Re: [OpenSIPS-Users] CDRtools Billing missed calls..
On Thursday 30 July 2009, ram wrote: Hi I have setup rates in my table.. 0/0 for the profile 24hours basis and defined subscriber to use that profile to make rating for the outbound calls. when the Opensips subscriber calls to PSTN Number 001732XX and wait for 2 or 3 rings and hangup the call. still i see the CDRtools billing with rate. The call was answered with a 200 OK, then ended with a BYE. Why exactly don't you expect to see it billed? *Signalling information* http://cdrtool.sbttalk.net/CDRTool/callsearch.phtml?cdr_source=opensips_radiuscdr_table=radius.radacct200907order_by=RadAcctIdorder_type=DESCbegin_datetime=1248904920end_datetime=1248990900maxrowsperpage=15action=searchcall_id=24271317073689-149641495610936%40202.63.111.2 Call id: 24271317073689-149641495610...@x.x.x.2 From/to tags: 2290420994/as2a1521b8 Start time: 2009-07-30 02:06:55 Stop time: 2009-07-30 02:07:09 Method: Invite from ip-of-voipphone*:5060* From: u...@domain.net Domain: domain.net To (dialed URI): 001732...@freeswitch.sbttalk.net Canonical URI: 001732...@freeswitch.sbttalk.net Next hop URI: 001732...@202.63.96.31 Destination: USA (1732) Billing Party: u...@domain.net Reseller: 0 *Rating information* Duration: 14 s App: audio Destination: 1732 Customer: subscriber=u...@domain.net Connect: 0. StartTime: 2009-07-30 02:06:55 -- Span: 1 Duration: 14 s ProfileId: sl_standard / weekday RateId: sl_standard / 0-24h Rate: 0.0009 / 60 s Price: 0.0002 Price in: 0.0002 -- Price out: 0.0002 Price in: 0.0002 Margin: 0. here is my siptrace SIP trace on proxy cdrtool.domain for session 24271317073689-149641495610...@voipphone-ip -- Packet 1 at from Opensip-IP to voipphone-ip (out) SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP voipphone-ip:5060;branch=z9hG4bK28385192501472111761;rport=5060 From: user sip:u...@domain.net:5060;tag=2290420994 To: 001732XX sip:001732xxx...@domain.net:5060 ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.6b91 Call-ID: 24271317073689-149641495610...@voipphone-ip CSeq: 1 INVITE Proxy-Authenticate: Digest realm=domain.net, nonce=4a7162cd01459588519a6132ccee82d5638acaecdff8 Server: OpenSIPS (1.5.1-notls (i386/linux)) Content-Length: 0 Warning: 392 Opensip-IP:5060 Noisy feedback tells: pid=17765 req_src_ip=voipphone-ip req_src_port=5060 in_uri= sip:001732xxx...@domain.net:5060 out_uri=sip:001732xxx...@domain.net:5060via_cnt==1 --- Packet 2 at from Opensip-IP to Opensip-IP (out) INVITE sip:001732xxx...@opensip-ip:5062 SIP/2.0 Record-Route: sip:Opensip-IP;lr=on;did=ee6.8459b117 Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKe754.5dacff85.0 Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK938015010138926320;rport=5060 From: user sip:u...@domain.net:5060;tag=2290420994 To: 001732XX sip:001732xxx...@domain.net:5060 Call-ID: 24271317073689-149641495610...@voipphone-ip CSeq: 2 INVITE Contact: sip:u...@voipphone-ip:5060 Max-Forwards: 69 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 319 P-hint: inbound-inbound v=0 o=4720779942 28362303 19011140 IN IP4 voipphone-ip s=A conversation c=IN IP4 voipphone-ip t=0 0 m=audio 10158 RTP/AVP 18 4 8 0 9 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- Packet 3 at from Opensip-IP to Opensip-IP (in) SIP/2.0 100 Trying Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKe754.5dacff85.0;received=Opensip-IP Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK938015010138926320;rport=5060 Record-Route: sip:Opensip-IP;lr=on;did=ee6.8459b117 From: user sip:u...@domain.net:5060;tag=2290420994 To: 001732XX sip:001732xxx...@domain.net:5060 Call-ID: 24271317073689-149641495610...@voipphone-ip CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:001732xxx...@opensip-ip:5062 Content-Length: 0 --- Packet 4 at from Opensip-IP to Opensip-IP (in) SIP/2.0 200 OK Via: SIP/2.0/UDP Opensip-IP;branch=z9hG4bKe754.5dacff85.0;received=Opensip-IP Via: SIP/2.0/UDP voipphone-ip:5060;received=voipphone- ip;branch=z9hG4bK938015010138926320;rport=5060 Record-Route: sip:Opensip-IP;lr=on;did=ee6.8459b117 From: user sip:u...@domain.net:5060;tag=2290420994 To: 001732XX sip:001732xxx...@domain.net:5060;tag=as2a1521b8 Call-ID: 24271317073689-149641495610...@voipphone-ip CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:001732xxx...@opensip-ip:5062 Content-Type:
