Also, if the network connectivity issue affected both sides of the call, then the BYE will not be received either. So you are right that the problem scenario requires both the ACK and BYE to be lost/misrouted/not sent. But as I said, it doesn’t happen often and even if it does many times the “stuck” calls cause no issues. But if billing or some other reporting/analytics are being done, the stuck calls can negatively affect those results. The INVITE refresh mechanism is part of the Dialog module and can be enabled when the dialog is created [1]. [1] http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp5828384 dialog Module - opensips.org<http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp5828384> www.opensips.org The dialog module provides dialog awareness to the OpenSIPS proxy. Its functionality is to keep trace of the current dialogs, to offer information about them (like ... Thanks, Ben Newlin From: Users <users-boun...@lists.opensips.org> on behalf of Rodrigo Pimenta Carvalho <pime...@inatel.br> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Tuesday, March 27, 2018 at 1:55 PM To: OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Hi Ben. Thank you very much! I didn't realized such problems, until you explain that. I will check if my project will need the same procedure. In that case, I will study about INVITE refreshes. What I have observed in my OpenSIPS is that when a ACK is lost for a SIP OK, the callee sends SIP OK again and again. Could you point the OpenSIPS web page (from OpenSIPS documentation) that explain about INVITE refresh, please? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 14:15 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? You don’t have to read very far back in the mailing list archives to see that misrouted ACKs are a fairly common problem when implementing SIP proxies. ☺ Mishandling of the Record-Route headers is the common problem, but loss of connectivity with the far end server can occur as well. Because the INVITE transaction is completed, the TM timers will not catch this and the dialog will stay in the CONFIRMED but not ACKed state until the $DLG_timeout expires. It doesn’t happen very often at all, but if it does and the timeout is set very high then you end up with a stuck call until the timer pops. If you are doing billing on the same endpoint then you potentially end up with a very long call being billed. There are also other ways to accomplish similar safeguards as this, including OPTIONS or INVITE refreshes using the Dialog module. We are still running 1.11 in production so the INVITE refreshes were not available to us and some of our partners do not accept OPTIONS refreshes. We plan to implement the INVITE refreshes once we have completed the upgrade to 2.X. Thanks, Ben Newlin From: Users <users-boun...@lists.opensips.org> on behalf of Rodrigo Pimenta Carvalho <pime...@inatel.br> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Tuesday, March 27, 2018 at 12:57 PM To: OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Hi. Just as curiosity, what would cause an ACK lost in your system? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 11:18 Para: OpenSIPS users mailling
Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
Hi Ben. Thank you very much! I didn't realized such problems, until you explain that. I will check if my project will need the same procedure. In that case, I will study about INVITE refreshes. What I have observed in my OpenSIPS is that when a ACK is lost for a SIP OK, the callee sends SIP OK again and again. Could you point the OpenSIPS web page (from OpenSIPS documentation) that explain about INVITE refresh, please? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 14:15 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? You don’t have to read very far back in the mailing list archives to see that misrouted ACKs are a fairly common problem when implementing SIP proxies. ☺ Mishandling of the Record-Route headers is the common problem, but loss of connectivity with the far end server can occur as well. Because the INVITE transaction is completed, the TM timers will not catch this and the dialog will stay in the CONFIRMED but not ACKed state until the $DLG_timeout expires. It doesn’t happen very often at all, but if it does and the timeout is set very high then you end up with a stuck call until the timer pops. If you are doing billing on the same endpoint then you potentially end up with a very long call being billed. There are also other ways to accomplish similar safeguards as this, including OPTIONS or INVITE refreshes using the Dialog module. We are still running 1.11 in production so the INVITE refreshes were not available to us and some of our partners do not accept OPTIONS refreshes. We plan to implement the INVITE refreshes once we have completed the upgrade to 2.X. Thanks, Ben Newlin From: Users <users-boun...@lists.opensips.org> on behalf of Rodrigo Pimenta Carvalho <pime...@inatel.br> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Tuesday, March 27, 2018 at 12:57 PM To: OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Hi. Just as curiosity, what would cause an ACK lost in your system? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 11:18 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Rodrigo, Yes, they do. I am using them to do exactly what you describe. The final reply (fr) timer is how long a transaction will wait to receive a final reply (>=200). If the timer expires without receiving a final reply the transaction will be canceled and failure route will be triggered with, I think, a local 408 response. As for $DLG_timeout, you can set that value multiple times in a call. We do this as well. Prior to the call being ACKed we set this value fairly low (~5s) in order to disconnect the dialog if the ACK is lost. Once we receive the ACK, we then extend it to a much longer value. Thanks, Ben Newlin From: Users <users-boun...@lists.opensips.org> on behalf of Rodrigo Pimenta Carvalho <pime...@inatel.br> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Tuesday, March 27, 2018 at 10:08 AM To: OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Ok Ben. I will check this possibility and see if reply times will change the duration of a not answered call. Thank you. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 10:43 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? You can also use the reply timers in TM to do this: http://www.opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout tm Module - openSIPS<http://www.opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout> www.opensips.org TM module enables stateful processing of SIP transactions. The main use of stateful logic, which is costly in terms of memory and CPU, is some services inherently ... Thanks, Ben Newlin From: Users <users-boun...@lists.opensips.org> on behalf of Laszlo <las...@voipfrea
Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
Hi. Just as curiosity, what would cause an ACK lost in your system? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 11:18 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Rodrigo, Yes, they do. I am using them to do exactly what you describe. The final reply (fr) timer is how long a transaction will wait to receive a final reply (>=200). If the timer expires without receiving a final reply the transaction will be canceled and failure route will be triggered with, I think, a local 408 response. As for $DLG_timeout, you can set that value multiple times in a call. We do this as well. Prior to the call being ACKed we set this value fairly low (~5s) in order to disconnect the dialog if the ACK is lost. Once we receive the ACK, we then extend it to a much longer value. Thanks, Ben Newlin From: Users <users-boun...@lists.opensips.org> on behalf of Rodrigo Pimenta Carvalho <pime...@inatel.br> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Tuesday, March 27, 2018 at 10:08 AM To: OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Ok Ben. I will check this possibility and see if reply times will change the duration of a not answered call. Thank you. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 10:43 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? You can also use the reply timers in TM to do this: http://www.opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout tm Module - openSIPS<http://www.opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout> www.opensips.org TM module enables stateful processing of SIP transactions. The main use of stateful logic, which is costly in terms of memory and CPU, is some services inherently ... Thanks, Ben Newlin From: Users <users-boun...@lists.opensips.org> on behalf of Laszlo <las...@voipfreak.net> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Tuesday, March 27, 2018 at 9:40 AM To: OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? On Tue, Mar 27, 2018 at 3:31 PM, Rodrigo Pimenta Carvalho <pime...@inatel.br<mailto:pime...@inatel.br>> wrote: Hi. When a peer invites another one to a call, there are calling and ringing tones for these peers. My SIP agents let these tones execute during 2 minutes. After this, the call is terminated, if no one answers the call. How to configure OpenSIPS, if possible, so that any call will be terminated after 1 minute? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 Maybe you can play with $DLG_timeout, see http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#timeout-pvar-id ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
Hi Laszlo. Thank you for the reply. I'm using $DLG_timeout to configure how long a call will be, after answered. If I change $DLG_timeout, the duration of an answered call will change too. I have to avoid changing this way. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Laszlo <las...@voipfreak.net> Enviado: terça-feira, 27 de março de 2018 10:39 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? On Tue, Mar 27, 2018 at 3:31 PM, Rodrigo Pimenta Carvalho <pime...@inatel.br<mailto:pime...@inatel.br>> wrote: Hi. When a peer invites another one to a call, there are calling and ringing tones for these peers. My SIP agents let these tones execute during 2 minutes. After this, the call is terminated, if no one answers the call. How to configure OpenSIPS, if possible, so that any call will be terminated after 1 minute? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 Maybe you can play with $DLG_timeout, see http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#timeout-pvar-id ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
Hi. When a peer invites another one to a call, there are calling and ringing tones for these peers. My SIP agents let these tones execute during 2 minutes. After this, the call is terminated, if no one answers the call. How to configure OpenSIPS, if possible, so that any call will be terminated after 1 minute? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to avoid/solve TCP blocked connection?
Hi. My softphone is registered with the following AOR: - AOR:: g1r2u3p4o5 Contact:: sip:g1r2u3p4o5@127.0.0.1:50353;transport=TLS;ob Q= Expires:: 10 Callid:: 53e387dc-81fe-45f9-a6f1-8a5cf4248d62 Cseq:: 37190 User-agent:: n/a Received:: sip:127.0.0.1:49678;transport=TLS State:: CS_SYNC Flags:: 0 Cflags:: NAT Socket:: tls:127.0.0.1:5061 Methods:: 8063 Attr:: in_same_network SIP_instance:: - Sometimes, when it receives a call and answers, the Opensips show the following error: [21532] INFO:core:probe_max_sock_buff: using snd buffer of 320 kb [21532] INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 22 [21532] ERROR:core:tcp_connect_blocking: poll error: flags 28 - 4 8 16 32 [21532] ERROR:core:tcp_connect_blocking: failed to retrieve SO_ERROR [server=127.0.0.1:50353] (111) Connection refused [21532] ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed [21532] ERROR:proto_tls:proto_tls_send: connect failed [21532] ERROR:core:msg_send: send() for proto 3 failed Is there a way to avoid this kind of problem? That is, can I configure the OpenSIPS to renew some TCP connection? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How can we avoid log completely ?
Hi. How can we avoid log completely? That is, how to configure the opensips.cfg file so that it will no more generate any log ? Should I remove something from the cfg file? Should I put something in such file? Can we avoid logs completely even when the cfg file still has xlog commands? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Is OpenSIPS fast to stop calling tone in forked calls?
Hi. When my OpenSIPS has just 2 peers on-line (user A and user B ), if A calls B, A will listen the calling tone just until B answers. So, if B sends a SIP OK, the calling tone will be immediately stopped. When user B is on-line in several devices (more than 1 contact for the same AOR), let's say 2 mobile phones and 2 desktops, 4 devices will ring and the user A will listen the calling tone normally. But, when B answers (in any device), there will be some calling tones still to be played in the A's device. That is, when the number of called devices increases for a same called subscriber, it seems that OpenSIPS become slower to stop the calling tone. But it is fast enough to stop the ring tones in the others devices. Is it a matter of OpenSIPS configurations file? (opensips.cfg) If yes, could someone point me what part of my configuration should I change or review? Any hint will be very helpful! Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Table location and command opensipsctl ul show.
Ok. Thank you very much! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org> Enviado: quinta-feira, 2 de novembro de 2017 11:27 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] Table location and command opensipsctl ul show. Hi, Rodrigo! The interaction between OpenSIPS memory and database is described here[1]. Depending on your db_mode configuration, you might find entries that are not yet synced in the database. [1] http://www.opensips.org/html/docs/modules/2.4.x/usrloc.html#idp5672576 usrloc Module - opensips.org<http://www.opensips.org/html/docs/modules/2.4.x/usrloc.html#idp5672576> www.opensips.org How the contacts are matched (for same AOR - Address of Record) is an important aspect of the usrloc modules, especialy in the context of NAT traversal - this raise ... Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com<http://www.opensips-solutions.com> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 11/02/2017 02:33 PM, Rodrigo Pimenta Carvalho wrote: Hi. When I delete all registers from table location, I still can see registers via command 'opensipsctl ul show'. Why a empty table location doesn't gives a null result in 'opensipsctl ul show' ? Is it caused by the configuration from opensips.cfg ? Thanks. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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
[OpenSIPS-Users] Table location and command opensipsctl ul show.
Hi. When I delete all registers from table location, I still can see registers via command 'opensipsctl ul show'. Why a empty table location doesn't gives a null result in 'opensipsctl ul show' ? Is it caused by the configuration from opensips.cfg ? Thanks. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to recover from tcp_blocking_connect failed ?
Dear OpenSIPS users, According to the OpenSIPS log, a callee answered a caller with SIP OK. Such SIP OK has the following contact: Contact: "Group" <sip:g1r2u3p4o5@127.0.0.1:40348;transport=TLS;ob>;+sip.ice However, it seems that there is no available connection to port 40348. That is, opensips cann't send any message to that port. So, after opensips relaying a SIP ACK to g1r2u3p4o5, it caused the following error: Jan 09 17:48:54 colibri-imx6 opensips[23115]: Jan 9 17:48:54 [23135] INFO:core:probe_max_sock_buff: using snd buffer of 320 kb Jan 09 17:48:54 colibri-imx6 opensips[23115]: Jan 9 17:48:54 [23135] INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 22 Jan 09 17:48:54 colibri-imx6 opensips[23115]: Jan 9 17:48:54 [23135] ERROR:core:tcp_connect_blocking: poll error: flags 28 - 4 8 16 32 Jan 09 17:48:55 colibri-imx6 opensips[23115]: Jan 9 17:48:54 [23135] ERROR:core:tcp_connect_blocking: failed to retrieve SO_ERROR [server=127.0.0.1:40348] (111) Connection refused Jan 09 17:48:55 colibri-imx6 opensips[23115]: Jan 9 17:48:54 [23135] ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed Jan 09 17:48:55 colibri-imx6 opensips[23115]: Jan 9 17:48:54 [23135] ERROR:proto_tls:proto_tls_send: connect failed Jan 09 17:48:55 colibri-imx6 opensips[23115]: Jan 9 17:48:54 [23135] ERROR:core:msg_send: send() for proto 3 failed What is happening here? How can a peer lost its connection or lost connect to opensips? How to avoid this issue or recover from it? I have no idea on what to do. Any hint will be very helpful! Thanks a lot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] $(avp(attr)[$T_branch_idx]) is NULL for second INVITE. How to solve it ?
Hi.
Very good.
Thank you very much!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: Users <users-boun...@lists.opensips.org> em nome de Liviu Chircu
<li...@opensips.org>
Enviado: quarta-feira, 30 de agosto de 2017 06:44
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] $(avp(attr)[$T_branch_idx]) is NULL for second
INVITE. How to solve it ?
You could persist it at dialog level, once the 200 OK reply arrives in an
onreply_route, like so:
onreply_route [store_attr] {
$dlg_val(callee_attr) = $(avp(attr)[$T_branch_idx])
}
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
[http://www.opensips-solutions.com/imgs/slideshow/slide1.jpg]<http://www.opensips-solutions.com/>
Home — OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 29.08.2017 20:33, Rodrigo Pimenta Carvalho wrote:
Hi Liviu.
Thank you for your reply!
I guess the second INVITE is a Re-INVITE, as you commented.
In this case, is it possible to keep stored the value of $avp(attr) and use it
when necessary even after receiving the Re-INVITE?
--
Yes we have a retry, not a parallel forked call, but just when the Re-INVITE is
received by OpenSIPS. By other side, the first INVITE is for a parallel forked
call, if I'm well understanding the SIP here.
The first INVITE is:
SIP Message: INVITE sip:g1r2u3p4o5@127.0.0.1 SIP/2.0
Via: SIP/2.0/TLS
127.0.0.1:42194;rport;branch=z9hG4bKPjd3128578-0158-4c58-8c1c-676aa864d8ca;alias
Max-Forwards: 70
From: "ext1"
<sip:intercomB_RYg7tf4xx6JV@127.0.0.1><sip:intercomB_RYg7tf4xx6JV@127.0.0.1>;tag=469b3a48-1548-4af6-8e03-a5c855ce85f4
To: <sip:g1r2u3p4o5@127.0.0.1><sip:g1r2u3p4o5@127.0.0.1>
Contact: "ext1"
<sip:intercomB_RYg7tf4xx6JV@127.0.0.1:42194;transport=TLS;ob><sip:intercomB_RYg7tf4xx6JV@127.0.0.1:42194;transport=TLS;ob>;+sip.ice
Call-ID: 22cb74fc-3d3b-4a37-9572-32f48c9943ff
CSeq: 21431 INVITE
Route:
<sip:intercomB_RYg7tf4xx6JV@127.0.0.1;transport=tls;lr><sip:intercomB_RYg7tf4xx6JV@127.0.0.1;transport=tls;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 1267
There is more than one registered (on-line) subscriber g1r2u3p4o5. So,
g1r2u3p4o5 has more than one AOR. I have 3 devices online for the subscriber
g1r2u3p4o5.
The reply SIP OK comes from another network (not the local one), from IP
10.0.60.246. After such reply, the Re-INVITE is:
SIP Message: INVITE sip:g1r2u3p4o5@10.0.60.246:59673;transport=TLS SIP/2.0
Via: SIP/2.0/TLS
192.168.0.81:54188;rport;branch=z9hG4bKPjdde63995-7ed0-436a-983f-61d0e5df9498;alias
Max-Forwards: 70
From: "ext1"
<sip:intercomB_RYg7tf4xx6JV@127.0.0.1><sip:intercomB_RYg7tf4xx6JV@127.0.0.1>;tag=469b3a48-1548-4af6-8e03-a5c855ce85f4
To: <sip:g1r2u3p4o5@127.0.0.1><sip:g1r2u3p4o5@127.0.0.1>;tag=393a402c
Contact: "ext1"
<sip:intercomB_RYg7tf4xx6JV@127.0.0.1:42194;transport=TLS;ob><sip:intercomB_RYg7tf4xx6JV@127.0.0.1:42194;transport=TLS;ob>;+sip.ice
Call-ID: 22cb74fc-3d3b-4a37-9572-32f48c9943ff
CSeq: 21433 INVITE
Route:
<sip:icchw.jflddns.com.br:5061;transport=tls;lr;did=d53.353e0122><sip:icchw.jflddns.com.br:5061;transport=tls;lr;did=d53.353e0122>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length: 332
In this moment I need to know if the device (callee) is in another network, in
fact, to take some fixes in SDP of INVITEs and SIP OKs.
The $(avp(attr)[$T_branch_idx]) should have the information that I need.
If it is not possible to keep the $(avp(attr)[$T_branch_idx]) stored, is it
possible to know if a device is in another network when it is a callee?
Any hint will be very helpful !!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: Users
<users-boun...@lists.opensips.org><mailto:users-boun...@lists.opensips.org> em
nome de Liviu Chircu <li...@opensips.org><mailto:li...@opensips.org>
Enviado: terça-feira, 29 de agosto de 2017 12:34
Para: users@lists.opensips.org<mailto:users@lists.opensips.org>
Assunto: Re: [OpenSIPS-Users] $(avp(attr)[$T_branch_idx]) is NULL for second
INVITE. How to
Re: [OpenSIPS-Users] $(avp(attr)[$T_branch_idx]) is NULL for second INVITE. How to solve it ?
Hi Liviu.
Thank you for your reply!
I guess the second INVITE is a Re-INVITE, as you commented.
In this case, is it possible to keep stored the value of $avp(attr) and use it
when necessary even after receiving the Re-INVITE?
--
Yes we have a retry, not a parallel forked call, but just when the Re-INVITE is
received by OpenSIPS. By other side, the first INVITE is for a parallel forked
call, if I'm well understanding the SIP here.
The first INVITE is:
SIP Message: INVITE sip:g1r2u3p4o5@127.0.0.1 SIP/2.0
Via: SIP/2.0/TLS
127.0.0.1:42194;rport;branch=z9hG4bKPjd3128578-0158-4c58-8c1c-676aa864d8ca;alias
Max-Forwards: 70
From: "ext1"
<sip:intercomB_RYg7tf4xx6JV@127.0.0.1>;tag=469b3a48-1548-4af6-8e03-a5c855ce85f4
To: <sip:g1r2u3p4o5@127.0.0.1>
Contact: "ext1"
<sip:intercomB_RYg7tf4xx6JV@127.0.0.1:42194;transport=TLS;ob>;+sip.ice
Call-ID: 22cb74fc-3d3b-4a37-9572-32f48c9943ff
CSeq: 21431 INVITE
Route: <sip:intercomB_RYg7tf4xx6JV@127.0.0.1;transport=tls;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 1267
There is more than one registered (on-line) subscriber g1r2u3p4o5. So,
g1r2u3p4o5 has more than one AOR. I have 3 devices online for the subscriber
g1r2u3p4o5.
The reply SIP OK comes from another network (not the local one), from IP
10.0.60.246. After such reply, the Re-INVITE is:
SIP Message: INVITE sip:g1r2u3p4o5@10.0.60.246:59673;transport=TLS SIP/2.0
Via: SIP/2.0/TLS
192.168.0.81:54188;rport;branch=z9hG4bKPjdde63995-7ed0-436a-983f-61d0e5df9498;alias
Max-Forwards: 70
From: "ext1"
<sip:intercomB_RYg7tf4xx6JV@127.0.0.1>;tag=469b3a48-1548-4af6-8e03-a5c855ce85f4
To: <sip:g1r2u3p4o5@127.0.0.1>;tag=393a402c
Contact: "ext1"
<sip:intercomB_RYg7tf4xx6JV@127.0.0.1:42194;transport=TLS;ob>;+sip.ice
Call-ID: 22cb74fc-3d3b-4a37-9572-32f48c9943ff
CSeq: 21433 INVITE
Route: <sip:icchw.jflddns.com.br:5061;transport=tls;lr;did=d53.353e0122>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length: 332
In this moment I need to know if the device (callee) is in another network, in
fact, to take some fixes in SDP of INVITEs and SIP OKs.
The $(avp(attr)[$T_branch_idx]) should have the information that I need.
