Re: [OpenSIPS-Users] parse_uri error
2009/7/13 Om Bikram Thapa om.th...@gmail.com: sip:x...@a.b.c.203:16088:16088 This is obviously a wrong URI (duplicated port?). *Most* probably due to the existance of a fuck*** router with SIP ALG enabled: http://www.voip-info.org/wiki/view/Routers+SIP+ALG -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy
On 11 Jul 2009, at 20:29, Jeff Pyle wrote: The only difference I can see between an inbound call and an outbound call from a media perspective is that in inbound has no pre-connect media (180 w/o SDP) while an outbound call has media (183 w/ SDP). MIght that be relevant? - Jeff The 183 may very well be relevant in this case, but I don't see a 183 in the trace you sent. As well as the SDP logs, please also post the Mediaproxy relay log for the relevant session. Ruud Klaver AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parse_uri error
Thanks Iñaki for pointing to the issue. Regards, Om. On Mon, Jul 13, 2009 at 1:56 PM, Iñaki Baz Castilloi...@aliax.net wrote: 2009/7/13 Om Bikram Thapa om.th...@gmail.com: sip:x...@a.b.c.203:16088:16088 This is obviously a wrong URI (duplicated port?). *Most* probably due to the existance of a fuck*** router with SIP ALG enabled: http://www.voip-info.org/wiki/view/Routers+SIP+ALG -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips Monitoring Current connections and Concurrent calls
Hi All, I would appreciate if someone let me know what tools are available out there to monitor Opensips current connections, concurrent calls and to create snmptraps (alerts) when calls are dropped below/above a min/max threshold I have found out some information using smnpstats module but not quite clear. Any help its much appreciated. Thanks in advance Alberto ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips Monitoring Current connections and Concurrent calls
Well, snmpstats *is* the answer to all of your questions. What's not clear? Hugo Serna wrote: Hi All, I would appreciate if someone let me know what tools are available out there to monitor Opensips current connections, concurrent calls and to create snmptraps (alerts) when calls are dropped below/above a min/max threshold I have found out some information using smnpstats module but not quite clear. Any help its much appreciated. Thanks in advance Alberto ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Trouble with perl module
Hello, I ve installed perl module. Also, i copied directory with perl lib OpenSIPS.pm to /usr/lib/perl. But when i am trying to execute a simple code: use OpenSIPS::Message; print Test\n; or some of examples, line functions.pl, i have error: Can't locate object method bootstrap via package OpenSIPS at /usr/lib/perl/5.10/OpenSIPS/Message.pm line 32. How can i fix it? -- Best regards, Maksim. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRTool - Rating engine does not rate
I did not see an error in the import. But I also solved the problem adding a profile for grn_premium. Thanks, Alberto - Original Message - From: bay2x1 r...@racequeen.ph To: users@lists.opensips.org Sent: Monday, July 13, 2009 2:05 AM Subject: Re: [OpenSIPS-Users] CDRTool - Rating engine does not rate I think the problem is in your profiles table, check if you have a profile entry/record for grn_premium . If you encounter an error while importing the sample data for CDRTool (importRatingTables.php) we might have the same problem. I resolve this issue by inserting a record for grn_premium using the CDRTool web application (profiles section). ASHWINI NAIDU wrote: Have you populated all the relavent data needed for CDRTool rating Destinations table,customers, prfiles and rates. if yes. Delete the entries in Your memcache table and restart ur rating engine and try to get search again. I guess this should work. On Sat, Jul 11, 2009 at 3:21 AM, Alberto Listas lis...@b2br.net wrote: Hi, I have installed OpenSips 1.5.1, FreeRadius 2.1.6, CDRTool 6.8.0. The calls work but they all appear as free in CDRTool. I am using the standard customer and rate tables that come with the software. When I telnet to the rating engine and use the example in the documentation I get the result below: ShowPrice From=sip:1...@example.com sip%3a...@example.com To= sip:0031650222...@example.com sip%3a0031650222...@example.com Gateway=10.0.0.1 Duration=59 0 Duration: 59 s App: audio Destination: 31650 Customer: domain=example.com (And nothing else...) I don't get a price and the /var/log/syslog displays this error: Jul 10 18:28:49 os2 cdrtool[11392]: Error: Cannot find rates for callid=, billing party=...@example.com, customer domain=example.com, gateway=10.0.0.1, destination=31650, profile=grn_premium, app=audio I don't see the error. There is an entry in customers for the domain example.com, there is a rate for destination 31650. But the rating engine does find any. Anybody has a suggestion??? Thanks, Alberto ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users - http://opensips.blogspot.com http://opensips.blogspot.com -- View this message in context: http://n2.nabble.com/CDRTool---Rating-engine-does-not-rate-tp3241434p3248819.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.375 / Virus Database: 270.13.6/2221 - Release Date: 07/06/09 17:54:00 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips+asterisk call dropping in 20 seconds