Re: [OpenSIPS-Users] [OpenSIPS-Devel] [NEW] Virtual DB module
On Thursday 30 July 2009, Razvan Pistolea wrote: Why a virtual db module? Until now all modules requiring db support had only one (real) db_url connection and it was bad if that connection went down. Here is where virtual db module comes in: it provides to other modules a (virtual) db_url that maps multiple (real) db_urls to real dbs. The real connections are handled transparently to the module using the virtual url. Now if the current real connection goes down the virtual db_url will just switch to a new one transparently to the module, giving uninterrupted db access. This mode is called failover and can be used for write and read operations. How does this work with operations that are separate, but still represent a single logical operation, like for example writing usrloc, or dialog data into the database not in real time but on a timer, where multiple records are inserted at a time. If a connection fails in the middle of an operation, some records will end up in one database and some in another and OpenSIPS will have troubles finding the information later. Without having transaction support for such operations, so that all the inserts that belong together fail and are retried together on the next connection, it would be problematic. Another issue is that even if transaction support would be implemented, there is still an uncertainty where the data is. If my usrloc or dialog data was saved over multiple connections, after a restart, from where is OpenSIPS supposed to read the data? I can see that this can work fine for read operations assuming that the databases are synchronized by external means and wherever you go you get the same information, but for write I do not see where the aggregation layer is to assure that the data is synchronized and consistent. -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS-Devel] [Fwd: Re: [NEW] Virtual DB module]
On Thursday 30 July 2009, Razvan Pistolea wrote: 2. Synchronization between dbs will be lost. 3b.use database managers (that know how to merge databases) 3c.use a cluster db That sounds good in theory, but in practice fails in so many ways. I've worked with this for years and even after so much time bidirectional database replication is extremely fragile and fails easily. Just one example: You start writing to db1, it performs the operation just fine, but right after it finished and is about to return you the success response to your query you lose the connection. If never see the answer, assume it failed and go on to write the record to db2, which succeeds as well and also returns the answer. Now you have the same record in both databases and when they will try to replicate from each other they'll fail. The problem gets even worse with multiple databases that replicate from each other. This is not just a theoretical example. I've seen this on a constant basis when performing a simple operation like heartbeat stop on the master to move the services to the slave, while somebody writes into the database as the same time (like for example opensips writing accounting requests for something as modest as 1.5 calls per second). This is so common, that you can consider yourself lucky if replication is not broken between the 2 databases when you stop one to activate the other, without taking measures to stop the influx of write operations to them during the switch. -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRtools Billing missed calls..
