Re: [OpenSIPS-Users] Does create_dialog behave differently on 1.6 than on 1.7?
Hello Saúl, The modifications to the script API should not have affected the behaviour. I have just done a test with the latest OpenSIPS trunk with a very similar scenario ( dialog end parties behind NAT ) and everything went fine ( the caller callee contacts are the fixed ones ). Could you please ngrep on such a call and check if your fixing on the Contact fields actually get applied ? Maybe it's some kind of scripting error. Regards, -- Vlad Paiu OpenSIPS Developer On 08/21/2011 01:28 PM, Saúl Ibarra Corretgé wrote: Hi, I was playing around with a 1.6 test machine and I upgraded it to 1.7. Required adjustments were minor :-) but something feels weird with the dialog module. The setup is really simple: request comes in, NAT is fixed, record route, relay the request. On the 1.6 configuration version I do create_dialog() and just after I set the flag matching the bye_on_timeout_flag parameter from the dialog module. And it works fine. On the 1.7 config, since that flag doesn't exist anymore, I just call create_dialog(B), but something is going wrong: the BYE requests are sent to the private address. If I connect through the fifo and list the dialogs I can clearly see that the caller_contact contains the unfixed contact. However, same config (with those minor changes) in 1.6 does work and I see the fixed contact address there. Is it a bug or am I missing something obvious here? Thanks and regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] topology_hiding without success
Hello, About the crash on the dialog topology_hiding feature, we can not extract any useful information from your backtrace. Seems you did not have the debugging symbols on, maybe ? About the B2B issue, it seems that you did not configure a db_url for the b2b_logic module and that's why OpenSIPS refuses to start. Regards, -- Vlad Paiu OpenSIPS Developer On 08/20/2011 04:47 PM, s...@veliko-turnovo.com wrote: Hello, I try to use topology_hiding without success. version: opensips 1.7.0-beta-notls (i386/linux) svnrevision: 2:8265 1. When I use topology_hiding() opensips crashs. Please see post from Sun, August 7, 2011 11:43 pm 2. When try to use B2B module I receive follow errors: . loadmodule b2b_entities.so loadmodule b2b_logic.so /usr/local/sbin/opensips[26102]: CRITICAL:b2b_logic:child_init: child_init: database not bound /usr/local/sbin/opensips[26102]: ERROR:core:init_mod_child: failed to initializing module b2b_logic, rank 10 /usr/local/sbin/opensips[26106]: CRITICAL:b2b_logic:child_init: child_init: database not bound /usr/local/sbin/opensips[26083]: ERROR:core:init_mod_child: failed to initializing module b2b_logic, rank 1 /usr/local/sbin/opensips[26082]: ERROR:core:init_mod_child: failed to initializing module b2b_logic, rank -2 /usr/local/sbin/opensips[26085]: ERROR:core:init_mod_child: failed to initializing module b2b_logic, rank 2 Where is problem? best regards, Plamen Petkov ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Ability to tell active calls per customer
Hi, I think you could use dialog profile, but not sure. Dani On 08/19/11 23:17, Robert Thomas wrote: Hi, I have a load balancer module to distribute calls among my Gateways. I can use the lb_list command to see the active calls per gw, but I would like something similar to graph my customer amount of active calls. I was thinking creating another set of resources on the load balancer, but this would be messy. Or somehow use the dialog module for this. Ideally if the variable could be exposed via snmp I could use cacti to graph each customer. Has anyone tried this, and what would be the best way? Sent from my iPhone ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] B2BUA | Refer scenario not sending SDP
Hello, I’ve been trying to configure the REFER scenariohttp://www.opensips.org/Resources/B2buaTutorial#toc15but so far no luck. I've successfully called the scenario on initial INVITE and seems like I'm getting B2B responses and replies but when I see sip traces the following two anomalies exist. 1- Refer-To header value is not used anywhere in new INVITE 2- SDP is not sent in new INVITE. Please help. Best Regards, *Sam*** ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Does create_dialog behave differently on 1.6 than on 1.7?