Hi Ram, a call drop (at signalling level) after 20 secs typically shows a problem with the ACK - the ACK does not get back to callee and callee bye's the call as it never think the call was not confirm. Could you post a trace for the KO call ? Regards, Bogdan ram wrote: Hi In continuation with the subject when i call intiated from Opensips the call drop in 20seconds but when i register directly from * box i dont see the call drop even for 20-30min of talk any suggestions Ram On Tue, Jun 30, 2009 at 8:35 PM, ram talk2...@gmail.com mailto:talk2...@gmail.com wrote: On Tue, Jun 30, 2009 at 5:20 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Ram, I found your email on the Asterisk mailing list also ;) So, to answer here also: do you get any reply back from Asterisk ? Hi Bogdan thanks for the reply I have made a quick Fix, iam not sure how far its good. Just put coment in secret , in the Asterisk Additional_a2billing_sip.conf. rather doing twise authentication. But i have another problem here with the Dispatcher, dispatcher sending calls round robin, 1 st call to 1st * 2nd call to 2nd * 3 call to 3rd * if 2nd Asterisk fails to respond still Dispatcher module sending calls to 2nd asterisk how can i fix this issue with Dispatcher, if any one of * box not reachable it should detect and send call to 3rd * if 2nd comes back in to network and live, it should send to 2nd * how can i achive this ? Ram ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips Monitoring Current connections and Concurrent calls
Hi, Yes, thanks for the link however i got the following problem: After following up the instructions as described on the link: run: snmpd -f -Dagentx -x tcp:localhost:705 21 | less No log handling enabled - turning on stderr logging registered debug token agentx, 1 ... Turning on AgentX master support. agentx/master: initializing ... /master: initializing... DONE NET-SNMP version 5.5.pre2 followed by running opensipsctl on other window: INFO: Starting OpenSIPS: INFO: Started (pid:22021) (all modparam(snmpstats,.) have been included HOWEVER, i don't see any agentx/master messages at all on the first widow under: NET-SNMP version 5.5.pre2 ... Any ideas please? Thanks in advance - Original Message From: Alex Balashov abalas...@evaristesys.com To: Hugo Serna aguil...@yahoo.com Sent: Monday, July 13, 2009 10:52:35 AM Subject: Re: [OpenSIPS-Users] Opensips Monitoring Current connections and Concurrent calls I found the information here sufficient: http://www.opensips.org/html/docs/modules/1.5.x/snmpstats.html I have done implementations on CentOS, Fedora and Debian. aguila: Thanks Alex for your prompt reply. I don't seem to find proper information of how to implement it. are there any configuration samples of how to implement it on Centos5.x/linux enter.. 5.x? thanks - Original Message From: Alex Balashov abalas...@evaristesys.com To: Hugo Serna aguil...@yahoo.com Cc: users@lists.opensips.org Sent: Monday, July 13, 2009 10:26:57 AM Subject: Re: [OpenSIPS-Users] Opensips Monitoring Current connections and Concurrent calls Well, snmpstats *is* the answer to all of your questions. What's not clear? Hugo Serna wrote: Hi All, I would appreciate if someone let me know what tools are available out there to monitor Opensips current connections, concurrent calls and to create snmptraps (alerts) when calls are dropped below/above a min/max threshold I have found out some information using smnpstats module but not quite clear. Any help its much appreciated. Thanks in advance Alberto ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Trouble with perl module
Hi; The same thing happened to me, But when i tried to call the script through the configuration file, it was okay and executed successfully. On Mon, 2009-07-13 at 13:55 +0400, M C wrote: Hello, I ve installed perl module. Also, i copied directory with perl lib OpenSIPS.pm to /usr/lib/perl. But when i am trying to execute a simple code: use OpenSIPS::Message; print Test\n; or some of examples, line functions.pl, i have error: Can't locate object method bootstrap via package OpenSIPS at /usr/lib/perl/5.10/OpenSIPS/ Message.pm line 32. How can i fix it? -- Best regards, Maksim. ___ 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] LDAP Authentication