If it is not possible to keep the $(avp(attr)[$T_branch_idx]) stored, is it
possible to know if a device is in another network when it is a callee?
Any hint will be very helpful !!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: Users <users-boun...@lists.opensips.org> em nome de Liviu Chircu
<li...@opensips.org>
Enviado: terça-feira, 29 de agosto de 2017 12:34
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] $(avp(attr)[$T_branch_idx]) is NULL for second
INVITE. How to solve it ?
Hi Rodrigo,
Just trying to understand the flow here - could it be actually a Re-INVITE that
goes through your sequential routing block, thus lookup() is not called,
leaving $avp(attr) NULL throughout that transaction?
Regardless of the above, in OpenSIPS terms, each "branch" points to a different
destination. In our case, we're talking about a retry, not a serial/parallel
forked call. Which means that you should only bother with $T_branch_idx if a
lookup() could yield more than one device to be contacted for the same AoR.
Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
[http://www.opensips-solutions.com/imgs/slideshow/slide1.jpg]<http://www.opensips-solutions.com/>
Home — OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 29.08.2017 18:05, Rodrigo Pimenta Carvalho wrote:
Dear SIPusers,
In my project I use to keep a valuable information in table location. This is
about the state of a subscriber's localization.
I have to read such information for the callees, every time a new branch is
created and every time a INVITE is answered with SIP OK.
So, my OpenSIPS configuration has something similar to the following code:
1route{
2 ... // hidden code for simplification.
3 lookup("location","m")
4 ...
5 route(relay);
6}
7route[relay]{
8 if (is_method("INVITE")) {
9
[OpenSIPS-Users] $(avp(attr)[$T_branch_idx]) is NULL for second INVITE. How to solve it ?
Dear SIPusers,
In my project I use to keep a valuable information in table location. This is
about the state of a subscriber's localization.
I have to read such information for the callees, every time a new branch is
created and every time a INVITE is answered with SIP OK.
So, my OpenSIPS configuration has something similar to the following code:
1route{
2 ... // hidden code for simplification.
3 lookup("location","m")
4 ...
5 route(relay);
6}
7route[relay]{
8 if (is_method("INVITE")) {
9...
10t_on_branch("per_branch_ops");
11t_on_reply("handle_nat");
12t_on_failure("missed_call");
13 }
14 ...
15 }
16branch_route[per_branch_ops] {
17
18$(avp(attr)[$T_branch_idx])
19...
20}
21onreply_route[handle_nat] {
22...
23$(avp(attr)[$T_branch_idx])
24...
25}
26...
In a determined call, when the OpenSIPS receives the INVITE and then a SIP OK
(200), the code gets right value in lines 18 and 23.
In such call, the SIP OK (from callee) offers a kind of video that the caller
can't support. In this case the caller sends another SIP INVITE with inactive
video (SDP).
In this moment, OpenSIPS gets this second INVITE and create a new branch.
However, for this new branch, lines 18 and 23 give me NULL for
$(avp(attr)[$T_branch_idx]).
How to solve this issue?
Any hint will be very helpful!!
Best regards!
P.S.: I'm not expert in SIP.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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[OpenSIPS-Users] How sequential forking works with OpenSIPS.
Dear OpenSIPS users, When sequential forking is used by means of OpenSIS, 1 - can we have every callee ringing simultaneously in sometime, when nobody answers the call? Or 2 - each callee will ring only after a previous one stop ringing if the call was not answered until that moment? If the answer is number 2, is it possible to change this behavior to put all callees ringing at same time in some moment? I would like to experiment sequential forking, just to see if helps me to avoid an UAC to mute dialogs (vide RFC 3960) when it receives multiple SIP 183 due to a parallel forking. Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Buffer size exceeded.
Ok. I got the point. Thank you very much! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Liviu Chircu <li...@opensips.org> Enviado: terça-feira, 27 de junho de 2017 03:11 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] Buffer size exceeded. Hi Rodrigo, By enabling "sip_warning", OpenSIPS will try to append a custom "Warning:" header field to all SIP messages it generates/proxies. As implemented, it's more of a debugging mechanism, as the header data will consist of strings such as SIP URIs involved, source/dest IPs, etc. In your case, this debugging info couldn't fit into 256 bytes, so OpenSIPS skipped appending the "Warning" for that message. Hardly anything to worry about. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com [http://www.opensips-solutions.com/imgs/slideshow/slide1.jpg]<http://www.opensips-solutions.com/> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 26.06.2017 23:04, Rodrigo Pimenta Carvalho wrote: Hi. What does mean the error: ERROR:core:warning_builder: buffer size exceeded ? How to avoid it? Thanks a lot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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
[OpenSIPS-Users] Buffer size exceeded.
Hi. What does mean the error: ERROR:core:warning_builder: buffer size exceeded ? How to avoid it? Thanks a lot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 2.2 changes CSeq numbers after SIP OPTIONS received. Is it a bug?
|<---| | CSeq = 2 | | |<---| | | ACK | | | CSeq = 2 | | |--->|ACK| | |CSeq = 3| | |--->| | | | | | SIP OK | | | CSeq = 2 | | |<---| <<--- This SIP OK never receives a SIP ACK with CSeq = 2...and the problem continues. Any hint will be very helpful!! Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS migth be changing CSeq numbers after SIP OPTIONS. How to avoid it?
|<---| | CSeq = 2 | | |<---| | | ACK | | | CSeq = 2 | | |--->|ACK| | |CSeq = 3| | |--->| | | | | | SIP OK | | | CSeq = 2 | | |<---| <<--- This SIP OK never receives a SIP ACK with CSeq = 2...and the problem continues. Any hint will be very helpful!! Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to see SIP messages in the log ?
Hi. Is it possible to see SIP messages in the OpenSIPS log ? Should I use some specific configuration in my opensips.cfg file? I would like tho see the entire SIP messages that is received and sent by the OpenSIPS. And, if peers are using TLS, is it still possible to see SIP messages from OpenSIPS? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to know wich UAS is rejecting a call?
Hi. In my system there are 2 UA (in different machines, M1 & M2) registered on my OpenSIPS, both as user 9000. So, if an UAC calls number 9000, these 2 UA will ring. That is fine for my project. If UA from machine M1 rejects the call it send a SIP 486 code (busy here). The UA from machine M2 has the same behavior. In addiction, if UA on M1 rejects the call, OpenSIPS must register data in the database. On the other hand, if UA on M2 rejects the call, the OpenSIPS must do nothing. How can I know wich of these reject messages (SIP 486) is coming from M1 or M2, by means of OpenSIPS functions? Any example? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to now if Cflags == NAT ?
Dear OpenSIPS users,
Every time a SIP UAC registers in my OpenSIPS, if such agent is behind a NAT,
the location record receives the NAT flag in the Cflags column. My Opensips
configuration file is responsible for that.
For example:
AOR:: 1000
Contact:: sip:1000@192.168.21.5:59047;transport=TLS;ob Q=
Expires:: 298
Callid:: ec88647d89564d9b8cf112cb254d0f04
Cseq:: 34066
User-agent:: MicroSIP/3.11.0
Received:: sip::59047;transport=TLS
State:: CS_SYNC
Flags:: 0
Cflags:: NAT <<=== HERE IS THE NAT FLAG.
Now, if a SIP INVITE arrives in my OpenSIPS, from a UAC that is behind a NAT, I
need do something like this:
if (Cflags == NAT) {
fix_nated_sdp("10")
};
What is the correct way to investigate if Cflags is equals to NAT ? I mean, how
to program such check in the script file?
Any hint will be very helpful!
Best regards!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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[OpenSIPS-Users] What does mean 'methods' in table location?
Hi. In my table location (sqlite data base) I have the column 'methods'. What does mean such column ? In this table I have: AOR:: 1 Contact:: sip:1@192.168.0.85:50356;transport=TLS;rinstance=87cc2284712a216c Q= Expires:: 60 Callid:: SYWM-mDwAojYlNfao6pTIQ.. Cseq:: 6 User-agent:: Zoiper rv2.8.15 State:: CS_SYNC Flags:: 0 Cflags:: Socket:: tls:192.168.0.84:5061 Methods:: 4294967295 <-- Attr:: in_same_network AOR:: 1000 Contact:: sip:1000@192.168.0.85:41170;transport=TLS Q= Expires:: 374 Callid:: 14e2707516170165935565k6650rmwp Cseq:: 10622 User-agent:: MizuDroid/2.0.2 State:: CS_SYNC Flags:: 0 Cflags:: Socket:: tls:192.168.0.84:5061 Methods:: 8063 < Attr:: in_same_networ What does means those different values of 'Methods' ? By the way, what does mean that 'rinstance' for the AOR 1 ? Right now, the contact 1000 can talk => rtp packets can be sent to and from it. However, contact 1 can receive rtp packets, but not send. Can it has some relation to those values? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is there new information about "WARNING ...tm-utimer...delay in execution" nowadays ?
Hi Bogdan. Thank you for answering the questions. In fact, today I moved my project to another hardware (2 CPUs and 516 MByte of RAM) and then I was able to see that OpenSIPS is working very well spending almost 0% of the CPUs. So there is no problem with the script opensips.cfg. On the other hand, we have a sip client (softphone implemented by us) that even on this new environment it spends almost 100% of CPUs during calls. So, this softphone has a problem that was preventing the OpenSIPS to use the CPU in a normal way, as I suppose. In this case, I will focus my attention in such softphone for a while. Hopefully, by solving such issue, the entire system will run well even with 1 CPU and 256 MByte of RAM. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Bogdan-Andrei Iancu <bog...@opensips.org> Enviado: quarta-feira, 22 de março de 2017 17:38 Para: OpenSIPS users mailling list; Rodrigo Pimenta Carvalho Assunto: Re: [OpenSIPS-Users] Is there new information about "WARNING ...tm-utimer...delay in execution" nowadays ? Hi Rodrigo, The issue you are reporting it is not the real problem, but a side effect of it. As you noticed, when opensips is under heavy load (CPU?), the internal timer system starts to generate warnings you see. Now, the questions is: why is your opensips using 100% or why is it blocked (no processes available). Do you have any input on this ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit May 2017 Amsterdam http://www.opensips.org/events/Summit-2017Amsterdam.html [http://www.opensips.org/events/img/conference-image-2.jpg]<http://www.opensips.org/events/Summit-2017Amsterdam.html> OpenSIPS Summit 2nd-5th May 2017, Amsterdam<http://www.opensips.org/events/Summit-2017Amsterdam.html> www.opensips.org OPENSIPS Summit 2017 "Great minds have purposes; others have wishes" Join us for three exciting days filled with VoIP and RTC presentations, workshops and design ... [http://www.opensips-solutions.com/imgs/slideshow/slide1.jpg]<http://www.opensips-solutions.com/> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 03/20/2017 09:10 PM, Rodrigo Pimenta Carvalho wrote: Hi. I have seen again that behavior from OpenSIPS that generates lots of warnings, like below: Jan 01 06:19:08 colibri-imx6 opensips[1785]: Jan 1 06:19:08 [1792] WARNING:core:utimer_ticker: utimer task already scheduled for 21873780 ms (now 21873970 ms), it may overlap.. Jan 01 06:19:08 colibri-imx6 opensips[1785]: Jan 1 06:19:08 [1792] WARNING:core:utimer_ticker: utimer task already scheduled for 21873990 ms (now 21873990 ms), it may overlap.. Jan 01 06:19:08 colibri-imx6 opensips[1785]: Jan 1 06:19:08 [1793] WARNING:core:handle_timer_job: utimer job has a 19 us delay in execution Jan 01 06:19:26 colibri-imx6 opensips[1785]: Jan 1 06:19:26 [1792] WARNING:core:utimer_ticker: utimer task already scheduled for 0 ms (now 21891940 ms), it may overlap.. Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1792] WARNING:core:utimer_ticker: utimer task already scheduled for 21908780 ms (now 21909000 ms), it may overlap.. Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1792] WARNING:core:utimer_ticker: utimer task already scheduled for 21909010 ms (now 21909010 ms), it may overlap.. Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1794] WARNING:core:handle_timer_job: utimer job has a 22 us delay in execution Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1797] WARNING:core:handle_timer_job: timer job has a 22 us delay in execution Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1795] WARNING:core:handle_timer_job: timer job has a 22 us delay in execution Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1793] WARNING:core:handle_timer_job: timer job has a 23 us delay in execution Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan 1 06:19:49 [1798] WARNING:core:handle_timer_job: utimer job has a 37 us delay in execution Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan 1 06:19:49 [1792] WARNING:core:utimer_ticker: utimer task already scheduled for 21914930 ms (now 21915300 ms), it may overlap.. Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan 1 06:19:49 [1792] WARNING:core:utimer_ticker: utimer task already scheduled for 21915320 ms (now 21915320 ms), it may overlap.. Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan 1 06:19:49 [1794] WARNING:core:handle_timer_job: utimer job has a 3 us delay in execution Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan
Re: [OpenSIPS-Users] OpenSIPS and 256 MByte of RAM.
Ok Răzvan. Thank you very much! I will take a look. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users <users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org> Enviado: quarta-feira, 22 de março de 2017 10:24 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] OpenSIPS and 256 MByte of RAM. Hi, Rodrigo! OpenSIPS by itself performs very well :). However, in a call, OpenSIPS is not all by itself: you need SIP clients, databases, DNS, and so forth and so on. In order to have better performance, you have to increase performance for each of these components. So you need to define what exactly is drawing OpenSIPS back: is it the processing itself, is it the database, is it the DNS and so on. In order to pinpoint the components that have poor performance, OpenSIPS provides some thresholds that you can use to measure different stuff in the script, such as DNS queries[1], message processing [2] or mysql queries[3]. You should try to profile each of these and figure out what exactly is happening in your platform, to find out what exactly you can/should improve. [1] http://www.opensips.org/Documentation/Script-CoreParameters-2-3#toc59 openSIPS | Documentation / Core Parameters - 2.3<http://www.opensips.org/Documentation/Script-CoreParameters-2-3#toc59> www.opensips.org 3. Core parameters. Global parameters that can be set in configuration file. Accepted values are, depending on the actual parameters strings, numbers and yes/ no. [2] http://www.opensips.org/Documentation/Script-CoreParameters-2-3#toc60 openSIPS | Documentation / Core Parameters - 2.3<http://www.opensips.org/Documentation/Script-CoreParameters-2-3#toc60> www.opensips.org 3. Core parameters. Global parameters that can be set in configuration file. Accepted values are, depending on the actual parameters strings, numbers and yes/ no. [3] http://www.opensips.org/html/docs/modules/2.3.x/db_mysql.html#id248019 mysql Module - opensips.org<http://www.opensips.org/html/docs/modules/2.3.x/db_mysql.html#id248019> www.opensips.org This is a module which provides MySQL connectivity for OpenSIPS. It implements the DB API defined in OpenSIPS. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> [http://www.opensips-solutions.com/imgs/slideshow/slide1.jpg]<http://www.opensips-solutions.com/> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 03/22/2017 03:12 PM, Rodrigo Pimenta Carvalho wrote: Hi. I was getting some problems with OpenSIPS and its performance in a hardware with 1 CPU and 256 MByte of RAM. When I moved my OpenSIPS to another hardware with 2 CPUs and 516 MByte of RAM it became working very well ! But, I have to move back my OpenSIPS to the 'poor' hardware! So, what configurations in OpenSIPS files should I set, to get better performance? Any suggestion will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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
[OpenSIPS-Users] OpenSIPS and 256 MByte of RAM.
Hi. I was getting some problems with OpenSIPS and its performance in a hardware with 1 CPU and 256 MByte of RAM. When I moved my OpenSIPS to another hardware with 2 CPUs and 516 MByte of RAM it became working very well ! But, I have to move back my OpenSIPS to the 'poor' hardware! So, what configurations in OpenSIPS files should I set, to get better performance? Any suggestion will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Is there new information about "WARNING ...tm-utimer...delay in execution" nowadays ?
Hi.
I have seen again that behavior from OpenSIPS that generates lots of warnings,
like below:
Jan 01 06:19:08 colibri-imx6 opensips[1785]: Jan 1 06:19:08 [1792]
WARNING:core:utimer_ticker: utimer task already scheduled for
21873780 ms (now 21873970 ms), it may overlap..
Jan 01 06:19:08 colibri-imx6 opensips[1785]: Jan 1 06:19:08 [1792]
WARNING:core:utimer_ticker: utimer task already scheduled for
21873990 ms (now 21873990 ms), it may overlap..
Jan 01 06:19:08 colibri-imx6 opensips[1785]: Jan 1 06:19:08 [1793]
WARNING:core:handle_timer_job: utimer job has a 19 us delay in
execution
Jan 01 06:19:26 colibri-imx6 opensips[1785]: Jan 1 06:19:26 [1792]
WARNING:core:utimer_ticker: utimer task already scheduled for 0 ms
(now 21891940 ms), it may overlap..
Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1792]
WARNING:core:utimer_ticker: utimer task already scheduled for
21908780 ms (now 21909000 ms), it may overlap..
Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1792]
WARNING:core:utimer_ticker: utimer task already scheduled for
21909010 ms (now 21909010 ms), it may overlap..
Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1794]
WARNING:core:handle_timer_job: utimer job has a 22 us delay in
execution
Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1797]
WARNING:core:handle_timer_job: timer job has a 22 us delay in
execution
Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1795]
WARNING:core:handle_timer_job: timer job has a 22 us delay in
execution
Jan 01 06:19:43 colibri-imx6 opensips[1785]: Jan 1 06:19:43 [1793]
WARNING:core:handle_timer_job: timer job has a 23 us delay
in execution
Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan 1 06:19:49 [1798]
WARNING:core:handle_timer_job: utimer job has a 37 us delay in
execution
Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan 1 06:19:49 [1792]
WARNING:core:utimer_ticker: utimer task already scheduled for
21914930 ms (now 21915300 ms), it may overlap..
Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan 1 06:19:49 [1792]
WARNING:core:utimer_ticker: utimer task already scheduled for
21915320 ms (now 21915320 ms), it may overlap..
Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan 1 06:19:49 [1794]
WARNING:core:handle_timer_job: utimer job has a 3 us delay in
execution
Jan 01 06:19:49 colibri-imx6 opensips[1785]: Jan 1 06:19:49 [1795]
WARNING:core:handle_timer_job: utimer job has a 3 us delay in
execution
When it happens, I can see that OpenSIPS is using the CPU almost 100% of the
time. And such behavior prevents others softwares in my system to work without
problems. I see 6 process with 'OpenSIPS name and each one using 11% of CPU,
for example. Now, the unique solution is to reboot the system. Otherwise, the
system remains instable and OpenSIPS continues using the CPU much more than
usual.
Is there some new information about such issue that I should to know nowadays?
Is my hardware under minimals requirements to run OpenSIPS?
Is my script opensips.cfg wrong?
My system has the following characteristics:
CPU clock = 996000
CPU model name= ARMv7 Processor rev 10 (v7l)
Hardware= Freescale i.MX6 Quad/DualLite (Device Tree)
total used free sharedbuffers
cached
Mem:251140 157208 93932 0196 26304
In my script opensips.cfg I have:
---
tcp_children=4
tcp_keepalive = 1
children=4
#fork=no
auto_aliases=no
Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 90)
modparam("tm", "fr_inv_timeout", 120)
modparam("tm", "T1_timer", 3000)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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[OpenSIPS-Users] Updating from OpenSIPS 2.2.2 to 2.2.3.
Dear OpenSIPS-users; We are finishing a project to create a product (Intercom) that uses OpenSIPS 2.2.2. Things are going well with OpenSIPS. I saw that OpenSIPS 2.2.3 fixed lots of bugs. In this case I could be interested in updating my OpenSIPS 2.2.2 to 2.2.3. However, before doing that, I would like to visualize whether such update will need any fix in my code (opensips.cfg file). 1 - Was there some modification in some function signature (name or parameters) from some module? 2 - Was there some modification in the opensips database schema? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Pseudo variable in Route {...} is empty. Why?
There is no error.
That result is observed when the message is a SIP ACK, for example.
Br.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Rodrigo Pimenta Carvalho <pime...@inatel.br>
Enviado: quinta-feira, 8 de dezembro de 2016 15:42
Para: users@lists.opensips.org
Assunto: [OpenSIPS-Users] Pseudo variable in Route {...} is empty. Why?
Hi.
In my OpenSIPS script I have a kind of code like this:
route{
if (nat_uac_test("114"))
{
if (is_method("REGISTER"))
{
} else
if($(ct.fields(uri){uri.host}) == "127.0.0.1" ) {
}
}
}
--
Some times I see the following error in the code:
ERROR:core:comp_scriptvar: cannot get left var value
This error is about the instruction '$(ct.fields(uri){uri.host})'. But, what is
wrong here ?
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
Users mailing list
Users@lists.opensips.org
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[OpenSIPS-Users] Pseudo variable in Route {...} is empty. Why?
Hi.
In my OpenSIPS script I have a kind of code like this:
route{
if (nat_uac_test("114"))
{
if (is_method("REGISTER"))
{
} else
if($(ct.fields(uri){uri.host}) == "127.0.0.1" ) {
}
}
}
--
Some times I see the following error in the code:
ERROR:core:comp_scriptvar: cannot get left var value
This error is about the instruction '$(ct.fields(uri){uri.host})'. But, what is
wrong here ?
