sr-users October 2007

sr-users@lists.kamailio.org

130 participants
285 discussions

[OpenSER-Users] How can I test if openser works well with openxcap?

by KevinKinnan

My testbed is built on Ubuntu 7.04 and OpenIMScore, with openser(including mysql, presence, presence_xml and mi_xmlrpc module) and openxcap. Each component seems working fine according to syslog. But I am wondering how can I exactly know they integrate well? I was running this testbed without openxcap to test presence using the following openser config(config 1). Notify request from openser is ok. However, the notify request disappears after shifting to integration of openser and openxcap(see config 2), and subscription state is 'pending' rather than 'active' as seen when testing without openxcap. CONFIG 1: ============================= loadmodule "presence.so" loadmodule "presence_xml.so" ... # -- presence params -- modparam("presence", "db_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'pres') #modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:148.88.xxx.xxx:5065") ... # -- presence_xml params -- modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "force_active", 1) ============================= CONFIG 2: ============================= loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "mi_xmlrpc.so" modparam("mi_xmlrpc", "log_file", "/var/log/openser-xmlrpc.log") modparam("mi_xmlrpc", "port", 8888) # -- presence params -- modparam("presence", "db_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'pres') #modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:148.88.xxx.xxx:5065") # -- presence_xml params -- modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "pidf_manipulation", 1) modparam("presence_xml", "integrated_xcap_server", 1) ============================== OPENXCAP config.ini: address=0.0.0.0 port=8000 root=http://localhost/xcap-root backend=Database document_valication=Yes type=basic cleartest_passwords=Yes default_realm=open-ims.test trusted_peers= authentication_db_uri=mysql://openser:openserrw@localhost:3306/openser storage_db_uri=mysql://openser:openserrw@localhost:3306/openser subscriber_table=subscriber xcap_table=xcap authentication_db_uri=mysql://openser:openserrw@localhost:3306/openser storage_db_uri=mysql://openser:openserrw@localhost:3306/openser xcap_table=xcap xmlrpc_url=http://localhost:8000 ============================== Anothter thing about apache, I have not create any virtual host just localhost points to /var/www/ (see the value of root in config.ini). According to the mechanism of openxcap, the subscribe request is sent as well as a HTTP GET. But I can't find anything about that http GET in both internal and external traces on port 4060,5060,6060,5065,8000,8888(no xml document in/var/www/xcap-root is created). Is this right? Or, do I need to redesign my sip client by adding a HTTP sending function when subscribing in order to test openxcap? Thanks a billion! Keivn, _________________________________________________________________ 用 Live Search 搜尽天下资讯！ http://www.live.com/?searchOnly=true

16 years, 8 months

[Serusers] Cutomize usr_preferences

by Roderick

Hello list, We are using SER usr_preferences table for handling a large number of preferences for each user, the problem is that if we use the default form of the MySQL table we would have several (around 10) records for each client (and maybe more as we develop new services). As we have a fixed number of preferences I'd prefer to change the table so it wouldn't be: CREATE TABLE `usr_preferences` ( `uuid` varchar(64) NOT NULL default '', `username` varchar(100) NOT NULL default '0', `domain` varchar(128) NOT NULL default '', `attribute` varchar(32) NOT NULL default '', `value` varchar(128) NOT NULL default '', `type` int(11) NOT NULL default '0', `modified` timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, PRIMARY KEY (`attribute`,`username`,`domain`) ) ENGINE=ndbcluster DEFAULT CHARSET=latin1 | But instead it would be something like CREATE TABLE `usr_preferences` ( `uuid` varchar(64) NOT NULL default '', `username` varchar(100) NOT NULL default '0', `domain` varchar(128) NOT NULL default '', `call_forward_on_busy` varchar(32) NOT NULL default '', ` call_forward_on_no_available` varchar(32) NOT NULL default '', ` call_forward_on_no_available` varchar(32) NOT NULL default '', ` call_forward_on_no_response` varchar(32) NOT NULL default '', ` language` varchar(32) NOT NULL default '', `type` int(11) NOT NULL default '0', `modified` timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, PRIMARY KEY (`attribute`,`username`,`domain`) ) ENGINE=ndbcluster DEFAULT CHARSET=latin1 | Can this be done, and if so, how can I handle the attribute-value pairs for each different column???? Thanks for any help you can provide, Roderick Click <http://www.vivophone.com/vivosoft/index.php?idm=Eng&promo=SR0979> here to get your own VIVOphone Account with 30 free minutes to the US and Canada and start calling NOW!!!!