Hi Alan, It is not OpenSIPS requiring it, it is how SIP works if you want to do it in a secure way :). But feel free and upload a feature request on the tracker for having dynamic binding. Regards, Bogdan Alan Rubin wrote: Bogdan, My site would actually be smaller than that, but that doesn't really address the argument. Is there basically no way, then, to have a single signon-type environment because OpenSIPS requires so much authentication/registration traffic? Regards, Alan Rubin -Original Message- From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Sent: Friday, 3 July 2009 8:46 PM To: Alan Rubin Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] LDAP Authentication But Alan, you will need to re-bind each time you do an Authentication. So, even on a system with 1000 online subscribers, registering each 30 minutes and making a call each 3 hours, means 1000 * 53 = 53000 binds per day - 36 binds per minute. Regards, Bogdan Alan Rubin wrote: Bogdan, If one request equals one user authentication/registration, then I don't think it would hit 1000 binds per week (small environment). If it has to bind each time a packet is sent, then that is pretty inefficient. Regards, Alan Rubin -Original Message- From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Sent: Thursday, 2 July 2009 12:34 AM To: Alan Rubin Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] LDAP Authentication Hi Alan, Got your point! Theoretically, dynamic ldap binding can be done, but the question is how efficient will be (to bind for each auth)..Think that you may process thousands of requests per second! Wouldn't be more reasonable to import the data into mysql? Regards, Bogdan Alan Rubin wrote: Bogdan, I'm not an LDAP expert either, but I will try to explain the scenario better. As you said, the LDAP bind is static - done once in the beginning and sourced from the ldap.cfg file. Unfortunately, we have a filter on our LDAP server that prevents ordinary users from seeing the password field in the LDAP entry. The way we verify authentication in our environment is by dynamically substituting the LDAP bind DN with the client's uid (and password) and making a simple LDAP query using that uid. If that bind is successful, then we know that the password is correct. It doesn't seem like there is anyway to configure opensips in that manner. The aim, with LDAP, was to have a single-signon environment for our LAN and SIP accounts. This doesn't seem possible, unless you or anyone else on the list has any further suggestions. We could use kerberos/AD authentication from the client if that is a possibility. Regards, Alan Rubin -Original Message- From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Sent: Monday, 29 June 2009 10:13 PM To: Alan Rubin Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] LDAP Authentication Hi Alan, I'm not an LDAP expert to get into details about how ldap should be configured or soWhat I can tell is that the bind is static (only once done at the beginning at that's it)Can you send me a link or something to read more about what this dynamic bind means in LDAP ? Thanks and regards, Bogdan Alan Rubin wrote: Bogdan, Apparently the email administrator had a regex on the SMTP gateway to reject messages with pass (and) word (combined) because of previous users succumbing to phishing exercises. It may work now, but I will continue to check the archives. Oh well. Regarding: Now, going to the actual issue, the problem is related to password - about how the client and server (ldap) are keeping the password - do they both keep it same format (like plain text) ? Regards, Bogdan I think I've figured out the issue, although I don't believe there is a solution. Hopefully you can verify, either way. The bind user in the ldap.cfg file does not have the privilege to retrieve the pass word field from our LDAP directory. The only way our LDAP setup is supposed to work is by binding using the user-to-be-authenticated directly with the LDAP directory server. It is my understanding, and this is where you can verify or correct me, that opensips and the LDAP module can not change the bind user dynamically. Regards, Alan Rubin
Re: [OpenSIPS-Users] Registration and Loose-Route
Hi Nathaniel,
It will return true only if:
1) matches one of the listen IPs from the script
2) matches one of the aliases you define in script
3) matches one of the SIP domains defined via the domain module.
Regards,
Bogdan
Nathaniel L Keeling III wrote:
Bogdan,
Does this mean that the statement uri==myself will only be true when
the domain table has an entry matching the sip server FQDN?