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Conflicting information from commands 'opensipsctl ul show' and ' opensipsctl fifo list_tcp_conns '.
Hi. Razvan. Thank you very much! If I remember, I guess that I had removed all records from table location, of user A. But I will pay more attention on it next time. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org> Enviado: quarta-feira, 30 de novembro de 2016 08:47 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] Conflicting information from commands 'opensipsctl ul show' and ' opensipsctl fifo list_tcp_conns '. Hi, Rodrigo! Before removing A from the user location, did you do an opensips ul show to see what registrations OpenSIPS knows? Are there multiple registrations? Are you deleting all of them? Opening a TCP connection to opensips doesn't necessarily mean that the client also sent a REGISTER message, and therefore the client is not yet registered from SIP perspective. What you might see there (with port 48695) might be an old (bogus) registration. After a while, when the client registers, you see the correct info. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 11/29/2016 10:33 PM, Rodrigo Pimenta Carvalho wrote: > Hi. > > > I have peer A and peer B online on my OpenSIPS. > > After removing peer A from table location and reseting peer A, I have: > > > Connection:: ID=29 Type=tcp State=0 Source=127.0.0.1:*52887* > Destination=127.0.0.1:5060 Lifetime=1970-01-01 08:44:41 > > It means that peer A is online on OpenSIPS via TCP socket with port > 52887. This is the result of command '/opensipsctl fifo list_tcp_conns/'. > > > However, the commad '/opensipsctl ul show/' gives me: > > > AOR:: intercomA_5dtUWgwgqzR6 > Contact:: > sip:intercomA_5dtUWgwgqzR6@127.0.0.1:*48694*;transport=TCP;ob Q= > Expires:: 479 > Callid:: 5694778c-6178-4e80-bf2e-7a4dc0deb5d1 > Cseq:: 37504 > User-agent:: n/a > State:: CS_SYNC > Flags:: 0 > Cflags:: > Socket:: tcp:127.0.0.1:5060 > Methods:: 8063 > Attr:: in_same_network > SIP_instance:: > > It means that peer A is online on OpenSIPS via TCP socket with port > 48694. So, I have a kind of conflict here. How can it be possible? > > So, if peer A calls peer B, when B answers I can see the following log: > > > Jan 1 08:36:14 [435] ERROR:core:tcpconn_async_connect: failed to > retrieve SO_ERROR [server=127.0.0.1:*48694*] (111) Connection refused > > > Why such behavior does exist in OpenSIPS? How to avoid it? > > And after a while a new TCP connection appered in port 52887. Like this: > > > Contact:: > sip:intercomA_5dtUWgwgqzR6@127.0.0.1:52887;transport=TCP;ob Q= > Expires:: 236 > Callid:: 96672dc5-a98c-468e-a07a-aca27748791a > Cseq:: 25094 > User-agent:: n/a > State:: CS_SYNC > Flags:: 0 > Cflags:: > Socket:: tcp:127.0.0.1:5060 > Methods:: 8063 > Attr:: in_same_network > SIP_instance:: > > Could it be a problem in OpenSIPS? > > > > Any hint will be very helpful! > > > Best regards. > > > > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users Users Info Page - OpenSIPS<http://lists.opensips.org/cgi-bin/mailman/listinfo/users> lists.opensips.org Discussions about how to use OpenSIPS (non-business). To see the collection of prior postings to the list, visit the Users Archives. > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users Users Info Page - OpenSIPS<http://lists.opensips.org/cgi-bin/mailman/listinfo/users> lists.opensips.org Discussions about how to use OpenSIPS (non-business). To see the collection of prior postings to the list, visit the Users Archives. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Conflicting information from commands 'opensipsctl ul show' and ' opensipsctl fifo list_tcp_conns '.
Hi. I have peer A and peer B online on my OpenSIPS. After removing peer A from table location and reseting peer A, I have: Connection:: ID=29 Type=tcp State=0 Source=127.0.0.1:52887 Destination=127.0.0.1:5060 Lifetime=1970-01-01 08:44:41 It means that peer A is online on OpenSIPS via TCP socket with port 52887. This is the result of command ' opensipsctl fifo list_tcp_conns '. However, the commad 'opensipsctl ul show' gives me: AOR:: intercomA_5dtUWgwgqzR6 Contact:: sip:intercomA_5dtUWgwgqzR6@127.0.0.1:48694;transport=TCP;ob Q= Expires:: 479 Callid:: 5694778c-6178-4e80-bf2e-7a4dc0deb5d1 Cseq:: 37504 User-agent:: n/a State:: CS_SYNC Flags:: 0 Cflags:: Socket:: tcp:127.0.0.1:5060 Methods:: 8063 Attr:: in_same_network SIP_instance:: It means that peer A is online on OpenSIPS via TCP socket with port 48694. So, I have a kind of conflict here. How can it be possible? So, if peer A calls peer B, when B answers I can see the following log: Jan 1 08:36:14 [435] ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server=127.0.0.1:48694] (111) Connection refused Why such behavior does exist in OpenSIPS? How to avoid it? And after a while a new TCP connection appered in port 52887. Like this: Contact:: sip:intercomA_5dtUWgwgqzR6@127.0.0.1:52887;transport=TCP;ob Q= Expires:: 236 Callid:: 96672dc5-a98c-468e-a07a-aca27748791a Cseq:: 25094 User-agent:: n/a State:: CS_SYNC Flags:: 0 Cflags:: Socket:: tcp:127.0.0.1:5060 Methods:: 8063 Attr:: in_same_network SIP_instance:: Could it be a problem in OpenSIPS? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Documentation about media relay with OpenSIPS
Hi. Thank all of you. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Bogdan-Andrei Iancu <bog...@opensips.org> Enviado: segunda-feira, 28 de novembro de 2016 07:47 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] Documentation about media relay with OpenSIPS Hi, I also recommend rtpproxy - it is a good and powerful (in terms of capabilities) media engine, even if not capable to handle WS. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home - OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 23.11.2016 15:53, Johan De Clercq wrote: I think it's better to use rtpengine. There is a small tutorial on the opensips site (webrtc) and there is extensive documentation on github. 2016-11-23 12:34 GMT+01:00 Rodrigo Pimenta Carvalho <pime...@inatel.br<mailto:pime...@inatel.br>>: Dear OpenSIPS users, I would like to learn about how to implement a media relay by means of OpenSIPS. I have found the documentation about the mediaproxy module. Is it the right documentation for media relaying with OpenSIPS, isn't it? Could someone here point some more documentations about it, available on Internet, please? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979 ___ 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
[OpenSIPS-Users] Documentation about media relay with OpenSIPS
Dear OpenSIPS users, I would like to learn about how to implement a media relay by means of OpenSIPS. I have found the documentation about the mediaproxy module. Is it the right documentation for media relaying with OpenSIPS, isn't it? Could someone here point some more documentations about it, available on Internet, please? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout?
Hi Rasvan Crainea. Thank you very much for the reply! You gave me new points to be checked and understood. I will work for a while in such points and then I will give you a feedback. So, wait for my next post, please. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org> Enviado: sexta-feira, 11 de novembro de 2016 11:30 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout? Hi, Rodrigo! Sorry, I've just seen the message, I've missed it earlier. As far as I understand, OpenSIPS is listening on two interfaces: 127.0.01:5060 and 192.168.0.101:5060. Is the UPDATE coming on the same TCP connection as the initial one? Or the client opens a new connection for it, over the PUBLIC interface? Could you send over (privately) a PCAP trace? Also, you'd probably need to make sure that you call fix_nated_contact() and force_rport() on the UPDATE request. Also, are you setting the tcp_accept_aliases[1] or force_tcp_alias()[2] in your script? [1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc95 [2] http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#force_tcp_alias Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 11/10/2016 09:59 PM, Rodrigo Pimenta Carvalho wrote: Hi Razvan. I answered your questions yesterday. I'm not sure if you saw my message. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org> <users-boun...@lists.opensips.org><mailto:users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org><mailto:raz...@opensips.org> Enviado: quarta-feira, 9 de novembro de 2016 08:29 Para: users@lists.opensips.org<mailto:users@lists.opensips.org> Assunto: Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout? Hi, Rodrigo! The only HACK that I can think of is when you get the BYE message, set the dialog timeout to 0, match it against the dialog, and then drop the message. OpenSIPS will behave as if the dialog expired in that moment. However, you seem to have a flow logic - most likely the Contact header in the BYE is not correct. Could you send us a trace to help you figure out what the problem is? Also, did you try to validate the message against the dialog[1] and fix it accordingly[2]? [1] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295894 [2] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295982 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com<http://www.opensips-solutions.com> OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 11/08/2016 10:07 PM, Rodrigo Pimenta Carvalho wrote: Hi. Dialogs in my OpenSIPS is programmed to finish after 60 seconds. (timeout = 1 minute). So, when 2 peers are in a dialog, OpenSIPS sends SIP BYE to both peers, automatically after 60 seconds. Is it possible to make OpenSIPS send this exact kind of SIP BYE to both peers, before the dialog timeout? I mean, in a configured way (opensips.cfg)? When OpenSISP sends SIP BYE automatically, both peers receive the SIP BYE correctly. However, when a peer sends SIP BYE, it reaches the OpenSIPS, but OpenSIPS is unable to forward this SIP BYE. Due to some unknown reason, in this moment there is no open socket to communicate with such peer. That is why I would like to make OpenSIPS send 'its own' SIP BYE, and see if such idea will simulate a normal situation, until I discover why there is a socket problem. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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-
Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout?
Hi. Răzvan. Thanks to your support, I have found the issue, after some weeks of tests! Now I need to discover how to fix it or a workaround. The problem is: Peer A and peer B is connected to the OpenSIPS (OpenSIPS is behind a NAT with public IP = 111.111.240.71). A is in the same hardware as OpenSIPs and B is in another machine. So, OpenSIPs has: Connection A :: ID=1 Type=tcp State=0 Source=127.0.0.1:32908 Destination=127.0.0.1:5060 Lifetime=1970-01-02 05:26:31 Connection B :: ID=2 Type=tcp State=0 Source=111.111.240.71:55229 Destination=192.168.0.101:5060 Lifetime=1970-01-02 05:29:50 When A calls B, the 'contact' in SIP INVITE has Source=127.0.0.1:3290. When B answers with SIP OK, A sends a SIP UPDATE. This UPDATE has a 'contact' with a new value: Source=111.111.240.71:57186. So, OpenSIPS has: Connection A':: ID=1 Type=tcp State=0 Source=127.0.0.1:32908 Destination=127.0.0.1:5060 Lifetime=1970-01-02 05:33:43 Connection B:: ID=2 Type=tcp State=0 Source=111.111.240.71:55229 Destination=192.168.0.101:5060 Lifetime=1970-01-02 05:29:51 Connection A":: ID=3 Type=tcp State=0 Source=111.111.240.71:57186 Destination=192.168.0.101:5060 Lifetime=1970-01-02 05:29:51 (new socket to A) However, the new socket between A and OpenSIPS expires in 30 seconds. Then, when B sends SIP BYE, OpenSIPS tries to forward such SIP message to Source=111.111.240.71:57186, which is expired. Peer A and B use ICE. I would like to fix this issue by means of the OpenSIPS script, to avoid changing the client's code. Could you suggest me some idea? Any hint will be very helpful! Best regards. P.S.: I'm using validate_dialog, but not fix_route_dialog() yet. Could this las function be the solution? RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org> Enviado: quarta-feira, 9 de novembro de 2016 08:29 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout? Hi, Rodrigo! The only HACK that I can think of is when you get the BYE message, set the dialog timeout to 0, match it against the dialog, and then drop the message. OpenSIPS will behave as if the dialog expired in that moment. However, you seem to have a flow logic - most likely the Contact header in the BYE is not correct. Could you send us a trace to help you figure out what the problem is? Also, did you try to validate the message against the dialog[1] and fix it accordingly[2]? [1] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295894 [2] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295982 Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 11/08/2016 10:07 PM, Rodrigo Pimenta Carvalho wrote: Hi. Dialogs in my OpenSIPS is programmed to finish after 60 seconds. (timeout = 1 minute). So, when 2 peers are in a dialog, OpenSIPS sends SIP BYE to both peers, automatically after 60 seconds. Is it possible to make OpenSIPS send this exact kind of SIP BYE to both peers, before the dialog timeout? I mean, in a configured way (opensips.cfg)? When OpenSISP sends SIP BYE automatically, both peers receive the SIP BYE correctly. However, when a peer sends SIP BYE, it reaches the OpenSIPS, but OpenSIPS is unable to forward this SIP BYE. Due to some unknown reason, in this moment there is no open socket to communicate with such peer. That is why I would like to make OpenSIPS send 'its own' SIP BYE, and see if such idea will simulate a normal situation, until I discover why there is a socket problem. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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
[OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout?
Hi. Dialogs in my OpenSIPS is programmed to finish after 60 seconds. (timeout = 1 minute). So, when 2 peers are in a dialog, OpenSIPS sends SIP BYE to both peers, automatically after 60 seconds. Is it possible to make OpenSIPS send this exact kind of SIP BYE to both peers, before the dialog timeout? I mean, in a configured way (opensips.cfg)? When OpenSISP sends SIP BYE automatically, both peers receive the SIP BYE correctly. However, when a peer sends SIP BYE, it reaches the OpenSIPS, but OpenSIPS is unable to forward this SIP BYE. Due to some unknown reason, in this moment there is no open socket to communicate with such peer. That is why I would like to make OpenSIPS send 'its own' SIP BYE, and see if such idea will simulate a normal situation, until I discover why there is a socket problem. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to find out which TCP sockes OpenSIPS is listening on?
Hi. Is it possible to find which is every socket that is currently opened to opensips listens on SIP messages from peers, while using TCP? I have examined opensipsctl command, but it doesn't show the sockets. I need see if a new socket is being created and opened when a peer sends a SIP UPDATE. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Can I read "$(avp(attr)[$T_branch_idx])" in "branch_route[per_branch_ops]" ??
Hi Liviu.
Thank you.
That is the point I want to discuss!
According to the example from the documentation, the lookup() is in
route[relay] {
}
However, in my code the lookup() is in
route{
}
Even with lookup() in rout{...}, does it will populate the attributes of each
branch in my "attr_avp" and I will be able to access them within branch_route?
Best regards!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Liviu Chircu <li...@opensips.org>
Enviado: segunda-feira, 31 de outubro de 2016 12:26
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Can I read "$(avp(attr)[$T_branch_idx])" in
"branch_route[per_branch_ops]" ??
Yes, lookup() will populate the attributes of each branch in your "attr_avp",
and you can access them through $T_branch_idx within branch_route. [1]
[1]: http://www.opensips.org/html/docs/modules/2.3.x/registrar.html#id293909
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
Home - OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 31.10.2016 16:08, Rodrigo Pimenta Carvalho wrote:
Hi.
In my project with OpenSIPS, I have the following kind of code in the
configuration:
modparam("registrar", "attr_avp", "$avp(attr)")
.
.
.
if register request come from machine M1 {
$avp(attr) = "User in M1"; //this mean that the user is behind a NAT, from
the point of view OpenSIPS.
}
if register request come from machine M2 {
$avp(attr) = "User in M2";
}
...
else{
$avp(attr) = "User in Mx"; //this mean that there is no NAT.
}
In my opensips.cfg file I need read "$(avp(attr)[$T_branch_idx])" in the
"branch_route[per_branch_ops]" .
Does $(avp(attr)[$T_branch_idx]) will give me the correct value, even if the
callee is always online in several machines (M1, M2...etc) ?
So, can I discover which machine is participating in the dialog?
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
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[OpenSIPS-Users] Can I read "$(avp(attr)[$T_branch_idx])" in "branch_route[per_branch_ops]" ??
Hi.
In my project with OpenSIPS, I have the following kind of code in the
configuration:
modparam("registrar", "attr_avp", "$avp(attr)")
.
.
.
if register request come from machine M1 {
$avp(attr) = "User in M1"; //this mean that the user is behind a NAT, from
the point of view OpenSIPS.
}
if register request come from machine M2 {
$avp(attr) = "User in M2";
}
...
else{
$avp(attr) = "User in Mx"; //this mean that there is no NAT.
}
In my opensips.cfg file I need read "$(avp(attr)[$T_branch_idx])" in the
"branch_route[per_branch_ops]" .
Does $(avp(attr)[$T_branch_idx]) will give me the correct value, even if the
callee is always online in several machines (M1, M2...etc) ?
So, can I discover which machine is participating in the dialog?
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is it a kind of TCP keep alive produced by OpenSIPS?
Hi Răzvan. Ok. No problem. Today morning I was investigating the softphone guys (running in the same hardware as OpenSIPS) and I have seen lots of its logs. But the logs didn't presented any problem. Everything seems to be ok. Those logs showed SIP and ICE messages. Maybe I will have to use TCP dump in the hardware where the problem is happening. And investigate at TCP level. I will prepare the log that you asked me. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org> Enviado: sexta-feira, 28 de outubro de 2016 06:09 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] Is it a kind of TCP keep alive produced by OpenSIPS? Unfortunately I have no other ideas about what you could do. You'd better ask for support from the softphone guys, to see why they are closing the connections. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 10/27/2016 08:48 PM, Rodrigo Pimenta Carvalho wrote: Hi. Răzvan. Thank you very much! So, I will keep using the flag "Pp" to create dialogs. As I understood, it will not cause any problem. Yes, it is the client that closes the connection, always. After some more investigation, I have discovered the following specific situation: When softphone A (which is always using ICE and STUN) calls B, if B is not using ICE and STUN, the TCP connection between A and OpenSIPS remains stable. However, in this scenario, if B is using ICE and STUN, A closes the TCP connection to OpenSIPS after 33 seconds of dialog. Here, SIP is over TCP and ICE uses UDP. A and OpenSIPS run in the same hardware. So, there is no NAT between A and OpenSIPS. B run in another hardware, but in the same local network (same network domain). So, there is no NAT between B and OpenSIPS. A is a proprietary softphone and B is Microsip. I have looked at the proprietary softphone log and there is no issues with SIP. Do you have some more hint about what to investigate next? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org> <users-boun...@lists.opensips.org><mailto:users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org><mailto:raz...@opensips.org> Enviado: quinta-feira, 27 de outubro de 2016 05:47 Para: users@lists.opensips.org<mailto:users@lists.opensips.org> Assunto: Re: [OpenSIPS-Users] Is it a kind of TCP keep alive produced by OpenSIPS? Hi, Rodrigo! See my answers inline. BR Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com<http://www.opensips-solutions.com> OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 10/26/2016 08:15 PM, Rodrigo Pimenta Carvalho wrote: Hi Răzvan. Thank you very much. I'm facing a problem here related to TCP connection teared down during dialogs. While a peer is not in dialogs, its TCP connection to OpenSIPS keeps online all the time. However, when such peer enters in a conversation (be part of a dialog), after few minutes there is a EOF received in a socket. After this, OpenSIPS can no more send SIP BYEs to the respective peer. In the log I can see: Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcp_read: EOF on 0x74e3d048, FD 24 Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcp_read_req: EOF received Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:io_watch_del: [TCP_worker] io_watch_del op on index 0 24 (0x1875e8, 24, 0, 0x10,0x3) fd_no=3 called Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcpconn_release: releasing con 0x74e3d048, state -1, fd=-1, id=3 Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcpconn_release: extra_data (nil) Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21029] DBG:core:handle_tcp_worker: reader response= 74e3d048, -1 from 2 Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21029] DBG:core:tcpconn_destroy: destroying
Re: [OpenSIPS-Users] OpenSIPS reopen TCP connectios and sends INVITE, but not BYE. How to change it?
Hi Razvan, Thank you very much again! See my comments and question in line, please. Best regards! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Razvan Crainea <raz...@opensips.org> Enviado: quinta-feira, 27 de outubro de 2016 05:58 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] OpenSIPS reopen TCP connectios and sends INVITE, but not BYE. How to change it? Hi, Rodrigo! Having OpenSIPS opening TCP connections towards client is a bit dangerous, especially if the clients are behind NAT. That's because most likely you will not be able to reach them, and opensips will get stuck trying to connect (until it triggers a timeout). That's why the best way to go is to try to keep the connection (ideally opened by the client at REGISTER) as much as possible. This is usually done by pinging (as discussed in a previous email). So my suggestion is to try to avoid opening new TCP connections with clients, unless you really know they will always be reachable. The client will be always reachable. Because in my specific case, the client(which break down the TCP connection) is in the same hardware as OpenSIPS. So, there will not be NATs here. As I saw in the log, OpenSIPS reopen the connection, like this: DBG:core:proto_tcp_send: no open tcp connection found, opening new one, async = 1 And this is opened in the moment after OpenSIPS trying to pass the SIP BYE to the local client. As long as OpenSIPS is already reopening the TCP connection, when it needs to send the SIP BYE, why the SIP BYE is not sent finally? I believe that I can use such new connection to send the SIP BYE. In this case, I intend to force OpenSIPS to send the SIP BYE after reopening such TCP connection. Is it possible in terms of script? I have just checked my script and I'm not using the flag tcp_no_new_conn_bflag. The behavior you are describing (INVITE vs BYE handling), might be related to the fact that you are setting the tcp_no_new_conn_bflag[1] flag for BYE messages, but not for INVITEs. Is this correct? If not, do you see any errors in the script? [1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc101 Razvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> Home - OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 10/26/2016 10:59 PM, Rodrigo Pimenta Carvalho wrote: Hi. After some log debug I have observed the following behavior in the OpenSISP (2.2.1): When OpenSIPS has to send a SIP INVITE to a peer through a TCP connection that was closed before by some way, OpenSIPS open a new one and then sends the SIP message to the peer successfully. However, when OpenSIPS has to send a SIP BYE to a peer through a TCP connection that was closed before, OpenSIPS open a new one, but doesn't send the SIP BYE. In this case SIP BYE is discarded. How to change the behavior of OpenSIPS to make it to send the SIP BYE is such case? I'm looking for ways of fix or workaround of a TCP tear down connection that happens during dialogs. Any hint will be very helpful! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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
Re: [OpenSIPS-Users] Is it a kind of TCP keep alive produced by OpenSIPS?