The call was answered with a 200 OK, then ended with a BYE. Why exactly don't you expect to see it billed? Hi thanks Dan the culprit was * box, with the dialplan answer, so its sending the 200K i have replaced with Rininging. i can see the calls are canceled thanks for your reply Ram ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS-Devel] [Fwd: Re: [NEW] Virtual DB module]
2009/7/31 Dan Pascu d...@ag-projects.com: On Thursday 30 July 2009, Razvan Pistolea wrote: 2. Synchronization between dbs will be lost. 3b. use database managers (that know how to merge databases) 3c. use a cluster db That sounds good in theory, but in practice fails in so many ways. I've worked with this for years and even after so much time bidirectional database replication is extremely fragile and fails easily. Just one example: You start writing to db1, it performs the operation just fine, but right after it finished and is about to return you the success response to your query you lose the connection. If never see the answer, assume it failed and go on to write the record to db2, which succeeds as well and also returns the answer. Now you have the same record in both databases and when they will try to replicate from each other they'll fail. The problem gets even worse with multiple databases that replicate from each other. This is not just a theoretical example. I've seen this on a constant basis when performing a simple operation like heartbeat stop on the master to move the services to the slave, while somebody writes into the database as the same time (like for example opensips writing accounting requests for something as modest as 1.5 calls per second). This is so common, that you can consider yourself lucky if replication is not broken between the 2 databases when you stop one to activate the other, without taking measures to stop the influx of write operations to them during the switch. Very good points. master-master replication or master-slave becoming inactive-master is really a pain, I've also suffered it a lot. I wonder if filesystem based replication (i.e. BRBD) is a more reliable choice even if it could seem fragile (it's a binary replication, so a single error could entirely corrupt the database). About it, I've listened every kind of opinions, so... :) -- 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] [OpenSIPS-Devel] Re-invite ACK Troubles - Not going to my NAT Handling route
Thank you Iñaki, it's this! I have put the t_on_branch(1) at the beginning of the main route and now, it works! Have a nice day! -- -- -- Marc LEURENT lf...@leurent.eu Le vendredi, 31 juillet 2009 10.52:30, Iñaki Baz Castillo a écrit : 2009/7/31 Marc Leurent marc.leur...@vtx-telecom.ch: Here is the rest of the opensip.cfg, I set t_on_branch(1); t_on_reply(1); t_on_failure(1); later The loose-routing section (that which handles in-dialog requests as re-INVITE) uses route[1]. I suspect that in route[1] you don't set the branch_route[1] as it appears at the end of your config file, just for initial requests. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] xcap_client module doesn't implement pidf-manipulation, right?
Hi, by inspecting the code of xcap_client.c it seems that
pidf-manipulation (RFC 4827) is not implemented:
int get_auid_flag(str auid)
{
static str pres_rules = str_init(pres-rules);
static str rls_services = str_init(rls-services);
Am I right?
--
Iñaki Baz Castillo
i...@aliax.net
___
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Re: [OpenSIPS-Users] BT IP Exchange interconnect
Yes - and in record time. :) On Jul 31, 2009, at 3:31 AM, Gavin Henry wrote: Hi All, Has anyone passed the tests using OpenSIPS: http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html Thanks. ___ 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-Devel] [NEW] Virtual DB module
Hi Dan Hi Thomas Thank you for your interest in my module and associated db dilemmas shortcomings. As you have pointed out: - there is not much that can be done NOW without db transaction support. - db virtual module is a simple (hopefully helpful) wrapper. Best regards, Razvan Pistolea --- On Fri, 7/31/09, Thomas Gelf tho...@gelf.net wrote: From: Thomas Gelf tho...@gelf.net Subject: Re: [OpenSIPS-Devel] [NEW] Virtual DB module To: de...