Thanks
Nathaniel
Bogdan-Andrei Iancu wrote:
Hi Nathaniel,
OpenSIPs gives you the 403 as it does not recognize itself in the
Route hdr of the REGISTER. By adding the entry in the domain table,
OpenSIPS will recognize its own SIP domains and it will validate the
request.
about the 401 - this is the normal step during the authentication
process. First the UAC sends a request with no credentials, the proxy
answers with a 401 asking for auth; then the UAC should send a new
request containing the credentials.
Regards,
Bogdan
Nathaniel L Keeling wrote:
If there is no entry in the domain table, the it will error in the
loose_route() function and the error message that I get is 403
Preload Route denied. When I add an entry to the domain table, it
passes the loose_route() function and then error while
authenticating. I have placed an xlog statement within the register
block of the config file and right before the loose_route() function
block is executed. I have included my config file.
thanks
Nathaniel
Eduardo Panciera wrote:
Are you sure that the message are been processed by a register
block of your configuration? can you attach your configuration
file? you can use log function in the differents blocks of your
configuration , in order to clarify your debug.
best regards.
Pancho.
On Mon, Jun 29, 2009 at 9:06 PM, Nathaniel L Keeling
keel...@akan-tech.com mailto:keel...@akan-tech.com wrote:
I am new and need an explanation. I have installed opensips 1.5
with
database support. I am trying to authenticate via the subscriber's
table. Utilizing the sample config file and uncommenting the
areas to
allow authentication via database, I try to register a sip
device. I
have added a user using opensipsctl. When the registration
requests
comes in, it dies in the loose_route() function with the error
403
Preload Route Denied. According to the documentation on the
loose_route() function, if there is no to-tag and there is only on
route
header indicating the localproxy, the function should return
false. It
is returning true. I then added the sip domain to the domain
table and
the error changes to 401Unauthorized. Please explain. I am
including
the SIP message and the debug output.
Jun 29 01:15:03 [15473] DBG:core:parse_msg: SIP Request:
Jun 29 01:15:03 [15473] DBG:core:parse_msg: method: REGISTER
Jun 29 01:15:03 [15473] DBG:core:parse_msg: uri:
sip:kwesi.chicagosip1.akan.us.com
http://kwesi.chicagosip1.akan.us.com/
Jun 29 01:15:03 [15473] DBG:core:parse_msg: version: SIP/2.0
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=2
Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type
232,
branch = z9hG4bK728627284; state=6
Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type
235,
rport = n/a; state=17
Jun 29 01:15:03 [15473] DBG:core:parse_via: end of header reached,
state=5
Jun 29 01:15:03 [15473] DBG:core:parse_headers: via found, flags=2
Jun 29 01:15:03 [15473] DBG:core:parse_headers: this is the
first via
Jun 29 01:15:03 [15473] DBG:core:receive_msg: After parse_msg...
Jun 29 01:15:03 [15473] DBG:core:receive_msg: preparing to run
routing
scripts...
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=100
Jun 29 01:15:03 [15473] DBG:core:parse_to: end of header reached,
state=10
Jun 29 01:15:03 [15473] DBG:core:parse_to: display={},
ruri={sip:3124530...@kwesi.chicagosip1.akan.us.com
mailto:sip%3a3124530...@kwesi.chicagosip1.akan.us.com}
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: To [48];
uri=[sip:3124530...@kwesi.chicagosip1.akan.us.com
mailto:sip%3a3124530...@kwesi.chicagosip1.akan.us.com]
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: to body
[sip:3124530...@kwesi.chicagosip1.akan.us.com
mailto:sip%3a3124530...@kwesi.chicagosip1.akan.us.com
]
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: cseq CSeq:
6493
REGISTER
Jun 29 01:15:03 [15473] DBG:maxfwd:is_maxfwd_present: value = 70
Starting to process request
Jun 29 01:15:03 [15473] DBG:uri:has_totag: no totag
we are about to check for cancel
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=78
Jun 29 01:15:03 [15473] DBG:tm:t_lookup_request: start searching:
hash=15692, isACK=0
Jun 29 01:15:03 [15473] DBG:tm:matching_3261: RFC3261 transaction
Re: [OpenSIPS-Users] How to insert the IP address of user in radius request.