Hi. Răzvan. Thank you very much! So, I will keep using the flag "Pp" to create dialogs. As I understood, it will not cause any problem. Yes, it is the client that closes the connection, always. After some more investigation, I have discovered the following specific situation: When softphone A (which is always using ICE and STUN) calls B, if B is not using ICE and STUN, the TCP connection between A and OpenSIPS remains stable. However, in this scenario, if B is using ICE and STUN, A closes the TCP connection to OpenSIPS after 33 seconds of dialog. Here, SIP is over TCP and ICE uses UDP. A and OpenSIPS run in the same hardware. So, there is no NAT between A and OpenSIPS. B run in another hardware, but in the same local network (same network domain). So, there is no NAT between B and OpenSIPS. A is a proprietary softphone and B is Microsip. I have looked at the proprietary softphone log and there is no issues with SIP. Do you have some more hint about what to investigate next? Any hint will be very helpful!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org> Enviado: quinta-feira, 27 de outubro de 2016 05:47 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] Is it a kind of TCP keep alive produced by OpenSIPS? Hi, Rodrigo! See my answers inline. BR Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 10/26/2016 08:15 PM, Rodrigo Pimenta Carvalho wrote: Hi Răzvan. Thank you very much. I'm facing a problem here related to TCP connection teared down during dialogs. While a peer is not in dialogs, its TCP connection to OpenSIPS keeps online all the time. However, when such peer enters in a conversation (be part of a dialog), after few minutes there is a EOF received in a socket. After this, OpenSIPS can no more send SIP BYEs to the respective peer. In the log I can see: Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcp_read: EOF on 0x74e3d048, FD 24 Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcp_read_req: EOF received Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:io_watch_del: [TCP_worker] io_watch_del op on index 0 24 (0x1875e8, 24, 0, 0x10,0x3) fd_no=3 called Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcpconn_release: releasing con 0x74e3d048, state -1, fd=-1, id=3 Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcpconn_release: extra_data (nil) Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21029] DBG:core:handle_tcp_worker: reader response= 74e3d048, -1 from 2 Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21029] DBG:core:tcpconn_destroy: destroying connection 0x74e3d048, flags 0006 ... When OpenSIPS try to send a SIP BYE via socket 0x74e3d048 , I can see the log: Jan 02 01:40:49 colibri-imx6-jfl opensips[21018]: Jan 2 01:40:49 [21026] DBG:core:proto_tcp_send: no open tcp connection found, opening new one, async = 1 I have already used the flag "Pp" in the creation of dialogs, but it didn't take effect. That is, even with "Pp" I'm still getting "EOF" in the TCP socket. 1 - Should the flag "Pp" avoid those EOFs during dialogs? Ideally, it should. However if the client does not "like" the pinging and closes the connection, there's not that much we can do. That flag causes the OpenSIPS to send SIP OPTIONS. The peers are replying with SIP 500. That's not really an issue. The SIP OPTIONs pinging has two purposes: 1. verify if the dialog is still active, and 2. keep the NAT pinhole open. If the SIP client doesn't know how to reply to in-dialog pinging, then 1. isn't really useful. So the reply code doesn't really matter, unless it is a 408, which means that the peer did not respond at all, and the dialog will be turn down. 2- Is a SIP 500 reply enough to OpenSIPS keep the dialog connected? Any communication between OpenSIPS and the client keeps the NAT pinhole open (see 2. above). From SIP perspective, that 500 could have a lot of meanings: the client does not know how to reply, or there was an internal error that could not process the message. However, this whole communication will keep the connection alive. 3 - Does it make sense getting absence of keep alive messages during dialogs? So as I said abov
[OpenSIPS-Users] OpenSIPS reopen TCP connectios and sends INVITE, but not BYE. How to change it?
Hi. After some log debug I have observed the following behavior in the OpenSISP (2.2.1): When OpenSIPS has to send a SIP INVITE to a peer through a TCP connection that was closed before by some way, OpenSIPS open a new one and then sends the SIP message to the peer successfully. However, when OpenSIPS has to send a SIP BYE to a peer through a TCP connection that was closed before, OpenSIPS open a new one, but doesn't send the SIP BYE. In this case SIP BYE is discarded. How to change the behavior of OpenSIPS to make it to send the SIP BYE is such case? I'm looking for ways of fix or workaround of a TCP tear down connection that happens during dialogs. Any hint will be very helpful! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is it a kind of TCP keep alive produced by OpenSIPS?
Hi Răzvan. Thank you very much. I'm facing a problem here related to TCP connection teared down during dialogs. While a peer is not in dialogs, its TCP connection to OpenSIPS keeps online all the time. However, when such peer enters in a conversation (be part of a dialog), after few minutes there is a EOF received in a socket. After this, OpenSIPS can no more send SIP BYEs to the respective peer. In the log I can see: Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcp_read: EOF on 0x74e3d048, FD 24 Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcp_read_req: EOF received Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:io_watch_del: [TCP_worker] io_watch_del op on index 0 24 (0x1875e8, 24, 0, 0x10,0x3) fd_no=3 called Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcpconn_release: releasing con 0x74e3d048, state -1, fd=-1, id=3 Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21027] DBG:core:tcpconn_release: extra_data (nil) Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21029] DBG:core:handle_tcp_worker: reader response= 74e3d048, -1 from 2 Jan 02 01:38:45 colibri-imx6-jfl opensips[21018]: Jan 2 01:38:45 [21029] DBG:core:tcpconn_destroy: destroying connection 0x74e3d048, flags 0006 ... When OpenSIPS try to send a SIP BYE via socket 0x74e3d048 , I can see the log: Jan 02 01:40:49 colibri-imx6-jfl opensips[21018]: Jan 2 01:40:49 [21026] DBG:core:proto_tcp_send: no open tcp connection found, opening new one, async = 1 I have already used the flag "Pp" in the creation of dialogs, but it didn't take effect. That is, even with "Pp" I'm still getting "EOF" in the TCP socket. 1 - Should the flag "Pp" avoid those EOFs during dialogs? That flag causes the OpenSIPS to send SIP OPTIONS. The peers are replying with SIP 500. 2- Is a SIP 500 reply enough to OpenSIPS keep the dialog connected? 3 - Does it make sense getting absence of keep alive messages during dialogs? Any hint will be very helpful! P.S.: I will check the TCP trace too, looking for keep alives. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Răzvan Crainea <raz...@opensips.org> Enviado: quarta-feira, 26 de outubro de 2016 13:08 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] Is it a kind of TCP keep alive produced by OpenSIPS? Hi, Rodrigo! The logs you are tracing are printed when OpenSIPS receives something from the client, and then immediately responds back. Due to the fact that we don't see any other debug messages, like SIP parsing & stuff, makes me think that it is a CRLF pinging - the client periodically sends a CRLFCRLF TCP message to OpenSIPS, and OpenSIPS responds with a single CRLF. Note that this is different from a TCP keep-alive, where each peer send a 0-length TCP message, without any body. That message doesn't even get to the application layer. However, tracing the communication between OpenSIPS and the client should confirm the above :). Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 10/26/2016 05:10 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS users, In the OpenSIPS log I see: Jan 01 19:30:38 colibri-imx6-jfl opensips[3444]: Jan 1 19:30:38 [3451] DBG:core:tcp_read_req: Using the global ( per process ) buff Jan 01 19:30:38 colibri-imx6-jfl opensips[3444]: Jan 1 19:30:38 [3451] DBG:core:tcp_handle_req: content-length= 0 Jan 01 19:30:38 colibri-imx6-jfl opensips[3444]: Jan 1 19:30:38 [3451] DBG:core:async_tsend_stream: Async successful write from first try on 0x74e13548 Jan 01 19:30:38 colibri-imx6-jfl opensips[3444]: Jan 1 19:30:38 [3451] DBG:core:tcp_read_req: tcp_read_req end The frequency is 1 time at each 1,5 minute. There is only one client online. I suspect that OpenSIPS uses the socket 0x74e13548 to send messages to such client. The client became online using TCP. Just to confirm, is this log a result of a TCP keep alive function enabled? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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
[OpenSIPS-Users] Is it a kind of TCP keep alive produced by OpenSIPS?
Dear OpenSIPS users, In the OpenSIPS log I see: Jan 01 19:30:38 colibri-imx6-jfl opensips[3444]: Jan 1 19:30:38 [3451] DBG:core:tcp_read_req: Using the global ( per process ) buff Jan 01 19:30:38 colibri-imx6-jfl opensips[3444]: Jan 1 19:30:38 [3451] DBG:core:tcp_handle_req: content-length= 0 Jan 01 19:30:38 colibri-imx6-jfl opensips[3444]: Jan 1 19:30:38 [3451] DBG:core:async_tsend_stream: Async successful write from first try on 0x74e13548 Jan 01 19:30:38 colibri-imx6-jfl opensips[3444]: Jan 1 19:30:38 [3451] DBG:core:tcp_read_req: tcp_read_req end The frequency is 1 time at each 1,5 minute. There is only one client online. I suspect that OpenSIPS uses the socket 0x74e13548 to send messages to such client. The client became online using TCP. Just to confirm, is this log a result of a TCP keep alive function enabled? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Why am I gettin error to terminate long calls and not short ones? Why it is "Ignoring callid"?
Hi Răzvan.
I'm still investigating the problem.
Now I'm using the'Pp' flag to create_dialog(), as you had suggested. In this
case I can see that OpenSIPS sends SIP OPTIONS to the 2 peers and they respond
with SIP 500 "Unhandled by dialog usages". It is ok to a ping purpose, isn't
it? That is, even if the response is SIP 500, OpenSIPS will know that the peer
is online. Ok?
One thing that let me curious is the log below (that rises even using 'Pp' flag
to create dialogs):
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
dlg=[sip:user_A@127.0.0.1:36427;transport=TCP;ob] ,
req=[sip:user_A@192.168.0.101:57985;transport=TCP;ob]
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: In-Dialog BYE from
192.168.0.102 (callid=ec4548a8-4207-4fc2-8ed8-81897ff62175) is not valid
according to dialog
The problem here exists when user B sends SIP BYE to user A. B sends it to the
contact user_A@192.168.0.101:57985. However, this contact is not known by
OpenSIPS and then the proxy complains with such log. OpenSIPS does know just
the contact in the location table, doesn't it?
In table location the contact of user A is "user_A@127.0.0.1:36427". But,
during the dialog, A sends a SIP UPDATE to B. And such UPDATE has the contact
"user_A@192.168.0.101:57985" when it arrives in B. Softphone for user B and
OpenSIPS is running in the same hardware, as I told before, with IP =
192.168.0.101. So, I suspect that UAC B decides to send SIP BYE to
"user_A@192.168.0.101:57985" due to that contact found in SIP UPDATE.
It seems that UA A sends SIP UPDATE just when it and OpenSIPS is running in the
same hardware. But I'm not sure...
Should I fix the contact in the SIP UPDATE before relaying it? Is it possible
by means of the opensips.cfg file script to fix the contact in the SIP UPDATE?
Or should I fix the SIP BYE request when it arrives in OpenSIPS, before the
proxy to investigate if the contact is in table location?
Any hint will be very helpful!!
Thanks a lot!
Best regards!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Răzvan Crainea <raz...@opensips.org>
Enviado: terça-feira, 18 de outubro de 2016 05:18
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Why am I gettin error to terminate long calls and
not short ones? Why it is "Ignoring callid"?
Hi, Rodrigo!
Most likely A closes the connection to OpenSIPS. You can check that by tracing
the communication between A and OpenSIPS.
In order to solve that, make sure that the TCP keepalive[1] is enabled. Also,
you can use the dialog pinging[2] feature ('Pp' flag to create_dialog()) to
keep the dialog connections open.
[1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc103
[2] http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id295792
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com<http://www.opensips-solutions.com>
Home — OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 10/17/2016 11:20 PM, Rodrigo Pimenta Carvalho wrote:
Dear OpenSIPS users,
In my hardware, with IP = 192.168.0.101, I have OpenSIPS and softphone A. Thre
is softphone B also, in another hardware.
A calls B.
B accept the call.
After t minutes...B hungs up the call.
In this moment, A enters in a wrong state, because OpenSIPS reports a problem
and probably due to it the proxy doesn't communicate with softphone A in such
moment. So, my softphone A considers that the call is not ended.
See what OpenSIPS reports in this moment:
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
INFO:core:probe_max_sock_buff: using snd buffer of 320 kb
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 20
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:core:tcpconn_async_connect: poll error: flags 1c
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR
[server=192.168.0.101:57985] (111) Connection refused
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:core:proto_tcp_send: async TCP connect failed
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:tm:msg_send: send() for proto 2 failed
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:tm:t_forward_nonack: sending request failed
If t is just few minutes, let's say 2 minutes,
Re: [OpenSIPS-Users] Why am I gettin error to terminate long calls and not short ones? Why it is "Ignoring callid"?
Ok.
Thank you very much for the hint.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Răzvan Crainea <raz...@opensips.org>
Enviado: terça-feira, 18 de outubro de 2016 05:18
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Why am I gettin error to terminate long calls and
not short ones? Why it is "Ignoring callid"?
Hi, Rodrigo!
Most likely A closes the connection to OpenSIPS. You can check that by tracing
the communication between A and OpenSIPS.
In order to solve that, make sure that the TCP keepalive[1] is enabled. Also,
you can use the dialog pinging[2] feature ('Pp' flag to create_dialog()) to
keep the dialog connections open.
[1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc103
[2] http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id295792
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com<http://www.opensips-solutions.com>
On 10/17/2016 11:20 PM, Rodrigo Pimenta Carvalho wrote:
Dear OpenSIPS users,
In my hardware, with IP = 192.168.0.101, I have OpenSIPS and softphone A. Thre
is softphone B also, in another hardware.
A calls B.
B accept the call.
After t minutes...B hungs up the call.
In this moment, A enters in a wrong state, because OpenSIPS reports a problem
and probably due to it the proxy doesn't communicate with softphone A in such
moment. So, my softphone A considers that the call is not ended.
See what OpenSIPS reports in this moment:
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
INFO:core:probe_max_sock_buff: using snd buffer of 320 kb
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 20
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:core:tcpconn_async_connect: poll error: flags 1c
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR
[server=192.168.0.101:57985] (111) Connection refused
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:core:proto_tcp_send: async TCP connect failed
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:tm:msg_send: send() for proto 2 failed
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:tm:t_forward_nonack: sending request failed
If t is just few minutes, let's say 2 minutes, there is no any issue.
However, if t is bigger, let's say 4 minutes, his issue is present.
What is happening here? Can someone give some help, please!
Any hint will be very helpful!
Some more details:
User B is g1r2u3p4o5@192.168.0.102<mailto:g1r2u3p4o5@192.168.0.102>.
User A is
intercomA_5dtUWgwgqzR6@192.168.0.101<mailto:intercomA_5dtUWgwgqzR6@192.168.0.101>.
Callid was "ec4548a8-4207-4fc2-8ed8-81897ff62175".
Before getting such error log, I saw another messages in the log like this:
Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: new branch at
sip:g1r2u3p4o5@192.168.0.102:61230;transport=TCP;ob<mailto:sip:g1r2u3p4o5@192.168.0.102:61230;transport=TCP;ob>
Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: od: invalid option -- 'A'
Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: BusyBox v1.22.1 (2016-03-29
09:43:20 BRT) multi-call binary.
Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: Usage: od
[-aBbcDdeFfHhIiLlOovXx] [FILE]
Jan 05 04:13:29 colibri-imx6-jfl opensips[431]: Ignoring callid
"ec4548a8-4207-4fc2-8ed8-81897ff62175"
Jan 05 04:13:34 colibri-imx6-jfl opensips[431]: Jan 5 04:13:34 [442]
INFO:core:probe_max_sock_buff: using snd buffer of 320 kb
an 05 04:13:34 colibri-imx6-jfl opensips[431]: Jan 5 04:13:34 [442]
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 25
an 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan 5 04:14:29 [438]
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
dlg=[sip:intercomA_5dtUWgwgqzR6@127.0.0.1:36427;transport=TCP;ob<mailto:sip:intercomA_5dtUWgwgqzR6@127.0.0.1:36427;transport=TCP;ob>]
,
req=[sip:intercomA_5dtUWgwgqzR6@192.168.0.101:57985;transport=TCP;ob<mailto:sip:intercomA_5dtUWgwgqzR6@192.168.0.101:57985;transport=TCP;ob>]
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: In-Dialog BYE from
192.168.0.102 (callid=ec4548a8-4207-4fc2-8ed8-81897ff62175) is not valid
according to dialog
------------
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] OpenSIPS in the market.
Hi people. Thank all of you very much! I need read more and more about the use of SIP in the industry. This will serve as complementary information that I will present in my next class. So, i will search about that "European standard for live radio broadcast" and also about that "standard for air traffic communication, connecting radio masts to a flight control tower". These sound very interesting. Here, where I work we are developing a intercom (door bell) for a company specialized in residential security. The intercom will have the OpenSIPS inside. As a SIP proxy, such intercom will connect the house with smartphones, for example. OpenSIPS is really helping me a lot! Because several project requirements I have programmed inside the opensips.cfg configuration file. Thanks for the support! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Olle E. Johansson <o...@edvina.net> Enviado: terça-feira, 18 de outubro de 2016 06:06 Para: OpenSIPS users mailling list Cc: Olle E Johansson; Rodrigo Pimenta Carvalho Assunto: Re: [OpenSIPS-Users] OpenSIPS in the market. On 18 Oct 2016, at 08:46, Bogdan-Andrei Iancu <bog...@opensips.org<mailto:bog...@opensips.org>> wrote: Hello Rodrigo, The questions your are asking are hard to answer and the Open Source world is most of the times opaque when comes to who is using and how much is used. As anyone can simply download and start using (information on how to do it) is quite enough, you may have many cases were companies do use an Open Source software without "leaking" or "leaving traces" about that. Yes, I agree. These kind of questions are really hard to answer. During the last ten years, acceptance of Open Source in production use for commercial companies have changed worldwide. Open Source proxys like OpenSIPS and the relatives (SER, Kamailio) and other code bases like resiprocate are taking the market together with media servers like Janus, Asterisk and FreeSwitch. This is also for OpenSIPS project - it is hard to say who is using and how the popularity is fluctuating and mainly because there is no much of a feedback (as it is free to use). Still, take a look at : http://www.opensips.org/About/WhoIsUsing This is an attempt to list the companies willing to do it, it is not a full listing. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com<http://www.opensips-solutions.com/> On 13.10.2016 15:44, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS users, As a teacher, I'm going to explain somethings about SIP and SIP proxies to my students. So, I intend to talk about OpenSIPS too. 1- Is there some ranking in any web page showing what is the most used SIP proxies in the telecommunication industries? The interesting part would be Open Source platforms vs commercial and non-Open platforms. But no, I am not either aware of such data. 2 - Does the adoption of OpenSIPS, by telecom players, have been increasing ? Yes 3 - What are examples of most remarkable use cases with OpenSIPS in telecommunications? If you widen the scope to “Open Source proxys” I would say that SIP as a protocol is making headway in interesting areas. SIP is part of a European standard for live radio broadcast. It’s also part of a standard for air traffic communication, connecting radio masts to a flight control tower. I’ve built solutions for both of this standards with Open Source SIP software. So it’s no longer just about “telecommunications” - SIP and the open source software is starting to deliver on the original SIP vision - a realtime communication platform for Internet users. To stay in telecommunications there are many presentations from our various conferences that will give you ideas on where and how it is used. The new black is of course Open Source mobile network platforms - after ISDN and SS7 this is the new interesting area to cover with Open solutions. Regards, /O Any information will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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
[OpenSIPS-Users] OpenSIPS in the market.
Dear OpenSIPS users, As a teacher, I'm going to explain somethings about SIP and SIP proxies to my students. So, I intend to talk about OpenSIPS too. 1- Is there some ranking in any web page showing what is the most used SIP proxies in the telecommunication industries? 2 - Does the adoption of OpenSIPS, by telecom players, have been increasing ? 3 - What are examples of most remarkable use cases with OpenSIPS in telecommunications? Any information will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] What does happen when location table has records for peer that is offline?