Hi Vlad, On Aug 22, 2011, at 11:52 AM, Vlad Paiu wrote: Hello Saúl, The modifications to the script API should not have affected the behaviour. I have just done a test with the latest OpenSIPS trunk with a very similar scenario ( dialog end parties behind NAT ) and everything went fine ( the caller callee contacts are the fixed ones ). Could you please ngrep on such a call and check if your fixing on the Contact fields actually get applied ? Maybe it's some kind of scripting error. Modifications were actually applied, I'll re-check but call was correctly established, otherwise I would have lost the ACK. It also worked fine on OpenSIPS 1.6.4. In order to test the feature I chained 2 proxies, the first one just fixed the NAT and the second one did the dialog stuff. In this case it worked, since the Contact was already fixed by the first proxy. If I call fix_contact and right after create_dialog will the create_contact dialog 'see' that the contact was modified? Nevertheless I'll test it again tonight. Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Does create_dialog behave differently on 1.6 than on 1.7?
Hello, Are you using fix_contact from the nat_traversal module ? I have only tested with fix_nated_contact from the nathelper module, but I just checked the code and they should both work without any issues. The scenario should work as long as you call the contact fixing function before the create_dialog(), but please check with calling fix_contact() right before of create_dialog(), maybe there is something spooky going on in-between the two function calls. Waiting for you testing results. Thanks. Regards, -- Vlad Paiu OpenSIPS Developer On 08/22/2011 01:32 PM, Saúl Ibarra Corretgé wrote: Hi Vlad, On Aug 22, 2011, at 11:52 AM, Vlad Paiu wrote: Hello Saúl, The modifications to the script API should not have affected the behaviour. I have just done a test with the latest OpenSIPS trunk with a very similar scenario ( dialog end parties behind NAT ) and everything went fine ( the caller callee contacts are the fixed ones ). Could you please ngrep on such a call and check if your fixing on the Contact fields actually get applied ? Maybe it's some kind of scripting error. Modifications were actually applied, I'll re-check but call was correctly established, otherwise I would have lost the ACK. It also worked fine on OpenSIPS 1.6.4. In order to test the feature I chained 2 proxies, the first one just fixed the NAT and the second one did the dialog stuff. In this case it worked, since the Contact was already fixed by the first proxy. If I call fix_contact and right after create_dialog will the create_contact dialog 'see' that the contact was modified? Nevertheless I'll test it again tonight. Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Wrong socket selected by proxy when forwarding requests
Hi Jan, By default OpenSIPS is preserving the socket interface (the outbound interface is the same as inbound interface). A change in the interface is done only if explicitly required from script via functions (like force_send_socket() or lookup(location)) or because of double Record-Routing in the message. What may happen is an improper usage of mhomed param - usually this is used for multi interfaces with different IPs - it cannot make distinction between interfaces with same IP but different ports. So, it will simply use the first interface that matches the needed IP (in this case the 5060 port interface) Regards, Bogdan On 08/19/2011 08:42 PM, Jan Blom wrote: Hi Bogdan, Yes! Best regards, Jan -Original Message- From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: den 19 augusti 2011 17:20 To: OpenSIPS users mailling list Cc: Jan Blom Subject: Re: [OpenSIPS-Users] Wrong socket selected by proxy when forwarding requests Hi Jam, So, what you say is that the A sends ACK on 5002 to opensips and opensips sends it to B via 5060 port ? Regards, Bogdan On 08/15/2011 03:56 PM, Jan Blom wrote: Hello, I have noticed a problem when using mhomed=1. The problem was discovered in 1.6.3, but has also been verified in 1.7.0-beta. I have Opensips (only) listening on two ports on the same IP address. Obviously this doesn't require mhomed=1, but that was a leftover from an older setup. And I wasn't