16 years, 8 months

[OpenSER-Users] Single MySQL view for OpenSER and Asterisk for billing

by lu

Hello mates, Is it possible to use A2bill to correctly obtain the CDRs and bill OpenSER subscribers who only use Asterisk as a PSTN GTW in just a single MySQL view without need of replicating them again into another Asterisk MySQL database? WBR, LU. On Tue, 2007-10-23 at 14:25 +0200, users-request(a)openser.org wrote: > Send Users mailing list submissions to > users(a)openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)openser.org > > You can reach the person managing the list at > users-owner(a)openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. "480 User not responding" instead of "408 Request Timeout" > when modparam("tm", "noisy_ctimer", 1)? (I?aki Baz Castillo) > 2. INVITE relayed with missing bytes (Papadopoulos Georgios) > 3. "480 User not responding" instead of "408 Request Timeout" > when modparam("tm", "noisy_ctimer", 1)? (Juha Heinanen) > 4. Re: "480 User not responding" instead of "408 Request > Timeout" when modparam("tm", "noisy_ctimer", 1)? (I?aki Baz Castillo) > 5. Re: case sensitivity with avp_db_load (Jiri Kuthan) > 6. Bug in 200 to CANCEL (wrong To_tag) (I?aki Baz Castillo) > 7. Re: About "q" values (Klaus Darilion) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 23 Oct 2007 12:29:02 +0200 > From: I?aki Baz Castillo <ibc(a)aliax.net> > Subject: [OpenSER-Users] "480 User not responding" instead of "408 > Request Timeout" when modparam("tm", "noisy_ctimer", 1)? > To: users(a)openser.org > Message-ID: <200710231229.02671.ibc(a)aliax.net> > Content-Type: text/plain; charset="utf-8" > > Hi, if modparam("tm", "noisy_ctimer", 1) and INVITE exceded "fr_inv_timer" > then OpenSer sends "408 Request Timeout" to caller and CANCEL to called. > > I'm sure this is correct according to RFC, but wouldn't be better replying > with "480 User not responding"? > > For example, Asterisk does nothing if it receives "408 Request Timeout" (it > ignores it). I've reported about about it: > http://bugs.digium.com/view.php?id=11058 > > Sure it's a fail of Asterisk who shoud accept "408" and terminate the call, > but anyway, wouldn't be correct to reply with "480" instead of "408"? > > Regards. > > > -- > Iaki Baz Castillo > > > > ------------------------------ > > Message: 2 > Date: Tue, 23 Oct 2007 13:36:02 +0300 > From: "Papadopoulos Georgios" <geop(a)altectelecoms.gr> > Subject: [OpenSER-Users] INVITE relayed with missing bytes > To: <users(a)openser.org> > Message-ID: > <9DB9DF2949D8774796CB054FBEC15699077F68F2(a)Tyran.int.acn.gr> > Content-Type: text/plain; charset="us-ascii" > > Hello all, > > I am noticing that OpenSER relays an INVITE and the packet that is sent > out is clipped. You can see in the following packets that the forwarded > packet is missing a few bytes from the SDP part. Counting the bytes > shows that the outgoing packet is always 1512 bytes. Is this something > that has do with OpenSER (module parameter, compile options) or is it OS > related? > > thank you for any help > > George > > > > U 2007/10/23 13:27:46.165232 213.5.1.6:57665 -> 213.5.43.4:5060 > INVITE sip:1012118204501@213.5.43.4:5060 SIP/2.0. > Via: SIP/2.0/UDP > 213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C > 1AB9. > Remote-Party-ID: "GEOP Papadopoul" > <sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off. > From: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821. > To: <sip:1012118204501@213.5.43.4>. > Date: Tue, 23 Oct 2007 10:27:46 GMT. > Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6. > Supported: 100rel,timer,resource-priority,replaces. > Min-SE: 1800. > Cisco-Guid: 1848010793-2156466652-3218142496-4228086245. > User-Agent: Cisco-SIPGateway/IOS-12.x. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, > SUBSCRIBE, NOTIFY, INFO, REGISTER. > CSeq: 101 INVITE. > Max-Forwards: 70. > Timestamp: 1193135266. > Contact: <sip:2116872933@213.5.1.6:5060>. > Call-Info: > <sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" > . > Expires: 180. > Allow-Events: telephone-event. > Content-Type: application/sdp. > Content-Disposition: session;handling=required. > Content-Length: 290. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6. > s=SIP Call. > c=IN IP4 213.5.1.6. > t=0 0. > m=audio 16454 RTP/AVP 18 0 8 100. > c=IN IP4 213.5.1.6. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=yes. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:100 X-NSE/8000. > a=fmtp:100 192-194. > > > U 2007/10/23 13:27:46.173826 213.5.43.4:5060 -> 213.5.1.6:57665 > SIP/2.0 100 Giving a try. > Via: SIP/2.0/UDP > 213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C > 1AB9;rport=57665. > From: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821. > To: <sip:1012118204501@213.5.43.4>. > Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6. > CSeq: 101 INVITE. > Server: Altec Telecoms SIP Proxy. > Content-Length: 0. > . > > > U 2007/10/23 13:27:46.174241 213.5.43.4:5060 -> 213.5.17.226:5060 > INVITE sip:demo1@192.168.1.39:5060 SIP/2.0. > Record-Route: <sip:213.5.43.4;lr=on;ftag=237E8130-1821>. > Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bK4acd.5092baa7.0. > Via: SIP/2.0/UDP > 213.5.1.6:5060;rport=57665;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch= > z9hG4bK1594C1AB9. > Remote-Party-ID: "GEOP Papadopoul" > <sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off. > From: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821. > To: <sip:1012118204501@213.5.43.4>. > Date: Tue, 23 Oct 2007 10:27:46 GMT. > Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6. > Supported: 100rel,timer,resource-priority,replaces. > Min-SE: 1800. > Cisco-Guid: 1848010793-2156466652-3218142496-4228086245. > User-Agent: Cisco-SIPGateway/IOS-12.x. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, > SUBSCRIBE, NOTIFY, INFO, REGISTER. > CSeq: 101 INVITE. > Max-Forwards: 10. > Timestamp: 1193135266. > Contact: <sip:2116872933@213.5.1.6:57665>. > Call-Info: > <sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" > . > Expires: 180. > Allow-Events: telephone-event. > Content-Type: application/sdp. > Content-Disposition: session;handling=required. > Content-Length: 290. > P-hint: NATed client request. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6. > s=SIP Call. > c=IN IP4 213.5.1.6. > t=0 0. > m=audio 16454 RTP/AVP 18 0 8 100. > c=IN IP4 213.5.1.6. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=yes. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:100 X-NSE/8000. > a=fmtp:100 > > > Disclaimer > The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > >

16 years, 8 months

RE: [OpenSER-Users] INVITE relayed with missing bytes

by Papadopoulos Georgios

Hi Ovidiu, So you are saying that despite ngrep does not show it, the whole packet is correctly sent? thank you George > -----Original Message----- > From: Ovidiu Sas [mailto:sip.nslu@gmail.com] > Sent: Tuesday, October 23, 2007 5:23 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] INVITE relayed with missing bytes > > Hi Georgios, > > It seems that your packet is to big and it was fragmented > (udp fragmentation). > OpenSER is able to deal with fragmented packets (both sending > and receiving). > > > Regards, > Ovidiu Sas > > On 10/23/07, Papadopoulos Georgios <geop(a)altectelecoms.gr> wrote: > > > > > > Hello all, > > > > I am noticing that OpenSER relays an INVITE and the packet that is > > sent out is clipped. You can see in the following packets that the > > forwarded packet is missing a few bytes from the SDP part. Counting > > the bytes shows that the outgoing packet is always 1512 > bytes. Is this > > something that has do with OpenSER (module parameter, > compile options) or is it OS related? > > > > thank you for any help > > > > George > > > > > > > > U 2007/10/23 13:27:46.165232 213.5.1.6:57665 -> > 213.5.43.4:5060 INVITE > > sip:1012118204501@213.5.43.4:5060 SIP/2.0. > > Via: SIP/2.0/UDP > > > 213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9 > hG4bK1594C1AB9. > > Remote-Party-ID: "GEOP Papadopoul" > > <sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off. > > From: "GEOP Papadopoul" > <sip:2116872933@213.5.1.6>;tag=237E8130-1821. > > To: <sip:1012118204501@213.5.43.4>. > > Date: Tue, 23 Oct 2007 10:27:46 GMT. > > Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6. > > Supported: 100rel,timer,resource-priority,replaces. > > Min-SE: 1800. > > Cisco-Guid: 1848010793-2156466652-3218142496-4228086245. > > User-Agent: Cisco-SIPGateway/IOS-12.x. > > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, > > SUBSCRIBE, NOTIFY, INFO, REGISTER. > > CSeq: 101 INVITE. > > Max-Forwards: 70. > > Timestamp: 1193135266. > > Contact: <sip:2116872933@213.5.1.6:5060>. > > Call-Info: > > > <sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Dura > tion=2000". > > Expires: 180. > > Allow-Events: telephone-event. > > Content-Type: application/sdp. > > Content-Disposition: session;handling=required. > > Content-Length: 290. > > . > > v=0. > > o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6. > > s=SIP Call. > > c=IN IP4 213.5.1.6. > > t=0 0. > > m=audio 16454 RTP/AVP 18 0 8 100. > > c=IN IP4 213.5.1.6. > > a=rtpmap:18 G729/8000. > > a=fmtp:18 annexb=yes. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:100 X-NSE/8000. > > a=fmtp:100 192-194. > > > > > > U 2007/10/23 13:27:46.173826 213.5.43.4:5060 -> 213.5.1.6:57665 > > SIP/2.0 100 Giving a try. > > Via: SIP/2.0/UDP > > > 213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9 > hG4bK1594C1AB9;rport=57665. > > From: "GEOP Papadopoul" > <sip:2116872933@213.5.1.6>;tag=237E8130-1821. > > To: <sip:1012118204501@213.5.43.4>. > > Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6. > > CSeq: 101 INVITE. > > Server: Altec Telecoms SIP Proxy. > > Content-Length: 0. > > . > > > > > > U 2007/10/23 13:27:46.174241 213.5.43.4:5060 -> 213.5.17.226:5060 > > INVITE sip:demo1@192.168.1.39:5060 SIP/2.0. > > Record-Route: <sip:213.5.43.4;lr=on;ftag=237E8130-1821>. > > Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bK4acd.5092baa7.0. > > Via: SIP/2.0/UDP > > > 213.5.1.6:5060;rport=57665;x-route-tag="cid:CID-ACN-3@213.5.1. > 6";branch=z9hG4bK1594C1AB9. > > Remote-Party-ID: "GEOP Papadopoul" > > <sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off. > > From: "GEOP Papadopoul" > <sip:2116872933@213.5.1.6>;tag=237E8130-1821. > > To: <sip:1012118204501@213.5.43.4>. > > Date: Tue, 23 Oct 2007 10:27:46 GMT. > > Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6. > > Supported: 100rel,timer,resource-priority,replaces. > > Min-SE: 1800. > > Cisco-Guid: 1848010793-2156466652-3218142496-4228086245. > > User-Agent: Cisco-SIPGateway/IOS-12.x. > > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, > > SUBSCRIBE, NOTIFY, INFO, REGISTER. > > CSeq: 101 INVITE. > > Max-Forwards: 10. > > Timestamp: 1193135266. > > Contact: <sip:2116872933@213.5.1.6:57665>. > > Call-Info: > > > <sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Dura > tion=2000". > > Expires: 180. > > Allow-Events: telephone-event. > > Content-Type: application/sdp. > > Content-Disposition: session;handling=required. > > Content-Length: 290. > > P-hint: NATed client request. > > . > > v=0. > > o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6. > > s=SIP Call. > > c=IN IP4 213.5.1.6. > > t=0 0. > > m=audio 16454 RTP/AVP 18 0 8 100. > > c=IN IP4 213.5.1.6. > > a=rtpmap:18 G729/8000. > > a=fmtp:18 annexb=yes. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:100 X-NSE/8000. > > a=fmtp:100 > > > > Disclaimer > > > > The information in this e-mail and any attachments is > confidential. It > > is intended solely for the attention and use of the named > > addressee(s). If you are not the intended recipient, or person > > responsible for delivering this information to the intended > recipient, please notify the sender immediately. > > Unless you are the intended recipient or his/her representative you > > are not authorized to, and must not, read, copy, distribute, use or > > retain this message or any part of it. E-mail transmission > cannot be > > guaranteed to be secure or error-free as information could be > > intercepted, corrupted, lost, destroyed, arrive late or > incomplete, or contain viruses. > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > >

16 years, 8 months

[OpenSER-Users] INVITE relayed with missing bytes

by Papadopoulos Georgios

Hello all, I am noticing that OpenSER relays an INVITE and the packet that is sent out is clipped. You can see in the following packets that the forwarded packet is missing a few bytes from the SDP part. Counting the bytes shows that the outgoing packet is always 1512 bytes. Is this something that has do with OpenSER (module parameter, compile options) or is it OS related? thank you for any help George U 2007/10/23 13:27:46.165232 213.5.1.6:57665 -> 213.5.43.4:5060 INVITE sip:1012118204501@213.5.43.4:5060 SIP/2.0. Via: SIP/2.0/UDP 213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C 1AB9. Remote-Party-ID: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off. From: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821. To: <sip:1012118204501@213.5.43.4>. Date: Tue, 23 Oct 2007 10:27:46 GMT. Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6. Supported: 100rel,timer,resource-priority,replaces. Min-SE: 1800. Cisco-Guid: 1848010793-2156466652-3218142496-4228086245. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 70. Timestamp: 1193135266. Contact: <sip:2116872933@213.5.1.6:5060>. Call-Info: <sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" . Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 290. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 16454 RTP/AVP 18 0 8 100. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 X-NSE/8000. a=fmtp:100 192-194. U 2007/10/23 13:27:46.173826 213.5.43.4:5060 -> 213.5.1.6:57665 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C 1AB9;rport=57665. From: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821. To: <sip:1012118204501@213.5.43.4>. Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6. CSeq: 101 INVITE. Server: Altec Telecoms SIP Proxy. Content-Length: 0. . U 2007/10/23 13:27:46.174241 213.5.43.4:5060 -> 213.5.17.226:5060 INVITE sip:demo1@192.168.1.39:5060 SIP/2.0. Record-Route: <sip:213.5.43.4;lr=on;ftag=237E8130-1821>. Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bK4acd.5092baa7.0. Via: SIP/2.0/UDP 213.5.1.6:5060;rport=57665;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch= z9hG4bK1594C1AB9. Remote-Party-ID: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off. From: "GEOP Papadopoul" <sip:2116872933@213.5.1.6>;tag=237E8130-1821. To: <sip:1012118204501@213.5.43.4>. Date: Tue, 23 Oct 2007 10:27:46 GMT. Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6. Supported: 100rel,timer,resource-priority,replaces. Min-SE: 1800. Cisco-Guid: 1848010793-2156466652-3218142496-4228086245. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 10. Timestamp: 1193135266. Contact: <sip:2116872933@213.5.1.6:57665>. Call-Info: <sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" . Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 290. P-hint: NATed client request. . v=0. o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6. s=SIP Call. c=IN IP4 213.5.1.6. t=0 0. m=audio 16454 RTP/AVP 18 0 8 100. c=IN IP4 213.5.1.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 X-NSE/8000. a=fmtp:100 Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

16 years, 8 months

[OpenSER-Users] About "q" values

by Iñaki Baz Castillo

Hi, for now I just have found a SIP device (Kphone) where I can choose "q" value for registration. I'd like to know if it's common to find SIP devices sending a "q" value. As I've seen, most of them send nothing so I use my "default_q" value 500 (0,5 in fact). In case there are comon devices sending a "q" in registration, what value is that "q"? I need to know it in order to make a serial forwarding adding contacts from a web application and playing with "q" value. Thanks a lot. -- Iñaki Baz Castillo

16 years, 8 months

[OpenSER-Users] case sensitivity with avp_db_load

by Klaus Darilion

Hi! Currently I have users stored in DB with lower case letters. But I want to allow my user to use upper case letters as well. But this does not work as avp_db_load is case sensitive. Is there a way to handle avp_db_load case insensitive? thanks klaus

16 years, 8 months

[OpenSER-Users] "480 User not responding" instead of "408 Request Timeout" when modparam("tm", "noisy_ctimer", 1)?

by Iñaki Baz Castillo

Hi, if modparam("tm", "noisy_ctimer", 1) and INVITE exceded "fr_inv_timer" then OpenSer sends "408 Request Timeout" to caller and CANCEL to called. I'm sure this is correct according to RFC, but wouldn't be better replying with "480 User not responding"? For example, Asterisk does nothing if it receives "408 Request Timeout" (it ignores it). I've reported about about it: http://bugs.digium.com/view.php?id=11058 Sure it's a fail of Asterisk who shoud accept "408" and terminate the call, but anyway, wouldn't be correct to reply with "480" instead of "408"? Regards. -- Iñaki Baz Castillo