Hi. In my table location (in OpenSIPS) I found some records for a peer that was not online. That is, some peer offline let some registers in the location table. For example: the peer A registered itself using several different devices, then turned off all of such devices. Such records would expires sometime in the future. Then, peer B called A, and in this moment I saw a log with several erros, from OpenSIPs. Does such logs relates to these invalid records in table location? After removing these records (opensipsctl ul rm ) those messages stopped appearing in the log. See the log below. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 Jan 01 01:19:16 colibri-imx6 opensips[609]: new branch at sip:g1r2u3p4o5@127.0.0.1:47340;transport=TCP;ob Jan 01 01:19:16 colibri-imx6 opensips[609]: new branch at sip:g1r2u3p4o5@127.0.0.1:54112;transport=TCP;ob Jan 01 01:19:16 colibri-imx6 opensips[609]: new branch at sip:g1r2u3p4o5@127.0.0.1:38220;transport=TCP;ob Jan 01 01:19:16 colibri-imx6 opensips[609]: new branch at sip:g1r2u3p4o5@127.0.0.1:54112;transport=TCP;ob Jan 01 01:19:16 colibri-imx6 opensips[609]: new branch at sip:g1r2u3p4o5@127.0.0.1:38220;transport=TCP;ob Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] INFO:core:probe_max_sock_buff: using snd buffer of 320 kb Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 21 Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:core:tcpconn_async_connect: poll error: flags 1c Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server=127.0.0.1:47340] (111) Connection refused Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:core:proto_tcp_send: async TCP connect failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:tm:msg_send: send() for proto 2 failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:tm:t_forward_nonack: sending request failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] INFO:core:probe_max_sock_buff: using snd buffer of 320 kb Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 21 Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:core:tcpconn_async_connect: poll error: flags 1c Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server=127.0.0.1:54112] (111) Connection refused Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:core:proto_tcp_send: async TCP connect failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:tm:msg_send: send() for proto 2 failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:tm:t_forward_nonack: sending request failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] INFO:core:probe_max_sock_buff: using snd buffer of 320 kb Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 21 Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:core:tcpconn_async_connect: poll error: flags 1c Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server=127.0.0.1:54112] (111) Connection refused Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:core:proto_tcp_send: async TCP connect failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:tm:msg_send: send() for proto 2 failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [616] ERROR:tm:t_forward_nonack: sending request failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [617] INFO:core:probe_max_sock_buff: using snd buffer of 320 kb Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [617] INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 23 Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [617] ERROR:core:tcpconn_async_connect: poll error: flags 1c Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [617] ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server=127.0.0.1:47340] (111) Connection refused Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [617] ERROR:core:proto_tcp_send: async TCP connect failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [617] ERROR:tm:msg_send: send() for proto 2 failed Jan 01 01:19:17 colibri-imx6 opensips[609]: Jan 1 01:19:16 [617] ERROR:tm:t_forward_nonack: sending request failed Jan 01 01:19:17 colibri-imx6 opensips[609
Re: [OpenSIPS-Users] How to take control of some timeouts in OpenSIPS?
Thanks for the hint!! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Johan De Clercq <jo...@democon.be> Enviado: sábado, 10 de setembro de 2016 02:14 Para: users@lists.opensips.org; OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to take control of some timeouts in OpenSIPS? You can use tm module to increase t1. Br, Get Outlook for iOS<https://aka.ms/o0ukef> On Fri, Sep 9, 2016 at 11:02 PM +0200, "Rodrigo Pimenta Carvalho" <pime...@inatel.br<mailto:pime...@inatel.br>> wrote: Dear SIP users, In my product with OpenSIPS we are using SIP over TCP, not UDP, because we are in the test days. And briefly we will start using SIP + TLS. As our network 3G has a considerable network latency, sometimes the OpenSIPS sends the SIP INVITE to a remote UAS and doesn't wait enough time for receiving the SIP RINGING or SIP TRYING. That is, a timeout occurs in the OpenSIPS when there is a considerable network latency, because the SIP response is too delayed. This fact is interrupting the SIP signalization. So, how to calibrate this kind of timeout value in OpenSIPS, so that it will be possible waiting for the SIP response until it be received? Is there a solution for TLS too? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to take control of some timeouts in OpenSIPS?
Dear SIP users, In my product with OpenSIPS we are using SIP over TCP, not UDP, because we are in the test days. And briefly we will start using SIP + TLS. As our network 3G has a considerable network latency, sometimes the OpenSIPS sends the SIP INVITE to a remote UAS and doesn't wait enough time for receiving the SIP RINGING or SIP TRYING. That is, a timeout occurs in the OpenSIPS when there is a considerable network latency, because the SIP response is too delayed. This fact is interrupting the SIP signalization. So, how to calibrate this kind of timeout value in OpenSIPS, so that it will be possible waiting for the SIP response until it be received? Is there a solution for TLS too? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What does mean "a=inactive"?
Hi. Thank you very much! Your explanation was sufficient and now I understood what is happening. There is no changes caused by OpenSIPS. Is the UAC that decides to put that "inactive" there. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Benjamin Cropley <benjamin.crop...@gmail.com> Enviado: quarta-feira, 7 de setembro de 2016 09:01 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] What does mean "a=inactive"? I would start by looking at a trace. Is it A changing the SDP attribute? or is OpenSIPS doing it? Inactive obviously means "Keep the session live, but dont send me any audio, and I won't send you any audio". I've seen that happen once, when both end points couldnt establish a codec.. due to processing error or something like that, but instead of sending an appropriate error, it just connects the call and send that. Hope that helps, Ben Cropley On Thu, Sep 1, 2016 at 1:12 PM, Rodrigo Pimenta Carvalho <pime...@inatel.br<mailto:pime...@inatel.br>> wrote: Dear OpenSIPS users; I'm not sure if the following question is about OpenSIPS, or SIP, or SDP, but... I have 2 softphones (Microsip) with SIP UAC: in phone A and in phone B. There is a SIP Proxy (OpenSIPS) between they too. When A calls B, I can see the SIP messages (via wireshark) and in some moment A sends a SIP UPDATE do B. The SIP UPDATE has SDP with lines like this: v=0 o=- 3681643549 3681643550 IN IP4 XXX.YYY.240.204 s=pjmedia b=AS:84 t=0 0 a=X-nat:1 m=audio 64568 RTP/AVP 110 8 101 c=IN IP4 XXX.YYY.240.204 b=TIAS:64000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ice-ufrag:139d7049 a=ice-pwd:692c4a80 a=rtcp:64571 IN IP4 XXX.YYY.240.204 a=candidate:Sc0a81485 1 UDP 1862270975 XXX.YYY.240.204 64568 typ srflx raddr 192.168.20.133 rport 64568 a=candidate:Sc0a81485 2 UDP 1862270974 XXX.YYY.240.204 64571 typ srflx raddr 192.168.20.133 rport 64571 a=remote-candidates:1 XXX.YYY.240.71 64993 2 XXX.YYY.240.71 64996 a=sendrecv However, if I replace the SIP Proxy with another one containing the same software (Same OpenSIPS, database, network, etc. Just hardware is different) and run the same call (A calls B), that "a=sendrecv" in SIP UPDATE changes to "a=inactive". If the peers are still the same, how could a media attribute changes? I have no idea what could cause this difference related to media attribute! Could OpenSIPS take care of this case? Could someone here give me some examples of what could cause an "a=inactive", so that I will have a point to start my analyze of the problem? I will also take a look in the SIP RFC to get some hint. Any hint will be very very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979 ___ Users mailing list Users@lists.opensips.org<mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- All the best, Ben Cropley 07539 366 905 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] What does mean "a=inactive"?
Dear OpenSIPS users; I'm not sure if the following question is about OpenSIPS, or SIP, or SDP, but... I have 2 softphones (Microsip) with SIP UAC: in phone A and in phone B. There is a SIP Proxy (OpenSIPS) between they too. When A calls B, I can see the SIP messages (via wireshark) and in some moment A sends a SIP UPDATE do B. The SIP UPDATE has SDP with lines like this: v=0 o=- 3681643549 3681643550 IN IP4 XXX.YYY.240.204 s=pjmedia b=AS:84 t=0 0 a=X-nat:1 m=audio 64568 RTP/AVP 110 8 101 c=IN IP4 XXX.YYY.240.204 b=TIAS:64000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ice-ufrag:139d7049 a=ice-pwd:692c4a80 a=rtcp:64571 IN IP4 XXX.YYY.240.204 a=candidate:Sc0a81485 1 UDP 1862270975 XXX.YYY.240.204 64568 typ srflx raddr 192.168.20.133 rport 64568 a=candidate:Sc0a81485 2 UDP 1862270974 XXX.YYY.240.204 64571 typ srflx raddr 192.168.20.133 rport 64571 a=remote-candidates:1 XXX.YYY.240.71 64993 2 XXX.YYY.240.71 64996 a=sendrecv However, if I replace the SIP Proxy with another one containing the same software (Same OpenSIPS, database, network, etc. Just hardware is different) and run the same call (A calls B), that "a=sendrecv" in SIP UPDATE changes to "a=inactive". If the peers are still the same, how could a media attribute changes? I have no idea what could cause this difference related to media attribute! Could OpenSIPS take care of this case? Could someone here give me some examples of what could cause an "a=inactive", so that I will have a point to start my analyze of the problem? I will also take a look in the SIP RFC to get some hint. Any hint will be very very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to Internet, from the script.
Thank all of you.
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Adrian Fretwell <adrian.fretw...@topgreen.co.uk>
Enviado: segunda-feira, 8 de agosto de 2016 04:27
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
Internet, from the script.
Johan,
If your internet connection is going up and down regularly, you may be better
off executing your test from a timer route:
timer_route[internet_check, 300] {
# -
# Timer Route every 5 minutes
# -
exec("/usr/local/bin/some_check _script");
}
There are many different ways to check if you have an internet connection, the
way you do it will depend on your environment and application, but here is a
very simple shell script as an example:
#!/bin/bash
ping -c 2 8.8.8.8 > /dev/null
if [ $? -eq 0 ]; then
echo "Internet Alive $(date)";
else
echo "Internet Dead $(date)";
fi
Kind regards,
Adrian Fretwell
On 08/08/16 07:58, Johan De Clercq wrote:
create a start up route startup_route, the use module exec to f.e. get your pub
ip with curl.
2016-08-04 15:21 GMT+02:00 Rodrigo Pimenta Carvalho
<pime...@inatel.br<mailto:pime...@inatel.br>>:
Hi.
How to discover if OpenSIPS is connected do Internet, from its configuration
script?
Sometimes the Internet Link is down and then just local calls will work. If I
can discover if OpenSIPS is "online" on Internet, I will use this information
to implement some specific logic in my script.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979
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Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to Internet, from the script.
Hi.
Forget the questions from my previous message, please.
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Rodrigo Pimenta Carvalho <pime...@inatel.br>
Enviado: quinta-feira, 4 de agosto de 2016 11:35
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
Internet, from the script.
Hi Liviu.
Thank you very much.
By the way, do you know if it is possible to discovery what is the IP address
from where the OpenSIPS is running?
For example, the node can have an ethernet IP or a WLAN IP. It depends on if
the node is connected to the DHCP server via wireless or cabe.
Can the OpenSIPS script tells me, by some way, what is the private IP address?
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Liviu Chircu <li...@opensips.org>
Enviado: quinta-feira, 4 de agosto de 2016 11:10
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
Internet, from the script.
Hi, Rodrigo!
That's quite a fun question. Off the top of my head, here are 3 possible ways
in which you can achieve this:
* relay over TCP: not sure if's relevant to your needs, but if you arm a
failure route and t_relay("0x02") out to the internet, you will be able to
properly tell if connectivity was down should you hit the failure route.
* ICMP test: you can do an exec("/bin/ping -w1 -c1 ")
and decide from the return code
* HTTP GET: you can use the rest_client module, attempt to fetch some page, and
decide from the return code. Be sure to set a proper TCP connect timeout!
Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
Home - OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 04.08.2016 16:21, Rodrigo Pimenta Carvalho wrote:
Hi.
How to discover if OpenSIPS is connected do Internet, from its configuration
script?
Sometimes the Internet Link is down and then just local calls will work. If I
can discover if OpenSIPS is "online" on Internet, I will use this information
to implement some specific logic in my script.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to Internet, from the script.
Hi Liviu.
Thank you very much.
By the way, do you know if it is possible to discovery what is the IP address
from where the OpenSIPS is running?
For example, the node can have an ethernet IP or a WLAN IP. It depends on if
the node is connected to the DHCP server via wireless or cabe.
Can the OpenSIPS script tells me, by some way, what is the private IP address?
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Liviu Chircu <li...@opensips.org>
Enviado: quinta-feira, 4 de agosto de 2016 11:10
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to discover if OpenSIPS is connected to
Internet, from the script.
Hi, Rodrigo!
That's quite a fun question. Off the top of my head, here are 3 possible ways
in which you can achieve this:
* relay over TCP: not sure if's relevant to your needs, but if you arm a
failure route and t_relay("0x02") out to the internet, you will be able to
properly tell if connectivity was down should you hit the failure route.
* ICMP test: you can do an exec("/bin/ping -w1 -c1 ")
and decide from the return code
* HTTP GET: you can use the rest_client module, attempt to fetch some page, and
decide from the return code. Be sure to set a proper TCP connect timeout!
Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
Home - OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 04.08.2016 16:21, Rodrigo Pimenta Carvalho wrote:
Hi.
How to discover if OpenSIPS is connected do Internet, from its configuration
script?
Sometimes the Internet Link is down and then just local calls will work. If I
can discover if OpenSIPS is "online" on Internet, I will use this information
to implement some specific logic in my script.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
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Users@lists.opensips.org<mailto:Users@lists.opensips.org>
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[OpenSIPS-Users] How to discover if OpenSIPS is connected to Internet, from the script.
Hi. How to discover if OpenSIPS is connected do Internet, from its configuration script? Sometimes the Internet Link is down and then just local calls will work. If I can discover if OpenSIPS is "online" on Internet, I will use this information to implement some specific logic in my script. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Function set_advertised_address() seems to have wrong decision. Workaround?
Hi.
The function set_advertised_address() is changing the wrong IP in record-routs
from the SIP OK message.
I need to avoid this issue. The caller is in a remote network and the callee is
in the local network.
What is my situation:
An UAS (the callee), running in the same hardware as OpenSIPS, is registered
with IP 127.0.0.1. It could be 192.168.0.100 too. However, due to some
specifics requirements of our project, such UAS must register itself using IP
127.0.0.1, not 192.168.0.100.
What is the problem:
---
When set_advertised_address("domain") is called, for the SIP OK message, this
function decides to change 127.0.0.1 to "domain".
For example, OpenSIPS receives:
SIP/2.0 200 OK
Via: SIP/2.0/TCP
XXX.YYY.240.204:61871;rport=61871;received=131.221.240.204;branch=z9hG4bKPj8bd4d5988f4a4a0ba3599eba77f42600;alias
Record-Route: <sip:127.0.0.1;transport=tcp;lr;r2=on;did=862.47466066>
Record-Route: <sip:192.168.0.100;transport=tcp;lr;r2=on;did=862.47466066>
and change it to:
SIP/2.0 200 OK
Via: SIP/2.0/TCP
XXX.YYY.240.204:61871;rport=61871;received=131.221.240.204;branch=z9hG4bKPj8bd4d5988f4a4a0ba3599eba77f42600;alias
Record-Route: <sip:domain;transport=tcp;lr;r2=on;did=862.47466066>
Record-Route: <sip:192.168.0.100;transport=tcp;lr;r2=on;did=862.47466066>
But, OpenSIPS should change the IP 192.168.0.100 to "domain", not the other
Record-Route.
As I have this issue, the UAC can't send the SIP ACK confirming the SIP OK.
What I need to provide:
I have to get/build a solution to make the set_advertised_address("domain")
change the Record-Route that contains the IP 1921.168.0.100.
Maybe, the OpenSIPS always change the top most Record-Route. If it is true, I
need a workaround for it.
So, how can I fix the record-routs as I need? Does it make sense to do what I'm
needing?
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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[OpenSIPS-Users] Is it possible to read parts of SDP? What is the module to do it?
Hi. A SDP message is: v=0 o=Z 0 0 IN IP4 192.168.21.40 s=Z c=IN IP4 192.168.21.40 t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv It can be in a SIP INVITE or in a SIP OK message. How can I read the IP4 from there, in case of SIP INVITE or SIP OK, and get the value 192.168.21.40 for example? Is there a module and function that provides such information in my script? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Get concurrent calls from sip server.
Hi. In my case I used to run: echo $'dlg_list\n' | xargs ./opensipsctl fifo > RespostasFIFO.txt or ./opensipsctl fifo dlg_list > RespostasFIFO.txt The file RespostasFIFO.txt will be created automatically. I my script I also have loadmodule "dialog.so". Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Carlos Eduardo <kad...@gmail.com> Enviado: quarta-feira, 27 de julho de 2016 16:20 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] Get concurrent calls from sip server. Cesar, Are you using the dialog module in your script? This MI command will only return a valid value if the dialog module is loaded (loadmodule "dialog.so") and if the dialogs are criated durign the script processing (create_dialog() command). 2016-07-27 16:12 GMT-03:00 Daniel Zanutti <daniel.zanu...@gmail.com<mailto:daniel.zanu...@gmail.com>>: On my sample, you should run: opensipsctl fifo profile_get_size inbound Using dlg_list you should get something like this: # opensipsctl fifo dlg_list dialog:: hash=84:1411689852 state:: 4 user_flags:: 0 timestart:: 1469646635 datestart:: 2016-07-27 16:10:35 timeout:: 1469653835 dateout:: 2016-07-27 18:10:35 ... dialog:: hash=289:324429409 state:: 2 user_flags:: 0 timestart:: 0 timeout:: 0 ... dialog:: hash=640:1695114669 state:: 4 ... Check if modules are successfully loaded. Regards On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro <c...@transtelco.net<mailto:c...@transtelco.net>> wrote: Thanks for your help. I trying to use FIFO in order to send requests to OpenSIPS, I have read some documentation about the Dialog Module. I guess using the "opensipsctl fifo dlg_list" command can be useful to obtain the current calls, but I am not sure why the command is not available in my OpenSIPS version. When I execute the command ./opensipsctl fifo version, I am getting the following information Server:: OpenSIPS (1.8.2-notls (x86_64/linux)). On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <pime...@inatel.br<mailto:pime...@inatel.br>> wrote: Now, thinking more about it, I would suggest you to put a SQL query in your proprietary software to query the OpenSIPS database directly. The table dialog will be always updated about current calls. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979 De: users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org> <users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org>> em nome de Rodrigo Pimenta Carvalho <pime...@inatel.br<mailto:pime...@inatel.br>> Enviado: quarta-feira, 27 de julho de 2016 14:39 Para: users@lists.opensips.org<mailto:users@lists.opensips.org> Assunto: Re: [OpenSIPS-Users] Get concurrent calls from sip server. With FIFO you can send requests to OpenSIPS, for example from a proprietary software. So, if a request wants to execute a query with avpops, I guess FIFO will be useful. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979 De: users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org> <users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org>> em nome de Cesar Alberto Rodriguez Fierro <c...@transtelco.net<mailto:c...@transtelco.net>> Enviado: quarta-feira, 27 de julho de 2016 14:29 Para: users@lists.opensips.org<mailto:users@lists.opensips.org> Assunto: [OpenSIPS-Users] Get concurrent calls from sip server. Hi ! I am currently working in a project related with display in real time the active calls of our VoIP traffic, I would like to get the active sip-calls from a Kamailio Sip Server (running opensips), is there any way to obtain this information. Best Regards. [Inline image 1] |Cesar Rodriguez | VoiceOPS | MX: +52 656-257-4112<tel:%2B52%20656-257-4112> | [https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif] CONFIDENTIALITY NOTICE: This communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential, and exempt from disclosure under applicable law. If you are not the intended recipient of this information, you are notified that any use, dissemination, distribution, or copying of the communication is strictly prohibited. ___ Users mailing li
Re: [OpenSIPS-Users] Get concurrent calls from sip server.
Now, thinking more about it, I would suggest you to put a SQL query in your proprietary software to query the OpenSIPS database directly. The table dialog will be always updated about current calls. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Rodrigo Pimenta Carvalho <pime...@inatel.br> Enviado: quarta-feira, 27 de julho de 2016 14:39 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] Get concurrent calls from sip server. With FIFO you can send requests to OpenSIPS, for example from a proprietary software. So, if a request wants to execute a query with avpops, I guess FIFO will be useful. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Cesar Alberto Rodriguez Fierro <c...@transtelco.net> Enviado: quarta-feira, 27 de julho de 2016 14:29 Para: users@lists.opensips.org Assunto: [OpenSIPS-Users] Get concurrent calls from sip server. Hi ! I am currently working in a project related with display in real time the active calls of our VoIP traffic, I would like to get the active sip-calls from a Kamailio Sip Server (running opensips), is there any way to obtain this information. Best Regards. [Inline image 1] |Cesar Rodriguez | VoiceOPS | MX: +52 656-257-4112<tel:%2B52%20656-257-4112> | [https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif] CONFIDENTIALITY NOTICE: This communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential, and exempt from disclosure under applicable law. If you are not the intended recipient of this information, you are notified that any use, dissemination, distribution, or copying of the communication is strictly prohibited. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Get concurrent calls from sip server.