Hi Uwe, We already work on changes (there were some discussion on the tracker and IRC) to solve this issue - in 1.6 there will be a AAA module to allow you to do generic RADIUS requests. Regards, Bogdan Uwe Kastens wrote: Hi, I am facing a similar situation. We need to verify that a REGISTER comes from the same srcip we have configured in our database. I am thinking about doing this by making a select into an AVP and verfying the value of the AVP with the $si. If this is successfull the UA would be saved into the location and/or would be able to make a call. This should be possible with radius_avp as well. Looking at performance I would make the DIGEST Auth 1st and if this is succesfull check the IPs. BR uwe Tung Tran schrieb: Hi Mr. Bogdan We need it for IP authorize besides DIGEST auth, that is not standard anyway but business requirements. We use MSSQL to do DIGEST authorize and we need an extra security layer based on source IP, that is also a request by govements in my contry. So last but not lease, I would like someone can help me how to add this feature as soon ass possible Thank you very much for your help Tung - Original Message - From: Bogdan-Andrei Iancu bog...@voice-system.ro To: Tung Tran tr.t...@gmail.com Cc: users@lists.opensips.org Sent: Friday, June 26, 2009 2:24 AM Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in radius request. Hi Tung, I see the difference - unfortunately there is no way (at the moment) to add custom info to the RADIUS auth header, but it should be an extension that can be done - out of curiosity? why do you need this in the AUTH request, as this info is not used in the DIGEST auth. Regards, Bogdan Tung Tran wrote: Dear Mr. Bogdan, I know we can insert the source IP address in account request before sending it to Radius, however can I insert it in AUTHORIZE request instead? Thank you very much for your reply. Tung - Original Message - From: Bogdan-Andrei Iancu bog...@voice-system.ro To: Tung Tran tr.t...@gmail.com Cc: users@lists.opensips.org Sent: Tuesday, June 23, 2009 6:04 PM Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in radius request. Hi Tung, First of all you should upgrade to 1.5 version (see http://www.opensips.org/Resources/Downloads). For your problem, use extra accounting - you can account whatever extra info you want. See: http://www.opensips.org/html/docs/modules/1.5.x/acc.html#ACC-extra-id To get the source IP, use the $si pseudo-variable (see http://www.opensips.org/Resources/DocsCoreVar15#toc71). Regards, Bogdan Tung Tran wrote: Hi all, I get a request to insert the public IP address of the sip softphone or IP Phone/ATA (end-point) in the Radius request sending to Radius server. I am thinking about to mod the auth_radius module to insert that IP in SIP-URI-User field, likely this one: Original Sip-Uri-User = 985512405 After mod: Sip-Uri-User = 985512...@1.2.3.4 Where 1.2.3.4 is the IP of SIP end-point, not the IP address of Opensips/Opensers servers. But I dont know where I should play with. Any one had done it before or know where we can edit, pls help me. BTW, I am using openser 1.2.2 version. Thanks in advance Tung ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips+dispatcher+asterisk problem
Hi Ram, By default, if failover is enabled, Dispatcher module will try all destinations from the set, until it finds one working. You need to catch the failures in failure_route and to use ds_next_domain|dst() functions to try the next available destinations. Also, by using the pringing option, you can re-enable automatically destinations that came back online. Regards, Bogdan ram wrote: On Tue, Jun 30, 2009 at 5:20 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Ram, I found your email on the Asterisk mailing list also ;) So, to answer here also: do you get any reply back from Asterisk ? Hi Bogdan thanks for the reply I have made a quick Fix, iam not sure how far its good. Just put coment in secret , in the Asterisk Additional_a2billing_sip.conf. rather doing twise authentication. But i have another problem here with the Dispatcher, dispatcher sending calls round robin, 1 st call to 1st * 2nd call to 2nd * 3 call to 3rd * if 2nd Asterisk fails to respond still Dispatcher module sending calls to 2nd asterisk how can i fix this issue with Dispatcher, if any one of * box not reachable it should detect and send call to 3rd * if 2nd comes back in to network and live, it should send to 2nd * how can i achive this ? Ram ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Trouble with perl module