@lists.opensips.org Cc: users@lists.opensips.org Date: Friday, July 31, 2009, 4:04 AM Dan Pascu wrote: On Thursday 30 July 2009, Razvan Pistolea wrote: How does this work with operations that are separate, but still represent a single logical operation, like for example writing usrloc, or dialog data into the database not in real time but on a timer, where multiple records are inserted at a time. If a connection fails in the middle of an operation, some records will end up in one database and some in another and OpenSIPS will have troubles finding the information later. Without having transaction support for such operations, so that all the inserts that belong together fail and are retried together on the next connection, it would be problematic. I agree that transaction support would be not only a good idea, I consider it really important - and probably not that hard to add (ok, this depends strongly on how all these backends ar abstracted - it could also be really tricky...). Another issue is that even if transaction support would be implemented, there is still an uncertainty where the data is. If my usrloc or dialog data was saved over multiple connections, after a restart, from where is OpenSIPS supposed to read the data? From the active one - of course this requires your databases to be somehow perfectly synchronised. I can see that this can work fine for read operations assuming that the databases are synchronized by external means and wherever you go you get the same information, but for write I do not see where the aggregation layer is to assure that the data is synchronized and consistent. That's not OpenSIPS job - it's up to who is configuring it to take care of synced DBs and such things. OpenSIPS has to drop it's queries somewhere - nothing more. And multiple configured DBs allows you to easily take down one DB server for maintainance without interrupting the service. I consider this a very worthful addition. And there is not so much trouble you can cause if OpenSIPS is either not inserting a record or writing it twice to usrloc - after some time everything will be fine again. The impact is much less than a real downtime. Just my 2 eurocent - please correct me, if I'm wrong! Best regards, Thomas Gelf ___ Devel mailing list de...@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] New MediaProxy release 2.3.6
Hello, There is a new release of MediaProxy available, it contains bug fixes and better compatibility with and checks for the software it depends upon. To upgrade your debian unstable installation: sudo apt-get update sudo apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy- web-sessions Or download the tar file from: http://download.ag-projects.com/MediaProxy/ The changelog follows: mediaproxy (2.3.6) unstable; urgency=low * Consider offered streams with a port of 0 to be removed * Don't remove hold timeout when the conntrack rule expires while on hold * Added synthetic test for on_hold_timeout (holdtest3.py) * Use learnt remote IP when ssending packets through userspace if possible * Adapted code to the latest API changes in python-application 1.1.5 * Removed old and now redundant test for the twisted version from relay.py * Refactored some datatypes from validators to actual types * Modified SIPThorDomain data validator to accept everything it generates * Allow the dispatcher list in the config file to be comma separated as well * Updated minimum version dependency for python-application * Fix for send_packet_count attribute not being set Kind regards, Adrian Georgescu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] New CallControl release 2.0.6
Hello, There is a new release of CallControl prepaid application available, it contains bug fixes and better compatibility with and checks for the software it depends upon. To upgrade your debian unstable installation: apt-get update apt-get install callcontrol Or download the tar file from: http://download.ag-projects.com/CallControl/ Changelog callcontrol (2.0.6) unstable; urgency=low * Replaced python-all-dev build dependency with python * Replaced use of deprecated __configfile__ with __cfgfile__ * Added dependency checking using application.dependency * Replaced use of deprecated mode argument of listenUNIX with chmod * Modified config.ini.sample to reflect the need for doubling percent signs callcontrol (2.0.5) unstable; urgency=low * Replaced use of deprecated _datatypes attribute with ConfigSetting * Replaced use of print statement with log.error * Fixed handling of debug option of daemon * Fixed use of process.system_config_directory * Replaced use of read_settings with ConfigSection attributes * Updated dependency on python-application to = 1.1.2 Regards, Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] CDRTool - Prepaid does not decrement balance