With FIFO you can send requests to OpenSIPS, for example from a proprietary software. So, if a request wants to execute a query with avpops, I guess FIFO will be useful. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Cesar Alberto Rodriguez Fierro <c...@transtelco.net> Enviado: quarta-feira, 27 de julho de 2016 14:29 Para: users@lists.opensips.org Assunto: [OpenSIPS-Users] Get concurrent calls from sip server. Hi ! I am currently working in a project related with display in real time the active calls of our VoIP traffic, I would like to get the active sip-calls from a Kamailio Sip Server (running opensips), is there any way to obtain this information. Best Regards. [Inline image 1] |Cesar Rodriguez | VoiceOPS | MX: +52 656-257-4112<tel:%2B52%20656-257-4112> | [https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif] CONFIDENTIALITY NOTICE: This communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential, and exempt from disclosure under applicable law. If you are not the intended recipient of this information, you are notified that any use, dissemination, distribution, or copying of the communication is strictly prohibited. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Get concurrent calls from sip server.
Hi. Table dialog from the OpenSIPS database will contain information useful for your needing. You will have to query the database to get data from such table and then handle data as you need. For querying the database, see about http://www.opensips.org/html/docs/modules/2.2.x/avpops.html Regards. AVPops Module - OpenSIPS<http://www.opensips.org/html/docs/modules/2.2.x/avpops.html> www.opensips.org AVPops (AVP-operations) modules implements a set of script functions which allow access and manipulation of user AVPs (preferences) and pseudo-variables. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Cesar Alberto Rodriguez Fierro <c...@transtelco.net> Enviado: quarta-feira, 27 de julho de 2016 14:29 Para: users@lists.opensips.org Assunto: [OpenSIPS-Users] Get concurrent calls from sip server. Hi ! I am currently working in a project related with display in real time the active calls of our VoIP traffic, I would like to get the active sip-calls from a Kamailio Sip Server (running opensips), is there any way to obtain this information. Best Regards. [Inline image 1] |Cesar Rodriguez | VoiceOPS | MX: +52 656-257-4112<tel:%2B52%20656-257-4112> | [https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif] CONFIDENTIALITY NOTICE: This communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential, and exempt from disclosure under applicable law. If you are not the intended recipient of this information, you are notified that any use, dissemination, distribution, or copying of the communication is strictly prohibited. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?
Sorry. The rest of message: 1 - check the IP from SDP. 2 - If such IP is equal to the IP of the gateway, then: we will fix the current IP, changing it to the OpenSIPS public IP. Hopefully, it is easy to read such IP from SDP, using some function of OpenSIPS. Could you comment about this decision, please? Thank you very much! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Rodrigo Pimenta Carvalho Enviado: quarta-feira, 27 de julho de 2016 14:04 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do? Hi. It sounds good for who is using rtpproxy. In our case we are using direct media without rtpproxy. As long as our OpenSIPS will be behind a NAT and such NAT is a gateway, we have decided to do the following: RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Johan De Clercq <jo...@democon.be> Enviado: quarta-feira, 27 de julho de 2016 11:57 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do? in the startup route, you should find your pub ip using curl. Put that in a var and then use that var in rtpproxy. 2016-07-27 15:07 GMT+02:00 Rodrigo Pimenta Carvalho <pime...@inatel.br<mailto:pime...@inatel.br>>: Dear OpenSIPS-users. The function nat_uac_test() (from module NATHELPER) works very well and tells me whether a cliente to my OpenSIPS is behind a NAT. But, for my specific network topology, is my OpenSIPS that is behind a NAT, from the client perspective. In this case I have to fix the SDP content that goes from OpenSIPS to the client, so that the client will be able to send its media to a public IP, when communicating to a peer in the same network domain of this SIP server. How can I be sure that for a client perspective the OpenSIPS is behind a NAT? In another words, is there a way to the OpenSIPS determine if its client is in the same network or in a remote network? -- P.S. I suspect that OpenSIPS should be used in a node with public IP to simplify our solution. But our customer asked us to put OpenSIPS in a residential device that will be always behind a NAT for some smartphones perspective and not behind a NAT for another home devices perspective. --- Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979 ___ 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
Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?
Hi. It sounds good for who is using rtpproxy. In our case we are using direct media without rtpproxy. As long as our OpenSIPS will be behind a NAT and such NAT is a gateway, we have decided to do the following: RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Johan De Clercq <jo...@democon.be> Enviado: quarta-feira, 27 de julho de 2016 11:57 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do? in the startup route, you should find your pub ip using curl. Put that in a var and then use that var in rtpproxy. 2016-07-27 15:07 GMT+02:00 Rodrigo Pimenta Carvalho <pime...@inatel.br<mailto:pime...@inatel.br>>: Dear OpenSIPS-users. The function nat_uac_test() (from module NATHELPER) works very well and tells me whether a cliente to my OpenSIPS is behind a NAT. But, for my specific network topology, is my OpenSIPS that is behind a NAT, from the client perspective. In this case I have to fix the SDP content that goes from OpenSIPS to the client, so that the client will be able to send its media to a public IP, when communicating to a peer in the same network domain of this SIP server. How can I be sure that for a client perspective the OpenSIPS is behind a NAT? In another words, is there a way to the OpenSIPS determine if its client is in the same network or in a remote network? -- P.S. I suspect that OpenSIPS should be used in a node with public IP to simplify our solution. But our customer asked us to put OpenSIPS in a residential device that will be always behind a NAT for some smartphones perspective and not behind a NAT for another home devices perspective. --- Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979 ___ 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
[OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?
Dear OpenSIPS-users. The function nat_uac_test() (from module NATHELPER) works very well and tells me whether a cliente to my OpenSIPS is behind a NAT. But, for my specific network topology, is my OpenSIPS that is behind a NAT, from the client perspective. In this case I have to fix the SDP content that goes from OpenSIPS to the client, so that the client will be able to send its media to a public IP, when communicating to a peer in the same network domain of this SIP server. How can I be sure that for a client perspective the OpenSIPS is behind a NAT? In another words, is there a way to the OpenSIPS determine if its client is in the same network or in a remote network? -- P.S. I suspect that OpenSIPS should be used in a node with public IP to simplify our solution. But our customer asked us to put OpenSIPS in a residential device that will be always behind a NAT for some smartphones perspective and not behind a NAT for another home devices perspective. --- Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS and Internet of Things. Where to find examples.
Dear OpenSIPS users; This year, in November, I will teach classes about Internet of Things, SIP, SIP Proxy and messaging systems, at Inatel. I would like to speak about the use of SIP and SIP Proxies in projects related to Internet of Things. But, I am beginning right now looking for the use of SIP in Internet of Things and possibly the use of SIP Proxies in such context. Does someone here could point me some web site or youtube video that shows any example of using OpenSIPS or any SIP Proxy in an Internet of Things context, please? For example, some article, or site, or document that shows what is the future of SIP in Internet of Things or why a SIP Proxy is useful in IoT will be very appreciated. Any indication will be very helpful! Thanks a lot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpensSIPS 2.2 memory leak issue fixed today.
Dear OpenSIPS users; Daniel Fússia has just pulled a request in GitHub. See the link: https://github.com/OpenSIPS/opensips/pull/919 <https://github.com/OpenSIPS/opensips/pull/919> It has one commit with 4 files, fixing some issues related to memory leaks and SQLite. This is for OpenSIPS 2.2 HEAD. We will appreciate comments about it. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Does OpenSIPS use SQLite configuration to hold on queries when database is locked?
Dear Ionut.
Thank you very much for the precise explanation!
I got the point.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Ionut Ionita <ionution...@opensips.org>
Enviado: quarta-feira, 29 de junho de 2016 11:02
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] Does OpenSIPS use SQLite configuration to hold on
queries when database is locked?
Hi Rodrigo,
I am the one who implemented SQLITE module. First of all I wasn't aware of
the BUSY_TIMEOUT option, but still if I was I wouldn't use it and I'll explain
why. Setting such a parameter will cause the OpenSIPS processes, except the one
that has the lock on the database, to sleep for X ms. After those X ms the
database may be locked again and you have to sleep again for that amout of
seconds, even though at some point in that sleeping interval you may have had
the change to get the lock. In the current implementation, all processes keep
trying to get the lock, avoiding dead times, when they could have had the lock
but they were sleeping. We can't do anything else while the database is locked
since we need to process the current message in order to get to the next one.
SQLITE has limitations based on the fact that for each query the whole database
is locked, and to explain that I would like to quote official documentation:
"However, client/server database engines (such as PostgreSQL, MySQL, or Oracle)
usually support a higher level of concurrency and allow multiple processes to
be writing to the same database at the same time. This is possible in a
client/server database because there is always a single well-controlled server
process available to coordinate access. If your application has a need for a
lot of concurrency, then you should consider using a client/server database.
But experience suggests that most applications need much less concurrency than
their designers imagine."[0]
We would be glad to implement such a mechanism if our software would
benefit from it, but in my humble opinion it would bring nothing useful for our
module.
[0] http://www.sqlite.org/faq.html#q5
Regards,
Ionut Ionita
OpenSIPS Developer
On 06/28/2016 11:52 PM, Rodrigo Pimenta Carvalho wrote:
Dear OpenSIPS users,
In my software, programmed in QT (framework for C++) and handling data with a
SQLite database, I have used this:
pDb.setConnectOptions("QSQLITE_BUSY_TIMEOUT=6000");
That is, if SQLite complains that "database is locked" sometime when my
software tries to register some datum there, such database keeps the query
paused (hold on), and then after 6 seconds let the query execute. This
mechanism is transparent for my software and certify that the query will be
tried every 6 seconds until it complete successfully.
Is there something similar to it in OpenSIPS ? That is, does OpenSIPS uses some
kind of configuration provided by SQLite?
If not, why the developers team decided not to use such mechanism?
Any comment will be very helpful!
Thanks a lot!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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[OpenSIPS-Users] Does OpenSIPS use SQLite configuration to hold on queries when database is locked?
Dear OpenSIPS users,
In my software, programmed in QT (framework for C++) and handling data with a
SQLite database, I have used this:
pDb.setConnectOptions("QSQLITE_BUSY_TIMEOUT=6000");
That is, if SQLite complains that "database is locked" sometime when my
software tries to register some datum there, such database keeps the query
paused (hold on), and then after 6 seconds let the query execute. This
mechanism is transparent for my software and certify that the query will be
tried every 6 seconds until it complete successfully.
Is there something similar to it in OpenSIPS ? That is, does OpenSIPS uses some
kind of configuration provided by SQLite?
If not, why the developers team decided not to use such mechanism?
Any comment will be very helpful!
Thanks a lot!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
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Re: [OpenSIPS-Users] I have a Patch that fixes memory leak on OpenSIPS. How to apply this path via github?
Hi Razvan Thank you very much for the instructions and the alert! I will fork and pull that. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Razvan Crainea <raz...@opensips.org> Enviado: terça-feira, 28 de junho de 2016 04:19 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] I have a Patch that fixes memory leak on OpenSIPS. How to apply this path via github? Hi, Rodrigo! Please fork opensips and open a pull request on Github[1]. The idea is simple: 1. fork the repository[2] 2. Apply the patch, commit it and push it in your fork 3. Open a pull request[3] [1] https://github.com/OpenSIPS/opensips/pulls [2] https://help.github.com/articles/fork-a-repo/ [3] https://help.github.com/articles/using-pull-requests/#initiating-the-pull-request PS: it is not a good idea to attach a file on a mailing list. Use gist.github.com, or pastebin.com next time :). Thanks and regards, Razvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> On 06/27/2016 09:05 PM, Rodrigo Pimenta Carvalho wrote: Dear OpenSIPS users, Daniel Fússia, from Inatel Competence Center (<http://www.inatel.br>www.inatel.br<http://www.inatel.br>) has discovered some issues related to the code in OpenSIPS 2.2 that handles some transactions in SQLite. He also has proposed the solution for such issues and his work is attached on this message. How could I resquet to the OpenSIPS development team to apply this fix? That is, can someone here give me the instructions on how to use github and request that fix? I? very new on github. Any hint will be very helpful! Thanks alot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 Brazil ___ 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
Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?
Hi Johan and Ben. Yes. AVPops is a easy solution. However, it is easy for dada stored in DB. What about data stored in RAM? I'm using db_mode = 0 for module usrloc (so user location is always in RAM). So, if AVPops could extract data from the RAM too, as it does with queries and DB, it would be very easy. I have looked for a solution using avp_db_query, but it works only over DB, not over RAM. That is why I started trying to use function lookup() and avp attr, to get caller specific information. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de johan de clercq <jo...@democon.be> Enviado: quarta-feira, 22 de junho de 2016 07:35 Para: 'OpenSIPS users mailling list'; 'sevpal' Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Ben is correct. In my opinion, a very easy solution. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Newlin, Ben Sent: Tuesday, June 21, 2016 5:24 PM To: sevpal <sev...@aol.com>; OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? It also seems like AVPOPS module [1] may be a good solution here as it has functions to pull data from a database into AVPs based by user. [1] http://www.opensips.org/html/docs/modules/2.2.x/avpops.html Ben Newlin From: <users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org>> on behalf of sevpal <sev...@aol.com<mailto:sev...@aol.com>> Reply-To: sevpal <sev...@aol.com<mailto:sev...@aol.com>>, OpenSIPS users mailling list <users@lists.opensips.org<mailto:users@lists.opensips.org>> Date: Tuesday, June 21, 2016 at 11:20 AM To: OpenSIPS users mailling list <users@lists.opensips.org<mailto:users@lists.opensips.org>> Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi, have you tried/considered running a simple query on the database and parsing for the information you need? From: Rodrigo Pimenta Carvalho<mailto:pime...@inatel.br> Sent: Tuesday, June 21, 2016 10:39 AM To: OpenSIPS users mailling list<mailto:users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects? Hi Răzvan. I have tried that idea. But that didn't work. The SIP INVITE message is being changed by the OpenSIPS in a wrong way, in my point of view. Do you know some way to save the entire SIP INVITE message before calling lookup() and then make the saved message take place after the lookup() execution? My original message is: INVITE sip:6...@mydomain.com.br SIP/2.0 Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215 From: <sip:crdphmacl_spnuv5xqt...@mydomain.com.br>;tag=179920819 To: <sip:6...@mydomain.com.br> Call-ID: 1410250893 CSeq: 21 INVITE Contact: <sip:crdphmacl_spnuv5xqt...@xxx.yyy.240.204:60033;transport=tcp> Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", realm="localhost", nonce="5769458c01cc263a7c0d6995dc48d42288ec6f8e4048", uri="sip:6...@mydomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 227 This is being changed to: INVITE sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81 SIP/2.0 Record-Route: <sip:myDomain.com.br;transport=tcp;lr;nat=yes;did=0b.b9e0cfe5> Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1 Via: SIP/2.0/TCP 192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970 From: <sip:crdphmacl_spnuv5xqt...@mydomain.com.br>;tag=12586028 To: <sip:6...@mydomain.com.br> Call-ID: 1106771604 CSeq: 21 INVITE Contact: <sip:crdphmacl_spnuv5xqt...@xxx.yyy.240.204:60672;transport=tcp> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.0.0) Subject: Phone call Content-Length: 224 So, the caller is receiving its own SIP INVITE. That is why when A calls B, is A that rings, not B. It is becoming a bit complicated. So, I suspect I'm going to the incorrect direction Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org<m
[OpenSIPS-Users] I have a Patch that fixes memory leak on OpenSIPS. How to apply this path via github?
Dear OpenSIPS users, Daniel Fússia, from Inatel Competence Center (www.inatel.br<http://www.inatel.br>) has discovered some issues related to the code in OpenSIPS 2.2 that handles some transactions in SQLite. He also has proposed the solution for such issues and his work is attached on this message. How could I resquet to the OpenSIPS development team to apply this fix? That is, can someone here give me the instructions on how to use github and request that fix? I? very new on github. Any hint will be very helpful! Thanks alot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 Brazil 0001-Fix-memory-leak-after-sqlite-prepare-was-deleted-stm.patch Description: 0001-Fix-memory-leak-after-sqlite-prepare-was-deleted-stm.patch ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to configure opensips to show passwords in column password from table subscriber?
Hi. When I execute opensipsctl add user password, the column password in table subscriber remains empty. Ha1 has a kind of encrypted password. How to configure opensips to show passwords in column password from table subscriber? I have tried changing some parameters in module auth_db, but it didn't take effect. So, what is the correct configuration? Any hint will be very helpful! Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?
Hi Răzvan Crainea.
Thank you very much for trying to help me.
Yesterday my boss asked me to work in another part of our project. So, I will
have to pause this verification for a while. When I return to it, I will check
the log.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Răzvan Crainea <raz...@opensips.org>
Enviado: quarta-feira, 22 de junho de 2016 03:57
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi, Rodrigo!
Can you print the $ru variable before and after each lookup() query? Something
like:
$var(ru) = $ru;
xlog("R-URI before caller lookup: $ru\n");
lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I
think is more suitable than from contact uri
$ru = $var(ru);
# continue your processing here
xlog("R-URI after caller lookup: $ru\n");
...
# now do the real lookup for the callee
xlog("R-URI before callee lookup: $ru\n");
lookup("location");
xlog("R-URI after callee lookup: $ru\n");
Make sure they are all correct, or if they are not, send me these logs.
Thanks,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com<http://www.opensips-solutions.com>
Home — OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 06/21/2016 07:39 PM, Rodrigo Pimenta Carvalho wrote:
Hi Sevpal.
Yes. That is what I was doing. It worked very well.
But, nowadays I'm using db_mode = 0 for usrloc. So, the information is always
only in RAM. In this case, the query will return no result. That is why I'm
trying to read the attr column from table location, from RAM, and get specific
information for the caller.
For the callee, everything is all right.
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org>
<users-boun...@lists.opensips.org><mailto:users-boun...@lists.opensips.org> em
nome de sevpal <sev...@aol.com><mailto:sev...@aol.com>
Enviado: terça-feira, 21 de junho de 2016 12:20
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi, have you tried/considered running a simple query on the database and
parsing for the information you need?
From: Rodrigo Pimenta Carvalho<mailto:pime...@inatel.br>
Sent: Tuesday, June 21, 2016 10:39 AM
To: OpenSIPS users mailling list<mailto:users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi Răzvan.
I have tried that idea. But that didn't work. The SIP INVITE message is being
changed by the OpenSIPS in a wrong way, in my point of view.
Do you know some way to save the entire SIP INVITE message before calling
lookup() and then make the saved message take place after the lookup()
execution?
My original message is:
INVITE sip:6...@mydomain.com.br<mailto:sip:6...@mydomain.com.br> SIP/2.0
Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215
From: <sip:crdphmacl_spnuv5xqt...@mydomain.com.br>;tag=179920819
To: <sip:6...@mydomain.com.br>
Call-ID: 1410250893
CSeq: 21 INVITE
Contact:
<sip:crdphmacl_spnuv5xqt...@xxx.yyy.240.204:60033;transport=tcp><mailto:sip:crdphmacl_spnuv5xqt...@xxx.yyy.240.204:60033;transport=tcp>
Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX",
realm="localhost", nonce="5769458c01cc263a7c0d6995dc48d42288ec6f8e4048",
uri="sip:6...@mydomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430",
algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length: 227
This is being changed to:
INVITE
sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81<mailto:sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81>
SIP/2.0
Record-Route: <sip:myDomain.com.br;transport=tcp;lr;nat=yes;did=0b.b9e0cfe5>
Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1
Via: SIP/2.0/TCP
192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970
From: <sip:crdphmacl_spnuv5xqt...@mydomain.com.br>;tag=12586028
To: <sip:6...@mydom
Re: [OpenSIPS-Users] Leak AVPOS + SQLITE
Ok.
Thank you very much!
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Razvan Crainea <raz...@opensips.org>
Enviado: quarta-feira, 22 de junho de 2016 04:04
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Leak AVPOS + SQLITE
Hi, Rodrigo!
Valgrind may report some memory allocated, and not freed, but that is not
necessarily a memory leak. There is a single block of 1024 bytes not freed
during runtime, so I think that is peanuts. The memory used by OpenSIPS is not
allocated with malloc, so cannot be traced by valgrind.
Regarding the system memory, it is normal to decrease as OpenSIPS uses that
memory during runtime. However, after some time, this should stabilize.
Anyhow, sometimes the system memory might generate false alarms, so if you are
tracing any memory leaks, you should check OpenSIPS's internal statistics.
Best regards,
Razvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com<http://www.opensips-solutions.com>
Home - OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 06/21/2016 10:38 PM, Rodrigo Pimenta Carvalho wrote:
Hi.
Does someone here is getting/handling memory leaks with OpenSIPS 2.2 and last
version of SQLite?
I'm using newest commit from OpenSIPS 2.2 and newest version of SQLite.
My query is :
avp_db_query("select Value from GeneralConfigurations where Attribute =
'CONFIGURATION_INTERCOM_A_NAME'");
Valgrind shows:
==16087== ERROR SUMMARY: 0 errors from 0 contexts (suppressed: 2 from 2)
==16088== Searching for pointers to 296,489 not-freed blocks
==16088== Checked 103,297,688 bytes
==16088==
==16088== 1,024 bytes in 1 blocks are possibly lost in loss record 184 of 246
==16088==at 0x4C2745D: malloc (in
/usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so)
==16088==by 0x8F8B05F: sqlite3MemMalloc (sqlite3.c:20167)
==16088==by 0x8F701C7: sqlite3Malloc (sqlite3.c:23846)
==16088==by 0x8F75459: pcache1Alloc (sqlite3.c:44312)
==16088==by 0x8F8019F: sqlite3BtreeCursor (sqlite3.c:44455)
==16088==by 0x8FD0FDD: sqlite3VdbeExec (sqlite3.c:80098)
==16088==by 0x8FDB89F: sqlite3_step (sqlite3.c:75131)
==16088==by 0x8FDC9A1: sqlite3_exec (sqlite3.c:108122)
==16088==by 0x8D20736: db_sqlite_raw_query (dbase.c:358)
==16088==by 0x9464DB8: db_query_avp (avpops_db.c:436)
==16088==by 0x946943E: ops_dbquery_avps (avpops_impl.c:840)
==16088==by 0x9459A61: w_dbquery_avps (avpops.c:1395)
==16088==
==16088== LEAK SUMMARY:
==16088==definitely lost: 0 bytes in 0 blocks
==16088==indirectly lost: 0 bytes in 0 blocks
==16088== possibly lost: 1,024 bytes in 1 blocks
==16088==still reachable: 47,457,573 bytes in 296,488 blocks
==16088== suppressed: 0 bytes in 0 blocks
==16088== Reachable blocks (those to which a pointer was found) are not shown.