16 years, 8 months

RE: [OpenSER-Users] question about nat pinging

by Papadopoulos Georgios

Hi Bogdan, This is nice to know. However I think it would be easier if someone could control the pinging from script. Maybe a new module parameter that could substitue ping_nated_only with the following values: 0 : ping all contacts 1 : ping contacts with nat flag on 2 : ping contacts with sipping_bflag on just my 2c. Thank you George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Tuesday, October 23, 2007 11:41 AM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] question about nat pinging > > Hi George, > > There is in 1.3 a new MI function for controlling natpings. > See nh_enable_ping : > http://www.openser.org/docs/modules/1.3.x/nathelper.html#AEN432 > > Regards, > Bogdan > > > Papadopoulos Georgios wrote: > > Hello all, > > > > I am trying to find a way to turn off nat pinging from > Openser during > > runtime. Here is the problem: openser1 replicates registrations to > > openser2 which is working as a stand-by. Openser1 sends > OPTIONS pings > > to all nated clients. When either of the two servers receive a > > replicated REGISTER, it saves it without setting the > sipping_bflag. So > > openser2 will not send unecessary OPTIONS pings. However openser2 > > sends 4 byte pings to all nated clients. Is there any way > to turn off > > these pings? > > > > thank you > > > > George > > > > > > Disclaimer > > > > The information in this e-mail and any attachments is > confidential. It > > is intended solely for the attention and use of the named > > addressee(s). If you are not the intended recipient, or person > > responsible for delivering this information to the intended > recipient, > > please notify the sender immediately. Unless you are the intended > > recipient or his/her representative you are not authorized to, and > > must not, read, copy, distribute, use or retain this message or any > > part of it. E-mail transmission cannot be guaranteed to be > secure or > > error-free as information could be intercepted, corrupted, lost, > > destroyed, arrive late or incomplete, or contain viruses. > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

16 years, 8 months

[OpenSER-Users] Help with "lcr" load_contacts() and next_contacts()

by Iñaki Baz Castillo

Hi, I want to try serial forwarding changing "q" values of "location" table. Unfortunatelly my script only does the first INVITE to contact(s) with mayor "q". I've tryed everything and read the entire doc and examples, but get nothing, could you help me please? This is my config and xlogs: ------------------------------------------ modparam("registrar", "append_branches", 1) modparam("usrloc", "db_mode", 3) modparam("lcr", "contact_avp", "$avp(i:711)") route { xlog("#### $rm $ru\n"); lookup("location"); if (load_contacts()) { xlog("L_INFO","#### load_contacts() - avp(i:711) = $avp(i:711) - ds = $ds\n"); } if (next_contacts()) { xlog("L_INFO","#### next_contacts() - avp(i:711) = $avp(i:711) - ds = $ds\n"); t_on_failure("31"); t_relay(); } } failure_route[31] { if (next_contacts()) { xlog("L_INFO","#### failure_route[31] - next_contacts() - avp(i:711) = $avp(i:711) - ds = $ds\n"); t_relay(); exit; } else { xlog("L_ERR","#### failure_route[31] - !next_contacts()\n"); exit; } } ------------------------------------------ Now I call to a user with 2 entries in "location": username domain contact q 800 mydomain.org sip:800@192.168.1.33:5060 0.80 800 mydomain.org sip:800@192.168.1.33:5080 0.50 #### INVITE sip:800@mydomain.org #### load_contacts() - avp(i:711) = <null> - ds = Contact: sip:800@192.168.1.33:5060;transport=udp #### next_contacts() - avp(i:711) = <null> - ds = Contact: sip:800@192.168.1.33:5060;transport=udp << I reject the call in first location >> #### failure_route[31] - next_contacts() - avp(i:711) = <null> - ds = Contact: sip:800@192.168.1.33:5060;transport=udp, <sip:800@192.168.1.33:5080>;q=0 << No other INVITE is sent to second location >> Why $avp(i:711) is always NULL? Why $ds has one contact the first time and both contacts the second time? What am I doing wrong? Thanks a lot for any help. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

16 years, 8 months

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sr-users October 2007