It's been a while since I used the perl module, but I think you can't manually call those scripts from the command line. However you can try to copy/symlink the OpenSIPs perl modules into your @INC path to see if that helps. As Ghaith mentions, you shouldn't have that problem with OpenSIPs itself calls the scripts. -Brett On Mon, Jul 13, 2009 at 4:55 AM, M C maximka1...@gmail.com wrote: Hello, I ve installed perl module. Also, i copied directory with perl lib OpenSIPS.pm to /usr/lib/perl. But when i am trying to execute a simple code: use OpenSIPS::Message; print Test\n; or some of examples, line functions.pl, i have error: Can't locate object method bootstrap via package OpenSIPS at /usr/lib/perl/5.10/OpenSIPS/Message.pm line 32. How can i fix it? -- Best regards, Maksim. ___ 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] How To Ask Questions The Smart Way
There is no question that OpenSIPs is a complicated project and that the beginner level docs are mediocre. However, that's mainly because the documentation is written such that it *assumes* that the reader has a decent knowledge of RFC3261. I think if you really know your SIP.. and you download the source, compile and review the default configs (installed by default), it should make a good amount of sense. Beginner questions at that point should start at How do I do string manipulation, What do I use to rewrite private IPs in my SDP and such, which can *easily* be answered by the module docs. Point is. If you don't know SIP, you're diving into the deep end and you'll end up learning SIP the hard way. By learning OpenSIPs first and you'll probably make a lot of mistakes along the way. For me, the confusion is usually something like, I want to rewrite private IPs so nat traversal works properly and then I have to figure out, which route blocks I need to perform fixup operations in and on which message types. But once again, much of that relates more to RFC compliance. Once you understand how it's supposed to work, then you simply look in the module docs (which, btw, are for the most part fantastic) and you just do what you need. The rest of the complication comes from Architecture related questions. And these are the kinds of things where I don't think anyone in here wants to build it for you, but the community would probably give you opinions on specific questions. Things like how do I use one LCR table for user X and a different one for user Y. Well, look at the tools, there's lots of ways to do that! What isn't documented well, and I think a lot of beginners pick up on is how to make a RFC compliant SIP Proxy. Which frankly is totally outside of the scope of the project; but admittedly so, would help a lot of people get started using OpenSIPs. -Brett On Mon, Jul 13, 2009 at 8:45 AM, Bradley, Todd todd.brad...@polycom.comwrote: Who let in the troll? Anonymous coward! One thing I will say, though, as someone who has tried to learn enough OpenSIPS to use it and then posted to the email list for help: It is more difficult to ask intelligent questions about OpenSIPS than many open source tools out there, just because it's so difficult to get started. The introductory-level documentation is weaker than average. Even the cookbooks are written such that they only make sense if you're already an experienced administrator of the software. Cheers, Todd. -Original Message- From: users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] On Behalf Of li...@grounded.net Sent: Friday, July 10, 2009 5:34 AM To: users Subject: Re: [OpenSIPS-Users] How To Ask Questions The Smart Way Bunch of self important blowhards, this is the only mailing list that acts this way! On Fri, 10 Jul 2009 02:48:05 -0400 (EDT), Alex Balashov wrote: Thank you for posting this. It is something that very, very often needs to be said and bears repeating. This a good read for those who show up on mailing lists without any guidance about how to ask the right questions and then complain that nobody answers their questions as they want. http://www.catb.org/~esr/faqs/smart-questions.html It was also a good read for me. Regards, Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Using snmptats with net-snmp
Hi All,
I have been trying to get snmpstats with net-snmp working..
this is what i've done so far:
1. added to opensips.conf
loadmodule snmpstats.so
loadmodule dialog.so# - smpstats params -
modparam(snmpstats, sipEntityType, registrarServer)
modparam(snmpstats, sipEntityType, proxyServer)
modparam(snmpstats, snmpCommunity, public)
modparam(snmpstats, snmpgetPath, /usr/
modparam(snmpstats, MsgQueueMinorThreshold, 2000)
modparam(snmpstats, MsgQueueMajorThreshold, 5000)
modparam(snmpstats, dlg_minor_threshold, 500)
modparam(snmpstats, dlg_major_threshold, 750)
it compiled ok. no errors.
2. have installed net-snmp with agentx and have created under
/usr/local/share/snmp/ two files:
a. snmpd.conf (copied from EXAMPLE.conf). master agentx was added to the file
b. created smnpstats.conf which is an exact copy of the above snmpd.conf.