==16088== To see them, rerun with: --leak-check=full --show-leak-kinds=all
After some time running that query, I can see, via command top, that the
available memory is decreasing.
In fact, the memory is not freed even after stop running the query for a time.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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[OpenSIPS-Users] Leak AVPOS + SQLITE
Hi.
Does someone here is getting/handling memory leaks with OpenSIPS 2.2 and last
version of SQLite?
I'm using newest commit from OpenSIPS 2.2 and newest version of SQLite.
My query is :
avp_db_query("select Value from GeneralConfigurations where Attribute =
'CONFIGURATION_INTERCOM_A_NAME'");
Valgrind shows:
==16087== ERROR SUMMARY: 0 errors from 0 contexts (suppressed: 2 from 2)
==16088== Searching for pointers to 296,489 not-freed blocks
==16088== Checked 103,297,688 bytes
==16088==
==16088== 1,024 bytes in 1 blocks are possibly lost in loss record 184 of 246
==16088==at 0x4C2745D: malloc (in
/usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so)
==16088==by 0x8F8B05F: sqlite3MemMalloc (sqlite3.c:20167)
==16088==by 0x8F701C7: sqlite3Malloc (sqlite3.c:23846)
==16088==by 0x8F75459: pcache1Alloc (sqlite3.c:44312)
==16088==by 0x8F8019F: sqlite3BtreeCursor (sqlite3.c:44455)
==16088==by 0x8FD0FDD: sqlite3VdbeExec (sqlite3.c:80098)
==16088==by 0x8FDB89F: sqlite3_step (sqlite3.c:75131)
==16088==by 0x8FDC9A1: sqlite3_exec (sqlite3.c:108122)
==16088==by 0x8D20736: db_sqlite_raw_query (dbase.c:358)
==16088==by 0x9464DB8: db_query_avp (avpops_db.c:436)
==16088==by 0x946943E: ops_dbquery_avps (avpops_impl.c:840)
==16088==by 0x9459A61: w_dbquery_avps (avpops.c:1395)
==16088==
==16088== LEAK SUMMARY:
==16088==definitely lost: 0 bytes in 0 blocks
==16088==indirectly lost: 0 bytes in 0 blocks
==16088== possibly lost: 1,024 bytes in 1 blocks
==16088==still reachable: 47,457,573 bytes in 296,488 blocks
==16088== suppressed: 0 bytes in 0 blocks
==16088== Reachable blocks (those to which a pointer was found) are not shown.
==16088== To see them, rerun with: --leak-check=full --show-leak-kinds=all
After some time running that query, I can see, via command top, that the
available memory is decreasing.
In fact, the memory is not freed even after stop running the query for a time.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?
Hi Sevpal.
Yes. That is what I was doing. It worked very well.
But, nowadays I'm using db_mode = 0 for usrloc. So, the information is always
only in RAM. In this case, the query will return no result. That is why I'm
trying to read the attr column from table location, from RAM, and get specific
information for the caller.
For the callee, everything is all right.
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de sevpal <sev...@aol.com>
Enviado: terça-feira, 21 de junho de 2016 12:20
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi, have you tried/considered running a simple query on the database and
parsing for the information you need?
From: Rodrigo Pimenta Carvalho<mailto:pime...@inatel.br>
Sent: Tuesday, June 21, 2016 10:39 AM
To: OpenSIPS users mailling list<mailto:users@lists.opensips.org>
Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi Răzvan.
I have tried that idea. But that didn't work. The SIP INVITE message is being
changed by the OpenSIPS in a wrong way, in my point of view.
Do you know some way to save the entire SIP INVITE message before calling
lookup() and then make the saved message take place after the lookup()
execution?
My original message is:
INVITE sip:6...@mydomain.com.br SIP/2.0
Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215
From: <sip:crdphmacl_spnuv5xqt...@mydomain.com.br>;tag=179920819
To: <sip:6...@mydomain.com.br>
Call-ID: 1410250893
CSeq: 21 INVITE
Contact: <sip:crdphmacl_spnuv5xqt...@xxx.yyy.240.204:60033;transport=tcp>
Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX",
realm="localhost", nonce="5769458c01cc263a7c0d6995dc48d42288ec6f8e4048",
uri="sip:6...@mydomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430",
algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length: 227
This is being changed to:
INVITE
sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81
SIP/2.0
Record-Route: <sip:myDomain.com.br;transport=tcp;lr;nat=yes;did=0b.b9e0cfe5>
Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1
Via: SIP/2.0/TCP
192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970
From: <sip:crdphmacl_spnuv5xqt...@mydomain.com.br>;tag=12586028
To: <sip:6...@mydomain.com.br>
Call-ID: 1106771604
CSeq: 21 INVITE
Contact: <sip:crdphmacl_spnuv5xqt...@xxx.yyy.240.204:60672;transport=tcp>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length: 224
So, the caller is receiving its own SIP INVITE.
That is why when A calls B, is A that rings, not B.
It is becoming a bit complicated. So, I suspect I'm going to the incorrect
direction
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Răzvan Crainea <raz...@opensips.org>
Enviado: terça-feira, 21 de junho de 2016 04:24
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi, Rodrigo!
Have you tried restoring the R-URI after the caller lookup? Something like:
$var(ru) = $ru;
lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I
think is more suitable than from contact uri
$ru = $var(ru);
# continue your processing here
# now do the real lookup for the callee
lookup("location");
Don't do the lookups in the reversed way, because you might loose some contacts.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com<http://www.opensips-solutions.com>
Home - OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote:
Dear OpenSIPS-users,
The table location has the column attr where I use to store specific additional
information for each registration.
Whenever A calls B, I have to read this specific information from the A record
and
Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?
Hi Răzvan.
I have tried that idea. But that didn't work. The SIP INVITE message is being
changed by the OpenSIPS in a wrong way, in my point of view.
Do you know some way to save the entire SIP INVITE message before calling
lookup() and then make the saved message take place after the lookup()
execution?
My original message is:
INVITE sip:6...@mydomain.com.br SIP/2.0
Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215
From: <sip:crdphmacl_spnuv5xqt...@mydomain.com.br>;tag=179920819
To: <sip:6...@mydomain.com.br>
Call-ID: 1410250893
CSeq: 21 INVITE
Contact: <sip:crdphmacl_spnuv5xqt...@xxx.yyy.240.204:60033;transport=tcp>
Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX",
realm="localhost", nonce="5769458c01cc263a7c0d6995dc48d42288ec6f8e4048",
uri="sip:6...@mydomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430",
algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length: 227
This is being changed to:
INVITE
sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81
SIP/2.0
Record-Route: <sip:myDomain.com.br;transport=tcp;lr;nat=yes;did=0b.b9e0cfe5>
Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1
Via: SIP/2.0/TCP
192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970
From: <sip:crdphmacl_spnuv5xqt...@mydomain.com.br>;tag=12586028
To: <sip:6...@mydomain.com.br>
Call-ID: 1106771604
CSeq: 21 INVITE
Contact: <sip:crdphmacl_spnuv5xqt...@xxx.yyy.240.204:60672;transport=tcp>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length: 224
So, the caller is receiving its own SIP INVITE.
That is why when A calls B, is A that rings, not B.
It is becoming a bit complicated. So, I suspect I'm going to the incorrect
direction
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Răzvan Crainea <raz...@opensips.org>
Enviado: terça-feira, 21 de junho de 2016 04:24
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi, Rodrigo!
Have you tried restoring the R-URI after the caller lookup? Something like:
$var(ru) = $ru;
lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I
think is more suitable than from contact uri
$ru = $var(ru);
# continue your processing here
# now do the real lookup for the callee
lookup("location");
Don't do the lookups in the reversed way, because you might loose some contacts.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com<http://www.opensips-solutions.com>
Home — OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote:
Dear OpenSIPS-users,
The table location has the column attr where I use to store specific additional
information for each registration.
Whenever A calls B, I have to read this specific information from the A record
and from the B record. That is, I need to get and handle specific information
about the caller and callee.
For the callee, I use to invoke the lookup("location") function that put the
needed information in the attr_avp. That is good and works very well. Then, I
just have to read the attr_avp to get such specific information.
For the caller, I use to invoke:
$var(aorChamador) = $(ct.fields(uri));
lookup("location","","$var(aorChamador)");
However it causes amazing side effect in the SIP signaling. Ex: When A calls B,
B stays quiet and A rings. So A can answer A. Crazy!
According to the documentation, lookup will overwritten the Request-URI. I
guess that is why the SIP signaling become incoherent.
How could I get the caller attr specific information without side effects?
Any hint will be very helpful!!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
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Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?
Hi Razvan Crainea.
I didn't know about this possibility.
I will try this idea now.
Thank you very much!!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Razvan Crainea <raz...@opensips.org>
Enviado: terça-feira, 21 de junho de 2016 04:24
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the
caller, without side effects?
Hi, Rodrigo!
Have you tried restoring the R-URI after the caller lookup? Something like:
$var(ru) = $ru;
lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I
think is more suitable than from contact uri
$ru = $var(ru);
# continue your processing here
# now do the real lookup for the callee
lookup("location");
Don't do the lookups in the reversed way, because you might loose some contacts.
Best regards,
Razvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com<http://www.opensips-solutions.com>
On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote:
Dear OpenSIPS-users,
The table location has the column attr where I use to store specific additional
information for each registration.
Whenever A calls B, I have to read this specific information from the A record
and from the B record. That is, I need to get and handle specific information
about the caller and callee.
For the callee, I use to invoke the lookup("location") function that put the
needed information in the attr_avp. That is good and works very well. Then, I
just have to read the attr_avp to get such specific information.
For the caller, I use to invoke:
$var(aorChamador) = $(ct.fields(uri));
lookup("location","","$var(aorChamador)");
However it causes amazing side effect in the SIP signaling. Ex: When A calls B,
B stays quiet and A rings. So A can answer A. Crazy!
According to the documentation, lookup will overwritten the Request-URI. I
guess that is why the SIP signaling become incoherent.
How could I get the caller attr specific information without side effects?
Any hint will be very helpful!!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
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[OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?
Dear OpenSIPS-users,
The table location has the column attr where I use to store specific additional
information for each registration.
Whenever A calls B, I have to read this specific information from the A record
and from the B record. That is, I need to get and handle specific information
about the caller and callee.
For the callee, I use to invoke the lookup("location") function that put the
needed information in the attr_avp. That is good and works very well. Then, I
just have to read the attr_avp to get such specific information.
For the caller, I use to invoke:
$var(aorChamador) = $(ct.fields(uri));
lookup("location","","$var(aorChamador)");
However it causes amazing side effect in the SIP signaling. Ex: When A calls B,
B stays quiet and A rings. So A can answer A. Crazy!
According to the documentation, lookup will overwritten the Request-URI. I
guess that is why the SIP signaling become incoherent.
How could I get the caller attr specific information without side effects?
Any hint will be very helpful!!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
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Users@lists.opensips.org
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Re: [OpenSIPS-Users] Parser memory Leak found.
Hi.
In addiction to my last message, the same problem also exists to the following
configuration:
modparam("uri", "use_uri_table", 0)
modparam("uri", "db_url", "sqlite:///usr/local/opensips/db/sisc.sqlite") #
CUSTOMIZE ME
If use_uri_table is equal to zero, we must comment the line that declare db_url.
But, what kind of side effect could I get with such decision?
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: sisc-requ...@listas.inatel.br <sisc-requ...@listas.inatel.br> em nome de
Rodrigo Pimenta Carvalho <pime...@inatel.br>
Enviado: sexta-feira, 17 de junho de 2016 14:57
Para: users@lists.opensips.org
Assunto: [sisc] Parser memory Leak found.
Hi.
Thanks Daniel Fússia, a coworker in my office, now I'm sending more details
about the memory leak we saw in OpenSIPS 2.2 (newest commit from today):
The following configuration doesn't causes memory leaks:
modparam("dialog", "db_mode", 1)
modparam("dialog", "db_url", "sqlite:///usr/local/opensips/db/sisc.sqlite")
But, when db_mode is 0, it causes memory leak. The problem is that when using
db_mode = 0 we can't declare db_url. That is, if db_mode is zero, we have to
comment the line that declares db_url.
See below the valgrind logs. One for the case without memory leak and another
one with the issue.
-
==1792==
==1792== HEAP SUMMARY:
==1792== in use at exit: 3,142,778 bytes in 2,894 blocks
==1792== total heap usage: 9,463 allocs, 6,569 frees, 4,960,116 bytes
allocated
==1792==
==1792== LEAK SUMMARY:
==1792==definitely lost: 0 bytes in 0 blocks
==1792==indirectly lost: 0 bytes in 0 blocks
==1792== possibly lost: 0 bytes in 0 blocks
==1792==still reachable: 3,142,778 bytes in 2,894 blocks
==1792== suppressed: 0 bytes in 0 blocks
==1792== Reachable blocks (those to which a pointer was found) are not shown.
==1792== To see them, rerun with: --leak-check=full --show-leak-kinds=all
==1792==
Now with the isse:
---
Thank you for flying opensips
==1762==
==1762== HEAP SUMMARY:
==1762== in use at exit: 2,887,898 bytes in 2,193 blocks
==1762== total heap usage: 7,991 allocs, 5,798 frees, 4,382,036 bytes
allocated
==1762==
==1762== 80 bytes in 1 blocks are definitely lost in loss record 31 of 100
==1762==at 0x4C2745D: malloc (in
/usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so)
==1762==by 0x52D50B9: strdup (strdup.c:42)
==1762==by 0x4DF87E: set_mod_param_regex (modparam.c:97)
==1762==by 0x5AD2FB: yyparse (cfg.y:1085)
==1762==by 0x4177DE: main (main.c:999)
==1762==
==1762== LEAK SUMMARY:
==1762==definitely lost: 80 bytes in 1 blocks
==1762==indirectly lost: 0 bytes in 0 blocks
==1762== possibly lost: 0 bytes in 0 blocks
==1762==still reachable: 2,887,818 bytes in 2,192 blocks
==1762== suppressed: 0 bytes in 0 blocks
==1762== Reachable blocks (those to which a pointer was found) are not shown.
==1762== To see them, rerun with: --leak-check=full --show-leak-kinds=all
==1762==
-
The problem rises in the yyparser. The parser causes a memory leak whenever
db_mode is zero and we still declare db_url, just in dialog module.
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
________
De: Daniel Lopes Fússia
Enviado: sexta-feira, 17 de junho de 2016 14:41
Para: Rodrigo Pimenta Carvalho
Assunto: Leak no Parser
Pimenta,
Os logs e as configurações estão em anexo.
Qualquer dúvida me dá um tok.
Att,
Daniel Fussia
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[OpenSIPS-Users] Parser memory Leak found.
Hi.
Thanks Daniel Fússia, a coworker in my office, now I'm sending more details
about the memory leak we saw in OpenSIPS 2.2 (newest commit from today):
The following configuration doesn't causes memory leaks:
modparam("dialog", "db_mode", 1)
modparam("dialog", "db_url", "sqlite:///usr/local/opensips/db/sisc.sqlite")
But, when db_mode is 0, it causes memory leak. The problem is that when using
db_mode = 0 we can't declare db_url. That is, if db_mode is zero, we have to
comment the line that declares db_url.
See below the valgrind logs. One for the case without memory leak and another
one with the issue.
-
==1792==
==1792== HEAP SUMMARY:
==1792== in use at exit: 3,142,778 bytes in 2,894 blocks
==1792== total heap usage: 9,463 allocs, 6,569 frees, 4,960,116 bytes
allocated
==1792==
==1792== LEAK SUMMARY:
==1792==definitely lost: 0 bytes in 0 blocks
==1792==indirectly lost: 0 bytes in 0 blocks
==1792== possibly lost: 0 bytes in 0 blocks
==1792==still reachable: 3,142,778 bytes in 2,894 blocks
==1792== suppressed: 0 bytes in 0 blocks
==1792== Reachable blocks (those to which a pointer was found) are not shown.
==1792== To see them, rerun with: --leak-check=full --show-leak-kinds=all
==1792==
Now with the isse:
---
Thank you for flying opensips
==1762==
==1762== HEAP SUMMARY:
==1762== in use at exit: 2,887,898 bytes in 2,193 blocks
==1762== total heap usage: 7,991 allocs, 5,798 frees, 4,382,036 bytes
allocated
==1762==
==1762== 80 bytes in 1 blocks are definitely lost in loss record 31 of 100
==1762==at 0x4C2745D: malloc (in
/usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so)
==1762==by 0x52D50B9: strdup (strdup.c:42)
==1762==by 0x4DF87E: set_mod_param_regex (modparam.c:97)
==1762==by 0x5AD2FB: yyparse (cfg.y:1085)
==1762==by 0x4177DE: main (main.c:999)
==1762==
==1762== LEAK SUMMARY:
==1762==definitely lost: 80 bytes in 1 blocks
==1762==indirectly lost: 0 bytes in 0 blocks
==1762== possibly lost: 0 bytes in 0 blocks
==1762==still reachable: 2,887,818 bytes in 2,192 blocks
==1762== suppressed: 0 bytes in 0 blocks
==1762== Reachable blocks (those to which a pointer was found) are not shown.
==1762== To see them, rerun with: --leak-check=full --show-leak-kinds=all
==1762==
-
The problem rises in the yyparser. The parser causes a memory leak whenever
db_mode is zero and we still declare db_url, just in dialog module.
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: Daniel Lopes Fússia
Enviado: sexta-feira, 17 de junho de 2016 14:41
Para: Rodrigo Pimenta Carvalho
Assunto: Leak no Parser
Pimenta,
Os logs e as configurações estão em anexo.
Qualquer dúvida me dá um tok.
Att,
Daniel Fussia
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Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak.
Hi.
We discovered another memory leak in OpenSIPS 2.2, even using newest SQLite.
Now the issue doesn't relate to the data base. There is a issue related with a
parser.
In few minutes I will post here more details, with valgrind log.
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Rodrigo Pimenta Carvalho <pime...@inatel.br>
Enviado: sexta-feira, 17 de junho de 2016 14:23
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with
OpenSIPS 2.2? We investigate a memory leak.
Hi Eric.
Probably not. Because I still don't know what is a federated-sip. And I didn't
have to take control of RTPs in opensips script.
However, a coworker in my office will check these details and help us to
conclude more things about it.
Is there a quick way to check if someone is using such federated-sip?
Our version 3.8.6 of SQLite presented the memory leak (when there was no SIP
requests), but the version 3.13 doesn't present.
P.S.: I still have to read about federated SIP and see what are its advantages.
Thanks a lot.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Eric Tamme <e...@uphreak.com>
Enviado: sexta-feira, 17 de junho de 2016 14:12
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with
OpenSIPS 2.2? We investigate a memory leak.
Hey Rodrigo,
Are you running https://github.com/etamme/federated-sip by chance? Your use of
the PCRE module made me think you might be. I run federated-sip and I do use
sqlite3 with opensips - my current sqlite version is:
sqlite-3.7.17-4.el7.x86_64
[https://avatars3.githubusercontent.com/u/21685?v=3=400]<https://github.com/etamme/federated-sip>
GitHub - etamme/federated-sip: Federated SIP
deployment<https://github.com/etamme/federated-sip>
github.com
README.md Federated SIP server. The Federated SIP project is a set of scripts
designed to run OpenSIPS + rtpengine in a way that will provide federated, open
...
I do not know that I have memory leaks outside of what I reported in the github
issue.
-Eric
On 06/17/2016 11:08 AM, Rodrigo Pimenta Carvalho wrote:
Thank you Ionut.
We will try it so.
Today morning, we noticed that OpenSIPS 2.2 while running and using SQLite,
without online clients, without registers and without calls, causes a memory
leak. That is, OpenSIPS even without any SIP request causes a memory leak due
to the use of SQLite.
After updating the SQLite to a new version, such memory leak was vanished.
However, even with the newest SQLite, we still get memory leaks again if the
proxy receives SIP REGISTER messages. That is, we get the issue every time some
client registers. In this case we saw the memory leak in : "
modparam("db_sqlite", "load_extension", "/usr/lib/sqlite3/pcre.so")"
Let us try the new solution and see what happens.
Best regards!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org>
<users-boun...@lists.opensips.org><mailto:users-boun...@lists.opensips.org> em
nome de Ionut Ionita <ionution...@opensips.org><mailto:ionution...@opensips.org>
Enviado: sexta-feira, 17 de junho de 2016 11:45
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with
OpenSIPS 2.2? We investigate a memory leak.