expecting it to cause trouble anyway. This is a snippet from the opensips.cfg: auto_aliases=no mhomed=1 listen=udp:178.16.xxx.xxx:5060 listen=udp:178.16.xxx.xxx:5002 I have two user agents that register with the proxy, both using port 5002. In the socket field from the subscriber table, I can verify that both are correctly associated with port 5002 (having value udp:178.16.xxx.xxx:5002). User agent A then calls user agent B, again by sending the INVITE to port 5002. The INVITE request is properly forwarded to user agent B, using src port 5002. However, the ACK request sent when B answers and the BYE request when B hangs up will both, incorrectly, be forwarded by the proxy using src port 5060. The wrong socket is selected by the proxy. This will then prevent proper call setup and termination if any NAT is present. If I remove the mhomed=1 setting, the problem will disappear. So this is currently not a problem for me since the workaround exists. However, it might be a problem for others as well, where you potentially do need a multi-homed setup. Best regards, Jan Blom People Interactive ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Installation of opensips tls version 1.7
Hi, in this case, was your opensips compiled from sources or still RPMs ? Regards, Bogdan On 08/20/2011 12:44 PM, isshed wrote: Hello Bogdan, Thanks for your reply. My installation was through after doing the following. ln -s ./libmysqlclient.so.16 ./libmysqlclient.so But when I run it is not compinng up. Bellow is the error message on console. INFO: Starting OpenSIPS : /usr/local/sbin/opensipsctl: line 1478: 29359 Segmentation fault (core dumped) $OSIPSBIN -P $PID_FILE $STARTOPTIONS /dev/null 2 /dev/null ERROR: PID file /var/run/opensips/opensips.pid does not exist -- OpenSIPS start failed Could you please assist me setting it up. Thanks Isshed On Fri, Aug 19, 2011 at 8:03 PM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hi, instead of manually downloading the RPM, better use package manager on your OS. If it is debian derived, use apt, if redhat derived use yum . Regards, Bogdan On 08/19/2011 07:38 AM, isshed wrote: Hello Vlad, Can you please let me know from where can i download the rpms. I am using mysql version 5.0-45-6.fc8. Thanks, On Thu, Aug 18, 2011 at 4:57 PM, isshed isshed@gmail.com mailto:isshed@gmail.com wrote: no I did not install it. i could not find the correct rpm.. mysqlclient15-devel-5.0.67-1.fc8.remi.i386.rpm is i downloaded from internet but while installing it says warning: mysqlclient15-devel-5.0.67-1.fc8.remi.i386.rpm: Header V4 DSA signature: NOKEY, key ID 00f97f56 error: Failed dependencies: mysqlclient15 = 5.0.67-1.fc8.remi is needed by mysqlclient15-devel-5.0.67-1.fc8.remi.i386 I would be really thankful to you sir. Thanks On Thu, Aug 18, 2011 at 4:40 PM, Vlad Paiu vladp...@opensips.org mailto:vladp...@opensips.org wrote: Hello, Do you have libmysqlclient-dev installed on your machine ? Seems like an error related to it. Regards, -- Vlad Paiu OpenSIPS Developer On 08/18/2011 02:08 PM, isshed wrote: Hi All, I am installing opensips-1.7.0-beta-tls. make prefix=/usr/local all the problem i am facing is .that it is failing while compiling mysql database. it does not find the files mysql.h etc. How to resolve it. Can you please let me know the link of mysql for fedora8. ___ 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 mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Ability to tell active calls per customer
Hi all, yes, the dialog profiles is the best way to get this done (profiles with values) . There is also some small tutorial on that: http://www.opensips.org/Resources/DocsTutConcurrentCalls Regards, Bogdan On 08/22/2011 01:02 PM, Dani Popa wrote: Hi, I think you could use dialog profile, but not sure. Dani On 08/19/11 23:17, Robert Thomas wrote: Hi, I have a load balancer module to distribute calls among my Gateways. I can use the lb_list command to see the active calls per gw, but I would like something similar to graph my customer amount of active calls. I was thinking creating another set of resources on the load balancer, but this would be messy. Or somehow use the dialog module for this. Ideally if the variable could be exposed via snmp I could use cacti to graph each customer. Has anyone tried this, and what would be the best way? Sent from my iPhone ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] topology_hiding without success