Hi, I am having a difficulty with prepaid in CDRTool. When I telnet to the server and give the commands manually as below the prepaid balance is decremented correctly. When I place the call thru Opensips the call is billed (as show below too) but the prepaid balance is not decremented. I did not install Call Control yet because it's not very important that I stop calls on the fly. Could that be the problem? Thanks in advance for any suggestions, Alberto _ os1:~# telnet os2 9024 Trying XXX.XXX.197.172... Connected to os2.voip.net. Escape character is '^]'. MaxSessionTimecallid=6432622...@1 From=sip:2...@os1.voip.net To=sip:005521850222...@xxx.xxx.197.171 Duration=7200 Gateway=XXX.XXX.188.229 Lock=1 402 ShowPrice From=sip:2...@os1.voip.net To=sip:005521850222...@xxx.xxx.197.171 Gateway=XXX.XXX.188.229 Duration=59 0.1300 Duration: 59 s App: audio Destination: 55218 Customer: subscriber=2...@os1.voip.net Connect: 0. StartTime: 2009-07-31 17:06:51 -- Span: 1 Duration: 60 s Increment: 6 Min duration: 30 ProfileId: plu / weekday RateId: plu / 0-24h Rate: 0.1300 / 60 s Price: 0.1300 DebitBalance callid=6432622...@1 From=sip:2...@os1.voip.net To=sip:005521850222...@xxx.xxx.197.171 Gateway=XXX.XXX.188.229 Duration=59 Ok MaxSessionTime=0 0.1300 Duration: 59 s App: audio Destination: 55218 Customer: subscriber=2...@os1.voip.net Connect: 0. StartTime: 2009-07-31 17:08:15 -- Span: 1 Duration: 60 s Increment: 6 Min duration: 30 ProfileId: plu / weekday RateId: plu / 0-24h Rate: 0.1300 / 60 s Price: 0.1300 ___ Id Start time Sip Proxy SIP caller SIP destination Dur Price KBIn KBOut Status Codecs 1N 2009-07-31 11:27:21 XXX.XXX.197.171 2...@os1.voip.net +552185022233 (BRAZIL CELL 55218) 00:04 0.0650 Ok (200) Signalling information Click here to show only this call id Call id: 1537464575-5264...@189.4.254.119 Click here to see the SIP trace for this call From/to tags: 1685903383/013be01d Start time: 2009-07-31 11:27:21 Stop time: 2009-07-31 11:27:25 Method: Invite from XXX.XXX.254.119:5264 From: 2...@os1.voip.net Domain: os1.voip.net To (dialed URI): 005521850222...@os1.voip.net Canonical URI: 005521850222...@os1.voip.net Next hop URI: 005521850222...@xxx.xxx.195.56 Destination: BRAZIL CELL (55218) Billing Party: 2...@os1.voip.net Reseller: 0 Rating information Duration: 4 s App: audio Destination: 55218 Customer: subscriber=2...@os1.voip.net Connect: 0. StartTime: 2009-07-31 16:27:21 -- Span: 1 Duration: 30 s Increment: 6 Min duration: 30 ProfileId: plu / weekday RateId: plu / 0-24h Rate: 0.1300 / 60 s Price: 0.0650 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Just as a proxy server
Just don't include the auth bits. In fact, I think in the examples the auth bits are commented out. Have you tried it? On Fri, Jul 31, 2009 at 11:02 AM, Ghaith ALKAYYEM ghaith.alkay...@telecom-bretagne.eu wrote: Hello, Does anybody know how we can cancel the authentication functionality in OpenSIPS and make it run as a proxy server router. Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRTool - Prepaid does not decrement balance
On Jul 31, 2009, at 5:33 PM, Alberto Listas wrote: Hi, I am having a difficulty with prepaid in CDRTool. When I telnet to the server and give the commands manually as below the prepaid balance is decremented correctly. When I place the call thru Opensips the call is billed (as show below too) but the prepaid balance is not decremented. I did not install Call Control yet because it's not very important that I stop calls on the fly. Could that be the problem? Exactly. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS-Devel] [NEW] Virtual DB module
On Friday 31 July 2009, Razvan Pistolea wrote: Hi Dan Hi Thomas Thank you for your interest in my module and associated db dilemmas shortcomings. As you have pointed out: - there is not much that can be done NOW without db transaction support. - db virtual module is a simple (hopefully helpful) wrapper. It can certainly be very useful for read operations, assuming the databases are synchronized. Write operations are the ones that are problematic and it's not something I expect such a module to solve by itself because the solution doesn't belong to this realm. Even with transaction support there are still issues that are not solved when writing. Distributed databases are a hard problem. My only intention was to highlight the potential issues so that there are no illusions about what can be done right now, even with this new module. -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Just as a proxy server
El Viernes, 31 de Julio de 2009, Ghaith ALKAYYEM escribió: Hello, Does anybody know how we can cancel the authentication functionality in OpenSIPS and make it run as a proxy server router. Strange question, really... OpenSIPS is not a black-box with some preconfigured features. The example config script is just that, an *example*. You decide what to enable or disable by creating your own config file and reading the documentation for the needed modules. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Scan all Contact adresses
Hi, Is there a way to display the hostport of all name-addr present in the Contact header? Yannick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