Hi Rodrigo,
Pushed a fix both into 2.2[0] and master[1] branches. If you still think
sqlite leaks even with this fix,
please feel free to open an issue on github.
[0]
https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf
[https://avatars3.githubusercontent.com/u/7924437?v=3=200]<https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf>
[sqlite][bugfix] free column names when freeing the result ·
OpenSIPS/opensips@c1aa55e<https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf>
github.com
(cherry picked from commit cf380773cec9f91ad08e343c03261154dabc36a5)
[1]
https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf
Regrads,
Ionut Ionita
OpenSIPS Developer
On 06/17/2016 05:19 PM, Rodrigo Pimenta Carvalho wrote:
Hi Liviu.
Very good.
We will see the resolution process.
Thank you very much!
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
D
Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak.
Hi Eric.
Probably not. Because I still don't know what is a federated-sip. And I didn't
have to take control of RTPs in opensips script.
However, a coworker in my office will check these details and help us to
conclude more things about it.
Is there a quick way to check if someone is using such federated-sip?
Our version 3.8.6 of SQLite presented the memory leak (when there was no SIP
requests), but the version 3.13 doesn't present.
P.S.: I still have to read about federated SIP and see what are its advantages.
Thanks a lot.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Eric Tamme <e...@uphreak.com>
Enviado: sexta-feira, 17 de junho de 2016 14:12
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with
OpenSIPS 2.2? We investigate a memory leak.
Hey Rodrigo,
Are you running https://github.com/etamme/federated-sip by chance? Your use of
the PCRE module made me think you might be. I run federated-sip and I do use
sqlite3 with opensips - my current sqlite version is:
sqlite-3.7.17-4.el7.x86_64
[https://avatars3.githubusercontent.com/u/21685?v=3=400]<https://github.com/etamme/federated-sip>
GitHub - etamme/federated-sip: Federated SIP
deployment<https://github.com/etamme/federated-sip>
github.com
README.md Federated SIP server. The Federated SIP project is a set of scripts
designed to run OpenSIPS + rtpengine in a way that will provide federated, open
...
I do not know that I have memory leaks outside of what I reported in the github
issue.
-Eric
On 06/17/2016 11:08 AM, Rodrigo Pimenta Carvalho wrote:
Thank you Ionut.
We will try it so.
Today morning, we noticed that OpenSIPS 2.2 while running and using SQLite,
without online clients, without registers and without calls, causes a memory
leak. That is, OpenSIPS even without any SIP request causes a memory leak due
to the use of SQLite.
After updating the SQLite to a new version, such memory leak was vanished.
However, even with the newest SQLite, we still get memory leaks again if the
proxy receives SIP REGISTER messages. That is, we get the issue every time some
client registers. In this case we saw the memory leak in : "
modparam("db_sqlite", "load_extension", "/usr/lib/sqlite3/pcre.so")"
Let us try the new solution and see what happens.
Best regards!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org>
<users-boun...@lists.opensips.org><mailto:users-boun...@lists.opensips.org> em
nome de Ionut Ionita <ionution...@opensips.org><mailto:ionution...@opensips.org>
Enviado: sexta-feira, 17 de junho de 2016 11:45
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with
OpenSIPS 2.2? We investigate a memory leak.
Hi Rodrigo,
Pushed a fix both into 2.2[0] and master[1] branches. If you still think
sqlite leaks even with this fix,
please feel free to open an issue on github.
[0]
https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf
[https://avatars3.githubusercontent.com/u/7924437?v=3=200]<https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf>
[sqlite][bugfix] free column names when freeing the result ·
OpenSIPS/opensips@c1aa55e<https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf>
github.com
(cherry picked from commit cf380773cec9f91ad08e343c03261154dabc36a5)
[1]
https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf
Regrads,
Ionut Ionita
OpenSIPS Developer
On 06/17/2016 05:19 PM, Rodrigo Pimenta Carvalho wrote:
Hi Liviu.
Very good.
We will see the resolution process.
Thank you very much!
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org>
<mailto:users-boun...@lists.opensips.org>
<users-boun...@lists.opensips.org><mailto:users-boun...@lists.opensips.org> em
nome de Liviu Chircu <li...@opensips.org><mailto:li...@opensips.org>
Enviado: sexta-feira, 17 de junho de 2016 11:14
Para: users@lists.opensips.org<mailto:users@lists.opensips.org>
Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with
OpenSIPS 2.2? We investigate a memory leak.
Hi Rodrigo!
A GitHub issue [1] regarding this leak was just reported today by Eric, so you
can track the resolution process over there! You can even subscribe to that
ticket if you
Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak.
Thank you Ionut.
We will try it so.
Today morning, we noticed that OpenSIPS 2.2 while running and using SQLite,
without online clients, without registers and without calls, causes a memory
leak. That is, OpenSIPS even without any SIP request causes a memory leak due
to the use of SQLite.
After updating the SQLite to a new version, such memory leak was vanished.
However, even with the newest SQLite, we still get memory leaks again if the
proxy receives SIP REGISTER messages. That is, we get the issue every time some
client registers. In this case we saw the memory leak in : "
modparam("db_sqlite", "load_extension", "/usr/lib/sqlite3/pcre.so")"
Let us try the new solution and see what happens.
Best regards!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Ionut Ionita <ionution...@opensips.org>
Enviado: sexta-feira, 17 de junho de 2016 11:45
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with
OpenSIPS 2.2? We investigate a memory leak.
Hi Rodrigo,
Pushed a fix both into 2.2[0] and master[1] branches. If you still think
sqlite leaks even with this fix,
please feel free to open an issue on github.
[0]
https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf
[https://avatars3.githubusercontent.com/u/7924437?v=3=200]<https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf>
[sqlite][bugfix] free column names when freeing the result ·
OpenSIPS/opensips@c1aa55e<https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf>
github.com
(cherry picked from commit cf380773cec9f91ad08e343c03261154dabc36a5)
[1]
https://github.com/OpenSIPS/opensips/commit/c1aa55e73f7c56ea4894997aeb7f25a03cb289cf
Regrads,
Ionut Ionita
OpenSIPS Developer
On 06/17/2016 05:19 PM, Rodrigo Pimenta Carvalho wrote:
Hi Liviu.
Very good.
We will see the resolution process.
Thank you very much!
Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org<mailto:users-boun...@lists.opensips.org>
<users-boun...@lists.opensips.org><mailto:users-boun...@lists.opensips.org> em
nome de Liviu Chircu <li...@opensips.org><mailto:li...@opensips.org>
Enviado: sexta-feira, 17 de junho de 2016 11:14
Para: users@lists.opensips.org<mailto:users@lists.opensips.org>
Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with
OpenSIPS 2.2? We investigate a memory leak.
Hi Rodrigo!
A GitHub issue [1] regarding this leak was just reported today by Eric, so you
can track the resolution process over there! You can even subscribe to that
ticket if you have an account, in order to receive emails.
[1]: https://github.com/OpenSIPS/opensips/issues/911
[https://avatars3.githubusercontent.com/u/21685?v=3=400]<https://github.com/OpenSIPS/opensips/issues/911>
2.2 runs out of pkg_mem because of db/db_res.c memory leak · Issue #911 ·
OpenSIPS/opensips<https://github.com/OpenSIPS/opensips/issues/911>
github.com
OpenSIPS 2.2 will run out of pkg_mem, i believe because of a leak in
db/db_res.c Full memlog dump is available here:
https://gist.github.com/etamme/7d42024ad684fe834b9fd514d2bd2412 I am using...
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote:
Hi.
People from my team is investigating a memory leak related to OpenSIPS 2.2.
As I had commented in another discussion in the past, it seems that the problem
comes from SQLite we are using as the Registrar for our OpenSIPS 2.2.
For example, a script opensips.cfg that doesn't use SQLite didn't cause memory
leak. But, a script that uses it and use another module that needs a database
(EX: auth.so) causes memory leak.
We are still in the beginning of the investigation.
So, what is the best version of SQLite to be used with OpenSIPS 2.2? That is,
what version of SQLite was very well tested with OpenSIPS 2.2 and worked
without memory leaks or others issues?
Any suggestion will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak.
Hi Liviu. Very good. We will see the resolution process. Thank you very much! Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Liviu Chircu <li...@opensips.org> Enviado: sexta-feira, 17 de junho de 2016 11:14 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak. Hi Rodrigo! A GitHub issue [1] regarding this leak was just reported today by Eric, so you can track the resolution process over there! You can even subscribe to that ticket if you have an account, in order to receive emails. [1]: https://github.com/OpenSIPS/opensips/issues/911 [https://avatars3.githubusercontent.com/u/21685?v=3=400]<https://github.com/OpenSIPS/opensips/issues/911> 2.2 runs out of pkg_mem because of db/db_res.c memory leak · Issue #911 · OpenSIPS/opensips<https://github.com/OpenSIPS/opensips/issues/911> github.com OpenSIPS 2.2 will run out of pkg_mem, i believe because of a leak in db/db_res.c Full memlog dump is available here: https://gist.github.com/etamme/7d42024ad684fe834b9fd514d2bd2412 I am using... Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.06.2016 17:02, Rodrigo Pimenta Carvalho wrote: Hi. People from my team is investigating a memory leak related to OpenSIPS 2.2. As I had commented in another discussion in the past, it seems that the problem comes from SQLite we are using as the Registrar for our OpenSIPS 2.2. For example, a script opensips.cfg that doesn't use SQLite didn't cause memory leak. But, a script that uses it and use another module that needs a database (EX: auth.so) causes memory leak. We are still in the beginning of the investigation. So, what is the best version of SQLite to be used with OpenSIPS 2.2? That is, what version of SQLite was very well tested with OpenSIPS 2.2 and worked without memory leaks or others issues? Any suggestion will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ 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
[OpenSIPS-Users] What is the best SQLite version to be used with OpenSIPS 2.2? We investigate a memory leak.
Hi. People from my team is investigating a memory leak related to OpenSIPS 2.2. As I had commented in another discussion in the past, it seems that the problem comes from SQLite we are using as the Registrar for our OpenSIPS 2.2. For example, a script opensips.cfg that doesn't use SQLite didn't cause memory leak. But, a script that uses it and use another module that needs a database (EX: auth.so) causes memory leak. We are still in the beginning of the investigation. So, what is the best version of SQLite to be used with OpenSIPS 2.2? That is, what version of SQLite was very well tested with OpenSIPS 2.2 and worked without memory leaks or others issues? Any suggestion will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Script to get attr_avp from caller. How to do?
Hi. Razvan Crainea.
Thank you very much!
I will try it.
As I can see, I should start by understanding the function lookup and is
capabilities very well.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Razvan Crainea <raz...@opensips.org>
Enviado: quinta-feira, 16 de junho de 2016 05:58
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Script to get attr_avp from caller. How to do?
Hi, Rodrigo!
You can do something like this:
lookup("location","","$fu"); # note that this might fail, so you'll have
to treat this separately
$avp(caller_attr) = $avp(attr); # note that the caller might have
multiple contacts/attributes, so you'll have to handle this
Then continue with your logic, call lookup("location") and will load the
attributes of the callee, per branch, as in your snippet.
Regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
[http://www.opensips-solutions.com/imgs/slideshow/slide1.jpg]<http://www.opensips-solutions.com/>
Home - OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is
more than a SIP proxy/router as it includes application-level functionalities.
On 06/15/2016 09:36 PM, Rodrigo Pimenta Carvalho wrote:
>
> Hi.
>
>
> In my script I use to use AVP to store specific additional information
> for each registration.
>
> When I have to get the attr_avp (Module Registrar) from the callee,
> during a call, I use to codify:
>
>
> branch_route[per_branch_ops] {
>
> myVariable = $(avp(attr)[$T_branch_idx]
>
> }
>
>
> What about the caller? I also need to get the attr_avp from caller, but
> I can't find a similar code in OpenSIPS documentation. How to do?
>
>
> P.S.: I'm using the configuration: modparam("usrloc", "db_mode", 0)
>
>
> Any hint will be very helpful!
>
> Best regards.
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
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[OpenSIPS-Users] Script to get attr_avp from caller. How to do?
Hi.
In my script I use to use AVP to store specific additional information for each
registration.
When I have to get the attr_avp (Module Registrar) from the callee, during a
call, I use to codify:
branch_route[per_branch_ops] {
myVariable = $(avp(attr)[$T_branch_idx]
}
What about the caller? I also need to get the attr_avp from caller, but I can't
find a similar code in OpenSIPS documentation. How to do?
P.S.: I'm using the configuration: modparam("usrloc", "db_mode", 0)
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
Users mailing list
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to update table location, but directly on memory cache (RAM)?
Hi Bogdan-Andrei.
You are right. I have been used the attr_avp, as you explained, to save a
specific information in each new record for table location. It works very well
and such information goes to column attr.
However, I have created today a new column for such table: column callerName.
And I have to save $fn in this new column for each new record too.
So, what I have just tried today is something like this:
modparam("registrar", "attr_avp", "$avp(attr)")
modparam("registrar", "attr_avp", "$avp(callerName)")
...
is_method("REGISTER")) {
$avp(attr) = "my_specific_information";
$avp(callerName) = $fn;
}
...
But, in this case, the $fn overwrites the specific information, because it
seems that attr_avp will pointer always to the same column Attr, no matter what
the name I give to the AVP.
Do you know how to put every information in its correct column?
Is it possble to have two attr_avps related to two different columns in table
location?
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Bogdan-Andrei Iancu <bog...@opensips.org>
Enviado: sexta-feira, 10 de junho de 2016 07:39
Para: OpenSIPS users mailling list
Cc: Cleide Aparecida Ribeiro do Prado; Daniel Lopes Fússia
Assunto: Re: [OpenSIPS-Users] How to update table location, but directly on
memory cache (RAM)?
Hi Rodrigo,
What you try to do is not consistent.
Either you use db_mode 1 to be have immediate writting in DB from usrloc module
(see http://www.opensips.org/html/docs/modules/1.11.x/usrloc.html#id294459) ->
it will be safe to run your script query after the save().
Either push the extra info you want to save into DB (and memory cache) via the
attr AVP (see
http://www.opensips.org/html/docs/modules/1.11.x/registrar.html#id293909) and
opensips will do everything for you.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 09.06.2016 17:42, Rodrigo Pimenta Carvalho wrote:
Hi.
My script has the configuration:
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "timer_interval",3)
Always after receiving a new register in table location, I must to execute a
code like this:
avp_db_query("UPDATE location...
That is, an update will complement data in the new register.
However, how could I immediately update table location if data might be in
memory cache (RAM) for 3 seconds. It could fail obviously.
The command avp_db_query UPDATE is acting over the database on hard disc, not
in obviously.
So, is there a way to update table location even still in cache (RAM)? If yes,
when data from RAM is recorded into the database, the register will already be
updated.
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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[OpenSIPS-Users] How to update table location, but directly on memory cache (RAM)?
Hi.
My script has the configuration:
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "timer_interval",3)
Always after receiving a new register in table location, I must to execute a
code like this:
avp_db_query("UPDATE location...
That is, an update will complement data in the new register.
However, how could I immediately update table location if data might be in
memory cache (RAM) for 3 seconds. It could fail obviously.
The command avp_db_query UPDATE is acting over the database on hard disc, not
in obviously.
So, is there a way to update table location even still in cache (RAM)? If yes,
when data from RAM is recorded into the database, the register will already be
updated.
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log
Hi Steve.
The values that you have commented is used by the "make menuconfig". That is,
as long as these are the values chosen by the standard configuration, I believe
that such values wouldn't be causing problems.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: Steve Woolley <steve.wool...@me.com>
Enviado: quarta-feira, 8 de junho de 2016 09:12
Para: OpenSIPS users mailling list
Cc: Bogdan-Andrei Iancu; Rodrigo Pimenta Carvalho
Assunto: Re: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task
already scheduled" messages in log
So i made two changes to my config:
1. per Bogdan, changed my timer_partitions to: modparam("tm",
"timer_partitions", 4)
2. I had previously used some settings found in a number of configuration
examples in my transaction module. After a little research, I commented them
out. Still doing some research into whether these new (default) values may
trigger other problems. My new config is as so (snippet):
Transaction Module
loadmodule "tm.so"
# modparam("tm", "fr_timeout", 5)
# modparam("tm", "fr_inv_timeout", 30)
# modparam("tm", "restart_fr_on_each_reply", 0)
# modparam("tm", "onreply_avp_mode", 1)
modparam("tm", "timer_partitions", 4)
Since making these changes and restarting, the messages have gone away.
On Jun 7, 2016, at 6:47 AM, Bogdan-Andrei Iancu
<bog...@opensips.org<mailto:bog...@opensips.org>> wrote:
Hello Steve,
What OpenSIPS tells you is that you the TM timer routine (which gets executed
once per second) takes longer than 1 second (as execution). Probably you have
retransmissions or many failure routes to be executed.
You can increase the level of parallelism in TM timer via the timer_partition
parameter:
http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294483
Try to set it to 4.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com<http://www.opensips-solutions.com/>
On 07.06.2016 04:21, Steve Woolley wrote:
Running opensips on a Raspberry Pi. At some point after starting opensips
(sometimes immediately, sometimes after quite a bit of time), the following
message fills the log — once a second — continuously.
...
Jun 7 01:10:41 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8258580 ms), it may
overlap..
Jun 7 01:10:42 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8259570 ms), it may
overlap..
Jun 7 01:10:43 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8260560 ms), it may
overlap..
Jun 7 01:10:44 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8261560 ms), it may
overlap..
Jun 7 01:10:45 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8262550 ms), it may
overlap..
Jun 7 01:10:46 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8263550 ms), it may
overlap..
Jun 7 01:10:47 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8264540 ms), it may
overlap..
Jun 7 01:10:48 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8265530 ms), it may
overlap..
Jun 7 01:10:49 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8266530 ms), it may
overlap..
Jun 7 01:10:50 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8267520 ms), it may
overlap..
root@pi1:~# opensips -V
version: opensips 2.2.0 (arm6/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC,
DBG_MALLOC, USE_PTHREAD_MUTEX
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.
git revision: 0e1cea7
main.c compiled on 21:56:59 Jun 3 2016 with gcc 4.9.2
Anyone experiencing the same?
--
Steve Woolley
steve.wool...@me.com<mailto:steve.wool...@me.com>
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Re: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task already scheduled" messages in log
Thank you!
I will compare your configuration with mine.
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: Steve Woolley <steve.wool...@me.com>
Enviado: quarta-feira, 8 de junho de 2016 09:12
Para: OpenSIPS users mailling list
Cc: Bogdan-Andrei Iancu; Rodrigo Pimenta Carvalho
Assunto: Re: [OpenSIPS-Users] Continuous "WARNING:core:timer_ticker: timer task
already scheduled" messages in log
So i made two changes to my config:
1. per Bogdan, changed my timer_partitions to: modparam("tm",
"timer_partitions", 4)
2. I had previously used some settings found in a number of configuration
examples in my transaction module. After a little research, I commented them
out. Still doing some research into whether these new (default) values may
trigger other problems. My new config is as so (snippet):
Transaction Module
loadmodule "tm.so"
# modparam("tm", "fr_timeout", 5)
# modparam("tm", "fr_inv_timeout", 30)
# modparam("tm", "restart_fr_on_each_reply", 0)
# modparam("tm", "onreply_avp_mode", 1)
modparam("tm", "timer_partitions", 4)
Since making these changes and restarting, the messages have gone away.
On Jun 7, 2016, at 6:47 AM, Bogdan-Andrei Iancu
<bog...@opensips.org<mailto:bog...@opensips.org>> wrote:
Hello Steve,
What OpenSIPS tells you is that you the TM timer routine (which gets executed
once per second) takes longer than 1 second (as execution). Probably you have
retransmissions or many failure routes to be executed.
You can increase the level of parallelism in TM timer via the timer_partition
parameter:
http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294483
Try to set it to 4.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com<http://www.opensips-solutions.com/>
On 07.06.2016 04:21, Steve Woolley wrote:
Running opensips on a Raspberry Pi. At some point after starting opensips
(sometimes immediately, sometimes after quite a bit of time), the following
message fills the log — once a second — continuously.
...
Jun 7 01:10:41 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8258580 ms), it may
overlap..
Jun 7 01:10:42 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8259570 ms), it may
overlap..
Jun 7 01:10:43 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8260560 ms), it may
overlap..
Jun 7 01:10:44 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8261560 ms), it may
overlap..
Jun 7 01:10:45 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8262550 ms), it may
overlap..
Jun 7 01:10:46 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8263550 ms), it may
overlap..
Jun 7 01:10:47 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8264540 ms), it may
overlap..
Jun 7 01:10:48 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8265530 ms), it may
overlap..
Jun 7 01:10:49 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8266530 ms), it may
overlap..
Jun 7 01:10:50 pi1 /usr/local/sbin/opensips[22396]: WARNING:core:timer_ticker:
timer task already scheduled for 7182080 ms (now 8267520 ms), it may
overlap..
root@pi1:~# opensips -V
version: opensips 2.2.0 (arm6/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC,
DBG_MALLOC, USE_PTHREAD_MUTEX
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.
git revision: 0e1cea7
main.c compiled on 21:56:59 Jun 3 2016 with gcc 4.9.2
Anyone experiencing the same?
--
Steve Woolley
steve.wool...@me.com<mailto:steve.wool...@me.com>
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