The second part (b2b related) should be fixed - if db_mode is 0 (as per default), b2b will not ask for a db_url to be set. The fix is on SVN trunk 1.7 Regards, Bogdan On 08/22/2011 12:56 PM, Vlad Paiu wrote: Hello, About the crash on the dialog topology_hiding feature, we can not extract any useful information from your backtrace. Seems you did not have the debugging symbols on, maybe ? About the B2B issue, it seems that you did not configure a db_url for the b2b_logic module and that's why OpenSIPS refuses to start. Regards, -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS db schema on PostgreSQL
Hi Ryan, I've just committed a fix for this issue on trunk. Please update your sources and try again. Let us know if your problem was solved. Regards Razvan Crainea OpenSIPS Developer On 11.08.2011 21:34, Ryan Revels wrote: Bogdan, This is a brand new install so no tables exist. Here is the complete output with verbose=1: # opensipsdbctl create -e database engine 'postgres' loaded -e \E[37;33mINFO: creating database opensips ... -e Creating core table: standard NOTICE: CREATE TABLE / UNIQUE will create implicit index version_t_name_idx for table version -e Creating core table: acc NOTICE: CREATE TABLE will create implicit sequence acc_id_seq for serial column acc.id http://acc.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index acc_pkey for table acc NOTICE: CREATE TABLE will create implicit sequence missed_calls_id_seq for serial column missed_calls.id http://missed_calls.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index missed_calls_pkey for table missed_calls -e Creating core table: domain NOTICE: CREATE TABLE will create implicit sequence domain_id_seq for serial column domain.id http://domain.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index domain_pkey for table domain NOTICE: CREATE TABLE / UNIQUE will create implicit index domain_domain_idx for table domain -e Creating core table: group NOTICE: CREATE TABLE will create implicit sequence grp_id_seq for serial column grp.id http://grp.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index grp_pkey for table grp NOTICE: CREATE TABLE / UNIQUE will create implicit index grp_account_group_idx for table grp NOTICE: CREATE TABLE will create implicit sequence re_grp_id_seq for serial column re_grp.id http://re_grp.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index re_grp_pkey for table re_grp -e Creating core table: permissions NOTICE: CREATE TABLE will create implicit sequence address_id_seq for serial column address.id http://address.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index address_pkey for table address -e Creating core table: registrar NOTICE: CREATE TABLE will create implicit sequence aliases_id_seq for serial column aliases.id http://aliases.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index aliases_pkey for table aliases -e Creating core table: usrloc NOTICE: CREATE TABLE will create implicit sequence location_id_seq for serial column location.id http://location.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index location_pkey for table location NOTICE: CREATE TABLE / UNIQUE will create implicit index location_account_contact_idx for table location -e Creating core table: msilo NOTICE: CREATE TABLE will create implicit sequence silo_id_seq for serial column silo.id http://silo.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index silo_pkey for table silo -e Creating core table: alias_db NOTICE: CREATE TABLE will create implicit sequence dbaliases_id_seq for serial column dbaliases.id http://dbaliases.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index dbaliases_pkey for table dbaliases NOTICE: CREATE TABLE / UNIQUE will create implicit index dbaliases_alias_idx for table dbaliases -e Creating core table: uri_db NOTICE: CREATE TABLE will create implicit sequence uri_id_seq for serial column uri.id http://uri.