The new snmpstats.conf contains agentXSocket tcp:localhost:705
Now when i run:
snmpd -f -Dagentx -x tcp:localhost:705 21 | less (to test it)
I get:
No log handling enabled - turning on stderr logging
registered debug token agentx, 1
agentx_register_app_config_handler: registering .conf token for agentxsocket
agentx_register_app_config_handler: registering .conf token for agentxperms
agentx_register_app_config_handler: registering .conf token for agentxRetries
agentx_register_app_config_handler: registering .conf token for agentxTimeout
Turning on AgentX master support.
agentx/master: initializing...
agentx/master: initializing... DONE
NET-SNMP version 5.5.pre3
I then run on a another window opensipsctl start
INFO: Starting OpenSIPS :
INFO: started (pid: 2816)
and starts fine but on the snmpd window I don't get what its suppose to get
according to :http://www.opensips.org/html/docs/modules/1.5.x/snmpstats.html
which is: I qoute:
Now, start up OpenSIPS in another window. In the snmpd window, you should see a
bunch of:
agentx/master: handle pdu (req=0x2c58ebd4,trans=0x0,sess=0x0)
agentx/master: open 0x81137c0
agentx/master: opened 0x814bbe0 = 6 with flags = a0
agentx/master: send response, stat 0 (req=0x2c58ebd4,trans=0x0,sess=0x0)
agentx_build: packet built okay
//
I don't get that at all.
i also run:
snmpget -v2c -c public 192.168.1.109 openserSIPEntityType.0
and i get:
openserSIPEntityType.0: Unknown Object Identifier (Sub-id not found: (top) -
openserSIPEntityType)
Can someone kindly tell what I am doing wrong here please?
Am i missing any statement in the snmpd.conf or snmpstats.conf?
Below its is a copy of the snmpd.conf which is the same as snmpstats.conf
- start file -
/usr/local/share/snmp/snmpd.conf---###
#
# EXAMPLE.conf:
# An example configuration file for configuring the Net-SNMP agent ('snmpd')
# See the 'snmpd.conf(5)' man page for details
#
# Some entries are deliberately commented out, and will need to be explicitly
activated
#
###
#
# AGENT BEHAVIOUR
#
# Listen for connections from the local system only#agentAddress
udp:127.0.0.1:163# Listen for connections on all interfaces (both IPv4 *and*
IPv6)
agentAddress udp:161#agentAddress udp:161,udp6:[::1]:161
###
#
# SNMPv3 AUTHENTICATION
#
# Note that these particular settings don't actually belong here.
# They should be copied to the file /var/net-snmp/snmpd.conf
# and the passwords changed, before being uncommented in that file *only*.
# Then restart the agent
# createUser authOnlyUser MD5 remember to change this password
# createUser authPrivUser SHA remember to change this one too DES
# createUser internalUser MD5 this is only ever used internally, but still
change the password
# If you also change the usernames (which might be sensible),
# then remember to update the other occurances in this example config file to
match.