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index uri_pkey for table uri NOTICE: CREATE TABLE / UNIQUE will create implicit index uri_account_idx for table uri -e Creating core table: nathelper NOTICE: CREATE TABLE will create implicit sequence nh_sockets_id_seq for serial column nh_sockets.id http://nh_sockets.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index nh_sockets_pkey for table nh_sockets -e Creating core table: speeddial NOTICE: CREATE TABLE will create implicit sequence speed_dial_id_seq for serial column speed_dial.id http://speed_dial.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index speed_dial_pkey for table speed_dial NOTICE: CREATE TABLE / UNIQUE will create implicit index speed_dial_speed_dial_idx for table speed_dial -e Creating core table: avpops NOTICE: CREATE TABLE will create implicit sequence usr_preferences_id_seq for serial column usr_preferences.id http://usr_preferences.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index usr_preferences_pkey for table usr_preferences -e Creating core table: auth_db NOTICE: CREATE TABLE will create implicit sequence subscriber_id_seq for serial column subscriber.id http://subscriber.id NOTICE: CREATE TABLE / PRIMARY KEY will create implicit index subscriber_pkey for table subscriber NOTICE: CREATE TABLE / UNIQUE will create implicit index subscriber_account_idx for table subscriber -e Creating core table: pdt NOTICE: CREATE TABLE will create implicit sequence pdt_id_seq for serial
Re: [OpenSIPS-Users] Cassandra DB Driver
Hi Pete, I will try to contact him. Thanks and regards, Bogdan On 08/19/2011 11:10 PM, Pete Kelly wrote: It was onsip. This is the contact details at the end of the presentation: https://picasaweb.google.com/106120232325050278321/ClueCon2011Day1?authuser=0feat=directlink#5638975817090500786 https://picasaweb.google.com/106120232325050278321/ClueCon2011Day1?authuser=0feat=directlink#5638975817090500786 develo...@junctionnetworks.com mailto:develo...@junctionnetworks.com John Riordan Want me to get in touch with him? Pete On 19 August 2011 16:44, David J da...@styleflare.com mailto:da...@styleflare.com wrote: Maybe 2600hz On Aug 19, 2011 11:22 AM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hi Pete, I recall that; haven;t received any contact yet and do not remember exactly the name /company of the speaker .something with Whistle ?? Maybe we can dig the ClueCon presentations from the first day. Regards, Bogdan On 08/14/2011 10:38 PM, Pete Kelly wrote: Hi At the ClueCon conference last week, one of the speakers mentioned they have an OpenSIPS Cassandra module that they are considering publishing as an OpenSIPS module. Bogdan do you know if this was discussed any further? Pete ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ 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 -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Some love for the Debian packaging
Hi, I've been testing the 1.7 branch over the weekend and I found some issues with the Debian packaging: - Some packages depend on -dev libraries when they shouldn't - Old Debian standards version - No TLS by default? [1] - Some scripts have /bin/sh in the shebang, but Debian Squeeze (stable) uses dash instead of sh, which doesn't provide the same features - Python module is not built [2] Looking at http://www.opensips.org/Development/Development I couldn't find who is in charge of the Debian packaging, so if there are no objections I'd like to give some love to it :-) [1]: Is there any reason why we don't build the Debian package with TLS by default? Feels very inconvenient if someone wants to test it but the binaries don't have support for it. [2]: The Python module is not built and it lacks documentation. Should we build it? Thanks and regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 1.7.0 - from beta to stable
Hi all, In the last weeks, we received some awesome feedback and support from the community regarding the success or issues the opensips 1.7.0 beta have. Also, a lot of work for troubleshooting and fixing was done, is done and it will be done. Thank you to all people who helped with this - we do appreciate it! The future plan is to have OpenSIPS 1.7.0 moved from beta to stable on 26th of August. Best regards, Bogdan -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Does create_dialog behave differently on 1.6 than on 1.7?
Hi Vlad, On Aug 22, 2011, at 12:43 PM, Vlad Paiu wrote: Hello, Are you using fix_contact from the nat_traversal module ? I have only tested with fix_nated_contact from the nathelper module, but I just checked the code and they should both work without any issues. The scenario should work as long as you call the contact fixing function before the create_dialog(), but please check with calling fix_contact() right before of create_dialog(), maybe there is something spooky going on in-between the two function calls. Waiting for you testing results. Thanks. I did run my test again, same result: works on 1.6.4, doesn't on 1.7. The only changes in the config are the following: 1.6.4: create_dialog(); setflag(5); # for bye_on_dialog_timeout 1.7: create_dialog(BPp); Here is what I noticed when looking at the output of the dlg_list command; 1.6.4 dialog:: hash=2154:1189469910 state:: 4 user_flags:: 0 timestart:: 1314037923 timeout:: 1314037933 callid:: LtlTeeEkNMt2IE9Nxz9bhOKu6vequbDG from_uri:: sip:sagh...@sipdoc.net to_uri:: sip:sag...@sipdoc.net caller_tag:: Bz-vwSGeputSfhxvjj.3yDAdDzbwAMsb caller_contact:: sip:wonqjpfd@62.X.Y.Z:49294 callee_cseq:: 0 caller_route_set:: caller_bind_addr:: udp:91.X.Y.Z:5060 callee_tag:: IPmI.nSTOMSRDJAD6wWdJlN.jt9z4DZc callee_contact:: sip:ohpwrcbd@62.X.Y.Z:50131 caller_cseq:: 10547 callee_route_set:: callee_bind_addr:: udp:91.X.Y.Z:5060 and in 1.7: dialog:: hash=1601:1367800701 state:: 4 user_flags:: 0 timestart:: 1314037262 timeout:: 1314058862 callid:: OrT1izagFZ07KH01VDAFZIweVVO.-d.s from_uri:: sip:sagh...@sipdoc.net to_uri:: sip:sag...@sipdoc.net caller_tag:: KJB84JwXsDlfI3bFkxeX-8Y5pmzLnY83 caller_contact:: sip:wonqjpfd@192.168.X.Y:60942 callee_cseq:: 0 caller_route_set:: caller_bind_addr:: udp:91.X.Y.Z:5060 callee_tag:: iTa9gOmx7GtXQ3R1J96E7-dtYsrwT.Ya callee_contact:: sip:ohpwrcbd@62.X.Y.Z:63447 caller_cseq:: 322 callee_route_set:: callee_bind_addr:: udp:91.X.Y.Z:5060 As you can see, the caller_contact is wrong on the 1.7 output, but its ok on the 1.6.4 output. Any clue? Thanks and regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA | Refer scenario not sending SDP
Hi Sam, Seems that you have a bad understanding of what the Refer scenario must do. Let's say A and B are in a call. When a Refer message is received from A, the B2B will terminate the call leg with A and will try connect B and the URI in the Refer-To header. The connection is done indeed by sending first an Invite without any SDP to B and doing a late SDP negotiation in 200OK and ACK. So what you observed, is in fact the wanted behavior. Regards, Anca Vamanu On Mon, Aug 22, 2011 at 1:07 PM, Sam Govind govoi...@gmail.com wrote: Hello, I’ve been trying to configure the REFER scenariohttp://www.opensips.org/Resources/B2buaTutorial#toc15but so far no luck. I've successfully called the scenario on initial INVITE and seems like I'm getting B2B responses and replies but when I see sip traces the following two anomalies exist. 1- Refer-To header value is not used anywhere in new INVITE 2- SDP is not sent in new INVITE. Please help. Best Regards, *Sam*** ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS+MediaProxy mangling ACKs?