###
#
# ACCESS CONTROL
#
# system + hrSystem groups only
view systemonly included .1.3.6.1.2.1.1
view systemonly included .1.3.6.1.2.1.25.1
# Full access from the local host
#rocommunity public localhost
# Default access to basic system info
rocommunity public default -V systemonly
# Full access from an example network
# Adjust this network address to match your local
# settings, change the community string,
# and check the 'agentAddress' setting above
#rocommunity secret 10.0.0.0/16
# Full read-only access for SNMPv3
rouser authOnlyUser
# Full write access for encrypted requests
# Remember to activate the 'createUser' lines above
#rwuser authPrivUser priv
# It's no longer typically necessary to use the full 'com2sec/group/access'
configuration
# r[ou]user and r[ow]community, together with suitable views, should cover most
requirements
Re: [OpenSIPS-Users] drouting is_from_gw prepending with pri_prefix
Hi Brett, Thank you for taking the time to respond. The do_routing function adds the prefix of the pri_prefix function for the gateway the call is destined to. I would like to add the prefix based on the gateway the call is from. The function is_from_gw is able to strip based on the originating gateway. I would instead like to prepend / add a prefix - my question pertained to adding a prefix based on the from gateway. So suppose I have: gwidtypeaddress strip pri_prefix attrs description 1 10 X.X.X.X 0 NULLInternal gateway 2 20 1.2.3.4 0 101 NULLExternal gateway 1 3 20 5.6.7.8 0 102 NULLExternal gateway 2 If a call comes from 1.2.3.4 I would like to prepend 101 before sending the call to the Internal gateway. If on the other hand the call comes from 5.6.7.8 I would like to prepend 102 before sending the call to the Internal gateway. I hope that makes sense. /Lasse On 13 Jul 2009, at 16:25, Brett Nemeroff wrote: That should be done automatically by the do_routing function, unless of course you are writing the output of that function to an AVP instead of a RURI. Where it'll end up in the AVP. -Brett On Sun, Jul 12, 2009 at 7:40 PM, Lasse Johnsen la...@freebsdcluster.org wrote: Hi, I use the is_from_gw function in drouting to authenticate what gateways are allowed to talk to my OpenSIPS server. I understand that I can strip something from the R-URI using this function. Is there anyway I can add the gateway db entry's pri_prefix in the same fashion. I essentially want to prepend the r-uri based on the originating gateway. What is the best way of doing this. /Lasse ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Best regards, Lasse L. Johnsen CEO, VoIP-X Communications Limited E lasse.john...@voip-x.net M +44 7832 335 392 F +44 2075 262 178 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] drouting is_from_gw prepending with pri_prefix
Well there are a lot of ways to do this. First you'll need to do *something* to convert the IP to an id of sorts. I convert IP to a trunk ID using the memcache module + avpops. Then you can pass that ID into do_routing to get a specific route/gateway with the right pri prefix. The other way to do it is the dialplan module. You'll also need to pass it some sort of ID to filter out which rule to use. Check it out: http://www.opensips.org/html/docs/modules/1.5.x/dialplan.html#id271074 Of course, you can do this literally dozens of other ways too. memcache, static routes, I think even dispatcher has some methods you can use.. -Brett On Mon, Jul 13, 2009 at 10:47 AM, Lasse Johnsen la...@freebsdcluster.orgwrote: Hi Brett, Thank you for taking the time to respond. The do_routing function adds the prefix of the pri_prefix function for the gateway the call is destined to. I would like to add the prefix based on the gateway the call is from. The function is_from_gw is able to strip based on the originating gateway. I would instead like to prepend / add a prefix - my question pertained to adding a prefix based on the from gateway. So suppose I have: gwidtypeaddress strip pri_prefix attrs description 1 10 X.X.X.X 0 NULLInternal gateway 2 20 1.2.3.4 0 101 NULLExternal gateway 1 3 20 5.6.7.8 0 102 NULLExternal gateway 2 If a call comes from 1.2.3.4 I would like to prepend 101 before sending the call to the Internal gateway. If on the other hand the call comes from 5.6.7.8 I would like to prepend 102 before sending the call to the Internal gateway. I hope that makes sense. /Lasse On 13 Jul 2009, at 16:25, Brett Nemeroff wrote: That should be done automatically by the do_routing function, unless of course you are writing the output of that function to an AVP instead of a RURI. Where it'll end up in the AVP. -Brett On Sun, Jul 12, 2009 at 7:40 PM, Lasse Johnsen la...@freebsdcluster.org wrote: Hi, I use the is_from_gw function in drouting to authenticate what gateways are allowed to talk to my OpenSIPS server. I understand that I can strip something from the R-URI using this function. Is there anyway I can add the gateway db entry's pri_prefix in the same fashion. I essentially want to prepend the r-uri based on the originating gateway. What is the best way of doing this. /Lasse ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Best regards, Lasse L. Johnsen CEO, VoIP-X Communications Limited E lasse.john...@voip-x.net M +44 7832 335 392 F +44 2075 262 178 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips+asterisk call dropping in 20 seconds
On Mon, Jul 13, 2009 at 6:30 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Ram, a call drop (at signalling level) after 20 secs typically shows a problem with the ACK - the ACK does not get back to callee and callee bye's the call as it never think the call was not confirm. Could you post a trace for the KO call ? Hi Bogdan thanks i have fix the problem it was some config issue Ram ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