Greetings all; I'm afraid I haven't solved this yet, but I did strip out all the dispatcher code and that hasn't helped, so it appears to be something to do with forwarding the calls out while the mediaproxy is enabled. Unfortunately as calls have to be directed through the specific IP of our own local OpenSER proxy to the carrier I can't bypass this in machine in the call flow for my testing. Tomorrow I'll try and pull mediaproxy back out of the config and see if it really is to blame, but I have to assume so since I have all of the rest of these features (dispatcher/etc) working happily in other environs. I could sure use a suggestion on what I could try removing/adding to help pinpoint what might be going on. Thanks again; - Jock On Fri, Aug 19, 2011 at 11:42 AM, Jock McKechnie jock.mckech...@gmail.comwrote: My deepest apologies. After I'd stuffed it into PasteBin an hour or so later (honestly) it told me it was suspicious of the content and wanted verification, which when I did, it told me the paste had been nuked already. Helpful. The CORRECT PasteBin URI (and working as of _now_ ;) is: http://pastebin.com/b2FGgTRX Thanks! - Jock On Tue, Aug 16, 2011 at 2:13 AM, Saúl Ibarra Corretgé s...@ag-projects.com wrote: Hi, On Aug 15, 2011, at 5:54 PM, Jock McKechnie wrote: Greetings all; We've recently started rolling out MediaProxy devices at work and I've noticed when we're chaining several systems together in a call path that the MediaProxy/OpenSIPS box likes to change the ACK header in a manner which breaks the calls. When MediaProxy gets the ACK it will remove the host information from the URI of the final SBC in the chain and instead replace it with the IP of the proxy that immediately follows the MediaProxy/OpenSIPS box... which, of course, the next OpenSIPS box then sees itself on the ACK and removes the host information, producing a broken ACK that the far-end carrier throws up their hands at and ignores. This is a little complicated, so bear with me, but the call flow looks like this (private IPs are for illustrative purposes only): Call source (currently an Asterisk machine) - 192.168.0.1 - OpenSIPS/MediaProxy system (v.1.7.0 and latest darcs MediaProxy source) - 192.168.1.1 - Local OpenSER proxy (v.1.3.2) - 192.168.2.2 - Carrier proxy (Unknown type) - 10.5.5.5 - Carrier SBC (Sonus) - 10.10.10.10 As I understand it, the ACK is supposed to be formatted like so: ACK sip:+16415551212@10.10.10.10:5060;transport=udp SIP/2.0 Where the ACK has the IP address of the carrier's SBC that's at the very end of the call chain in its URI. Instead the MediaProxy/OpenSIPS box produces an ACK like so: ACK sip:+16415551212@192.168.2.2:5060;transport=udp SIP/2.0 Which is the IP of the final proxy in our company that hands the calls off to the carrier. The proxy then strips its own IP out of the ACK and sends this to the carrier: ACK sip:10.5.5.5;lr;ftag=as6e98d5f7 Without a user in the SIP URI and the wrong IP in the ACK, the carrier completely ignores this response and continues to send '200 OK's which we don't respond to, so eventually the carrier terminates the call as it naturally assumes we went missing. I could sure use some suggestions. The OpenSER box is using a very simple accept and forward stateless configuration and its only job is to aggregate calls from several boxes behind it and send it on to a single carrier address (10.5.5.5). The OpenSIPS config includes MediaProxy and also a local dispatcher file to fail-over calls between local proxies. The MediaProxy config has tested good when used directly between Asterisk and the carrier proxy. If I connect Asterisk directly to the local proxy, the call works fine as there's no funny business going on with the ACKs w/o the OpenSIPS/MediaProxy box in between to rewrite the ACK. MediaProxy only mangles the SDP, it never touches the RURI, so you can discard it as the culprit :-) There are many hops in the scenario you described, so the best would be to look at SIP traces and see which one of the hops is mangling the data in a bogus way and why. The MediaProxy config can be found on PasteBin (for some goofy reason pasting directly into gmail under Opera removes all line-feeds. Handy.) http://pastebin.com/XruQ2rPk I get a Unknown paste ID :-S Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA | Refer scenario not sending SDP
Thanks Anca for the explanation, but can you just tell me if what I'm trying to do here is possible with B2BUA or not? if you are saying that A side Refer-to header will be read then obviously I need to take care of the client1 and server1 parameters accordingly. I've a FreeSwitch as Media-Server to play some IVR to User-A and collect DTMF.OpenSIPS will be acting as B2BUA for this call. Once DTMF are collected, Media-Server Refers the call back to OpenSIPS At OpenSIPS I need to do some accounting on the collected DTMF(I may need more than one XML script to get the control back) If I get OK from AAA then Send INVITE to the DTMF destination else Send INVITE to Media-Server to play some Message. On Tue, Aug 23, 2011 at 12:22 AM, Anca Vamanu anca.vam...@gmail.com wrote: Hi Sam, Seems that you have a bad understanding of what the Refer scenario must do. Let's say A and B are in a call. When a Refer message is received from A, the B2B will terminate the call leg with A and will try connect B and the URI in the Refer-To header. The connection is done indeed by sending first an Invite without any SDP to B and doing a late SDP negotiation in 200OK and ACK. So what you observed, is in fact the wanted behavior. Regards, Anca Vamanu On Mon, Aug 22, 2011 at 1:07 PM, Sam Govind govoi...@gmail.com wrote: Hello, I’ve been trying to configure the REFER scenariohttp://www.opensips.org/Resources/B2buaTutorial#toc15but so far no luck. I've successfully called the scenario on initial INVITE and seems like I'm getting B2B responses and replies but when I see sip traces the following two anomalies exist. 1- Refer-To header value is not used anywhere in new INVITE 2- SDP is not sent in new INVITE. Please help. Best Regards, *Sam*** ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Installation of opensips tls version 1.7
Yes, It was compiled from sources. On Mon, Aug 22, 2011 at 4:34 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: ** Hi, in this case, was your opensips compiled from sources or still RPMs ? Regards, Bogdan On 08/20/2011 12:44 PM, isshed wrote: Hello Bogdan, Thanks for your reply. My installation was through after doing the following. ln -s ./libmysqlclient.so.16 ./libmysqlclient.so But when I run it is not compinng up. Bellow is the error message on console. INFO: Starting OpenSIPS : /usr/local/sbin/opensipsctl: line 1478: 29359 Segmentation fault (core dumped) $OSIPSBIN -P $PID_FILE $STARTOPTIONS /dev/null 2 /dev/null ERROR: PID file /var/run/opensips/opensips.pid does not exist -- OpenSIPS start failed Could you please assist me setting it up. Thanks Isshed On Fri, Aug 19, 2011 at 8:03 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: Hi, instead of manually downloading the RPM, better use package manager on your OS. If it is debian derived, use apt, if redhat derived use yum . Regards, Bogdan On 08/19/2011 07:38 AM, isshed wrote: Hello Vlad, Can you please let me know from where can i download the rpms. I am using mysql version 5.0-45-6.fc8. Thanks, On Thu, Aug 18, 2011 at 4:57 PM, isshed isshed@gmail.com wrote: no I did not install it. i could not find the correct rpm.. mysqlclient15-devel-5.0.67-1.fc8.remi.i386.rpm is i downloaded from internet but while installing it says warning: mysqlclient15-devel-5.0.67-1.fc8.remi.i386.rpm: Header V4 DSA signature: NOKEY, key ID 00f97f56 error: Failed dependencies: mysqlclient15 = 5.0.67-1.fc8.remi is needed by mysqlclient15-devel-5.0.67-1.fc8.remi.i386 I would be really thankful to you sir. Thanks On Thu, Aug 18, 2011 at 4:40 PM, Vlad Paiu vladp...@opensips.orgwrote: Hello, Do you have libmysqlclient-dev installed on your machine ? Seems like an error related to it. Regards, -- Vlad Paiu OpenSIPS Developer On 08/18/2011 02:08 PM, isshed wrote: Hi All, I am installing opensips-1.7.0-beta-tls. make prefix=/usr/local all the problem i am facing is .that it is failing while compiling mysql database. it does not find the files mysql.h etc. How to resolve it. Can you please let me know the link of mysql for fedora8. ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS eBootcamp - 19th of September 2011 OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
