Re: [OpenSIPS-Users] Opensips with MediaProxy
On Tuesday 07 April 2009, Stefano Favaro wrote: Hi, I'm trying to use Opensips 1.5 together with mediaproxy 2.0 in the following scenario: Opensips + Mediarelay and dispatcher on the same machine. The server has 2 network interfaces: the first interface has a public ip, the second one has a private ip. My internal sip systems (switch, gateways, softphones etc.) connect to the private ip. External softphones are registered on the public ip. Mediaproxy is binding on both ip addresses. How exactly do you do that? Mediaproxy can only bind to a single IP. RTP seems to work correctly, but i have some problems with the protocol: I've found these errors on opensips: ERROR:core:udp_send: sendto(sock,0x81b0c30,474,0,0xbf898d78,16): Operation not permitted(1) After the call is connected the 200 OK message is sent continously from the remote party and after 30 seconds the call terminates. Looks like the ACK is not routed properly. Do you think that this solution can be the right one or have you got better suggestions? Can you help me? Thanks. I think things will get much simpler if you only use the public IP and allow forwarding from the private LAN to the public LAN for the devices on the private LAN to be able to use the public addresses of the proxy and the relay. -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Modifying INVITE header to add phone-context
I pretty much solved the issue.
This is what I used:
subst_uri('/^sip:([0-9]+)@(.*)$/sip:\...@\2;phone-context=sip.server.com/i
On Tue, Apr 7, 2009 at 4:25 PM, Julian Yap julianok...@gmail.com wrote:
I have a PSTN gateway which requires a Phone-Context value in the
outgoing SIP INVITE message to further apply ISDN NPI/TON details.
Here's an example of what I currently have going out to the PSTN gateway:
INVITE sip:1...@sip.server.com:5060;user=phone SIP/2.0.
This is what I require:
INVITE sip:1...@sip.server.com:5060;phone-context=sip.server.com;user=phone
SIP/2.0.
Any clues on how to add the Phone-Context value?
Thanks,
Julian
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[OpenSIPS-Users] New MediaProxy release 2.3.3
Hello, There is a new release of MediaProxy available, it contains various bug fixes. To upgrade your debian installation: apt-get update apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- sessions Or download the tar file from: http://download.ag-projects.com/MediaProxy/ The changelog since 2.3.2 is below: mediaproxy (2.3.3) unstable; urgency=low * Re-raise the exception on failing to read RADIUS config file so we get a full traceback * Have dispatcher close TLS connection cleanly when relay has duplicate IP * Added log_level to both Relay and Dispatcher configuration sections * Improved reconnection behaviour in relay to dispatcher When the connection from the relay to the dispatcher is lost, first retry in 1 second, then retry in 10 second on subsequent attempts if it loses the connection again. * Fix bug where relay connects needlessly to previously removed dispatcher * Implemented a keepalive mechanism from relay to dispatcher * On relay reconnect don't have dispatcher query expired sessions * Removed superfluous datatype declaration * Only allow positive integers for time intervals and delays * Updated version dependency for python-application * In dispatcher, replace old connection from relay with new one instead of giving an error * In the dispatcher, check if the reported expired session belongs to the relay that reported it * Improved log messages when a relay reconnects to the dispatcher * In dispatcher, break the connection to a relay if a request times out * In dispatcher check if we know about the session that expired at relay * Use a more robust strategy to disconnect an unresponsive relay Kind regards, Adrian Georgescu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] generate key
Hello, I will generate a certificate and a private key for my server (openxcap) - tls/server.crt - tls/server.key i dont know how to generate this files. regards michael ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] (no subject)
hi all, i have some problems with the configuration from opensips for the openxcap. the route from the file opensips.cfg must be configure my problem is with the main request routing logic I paste my main from the opensips.cfg now i must configure it with this help http://openxcap.org/wiki/Installation; when i copy and paste the main from the homepage and adjust the IP i get errors when i compile the file (/etc/init.d/ opensips start). can somewhere help me to configure the main or can somewhere paste his/her main which works with opensips and opnexcap? regards michael # # $Id: opensips.cfg 5503 2009-03-22 16:22:32Z bogdan_iancu $ # # OpenSIPS basic configuration script # by Anca Vamanu a...@voice-system.ro # # Please refer to the Core CookBook at: # http://www.opensips.org/index.php?n=Resources.DocsCookbooks # for a explanation of possible statements, functions and parameters. # ### Global Parameters # debug=6 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = /etc/opensips/tls/user/user-cert.pem #tls_private_key = /etc/opensips/tls/user/user-privkey.pem #tls_ca_list = /etc/opensips/tls/user/user-calist.pem port=5065 /* uncomment and configure the following line if you want opensips to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.2.5:5065 alias=presence.open-ims.test:5065 #alias=open-ims.test:5065 #alias=scscf.open-ims.test:5065 #alias=presence-server.open-ims.test:5065 ### Modules Section #set module path mpath=/usr/lib/opensips/modules/ /* uncomment next line for MySQL DB support */ loadmodule db_mysql.so loadmodule signaling.so loadmodule sl.so loadmodule tm.so loadmodule rr.so loadmodule maxfwd.so loadmodule usrloc.so loadmodule registrar.so loadmodule textops.so loadmodule mi_fifo.so loadmodule uri_db.so loadmodule uri.so loadmodule xlog.so loadmodule acc.so #loadmodule mi_datagram.so #loadmodule mysql.so #loadmodule presence_mwi.so #loadmodule presence_xcapdiff.so #loadmodule pua.so #loadmodule pua_mi.so #loadmodule rls.so /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule auth.so loadmodule auth_db.so /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule alias_db.so /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see multi-module params section ) */ loadmodule domain.so /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule presence.so loadmodule presence_xml.so # - setting module-specific parameters --- # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) # - rr params - # add value to ;lr param to cope with most of the UAs modparam(rr, enable_full_lr, 1) # do not append from tag to the RR (no need for this script) modparam(rr, append_fromtag, 0) # - registrar params - modparam(registrar, method_filtering, 1) /* uncomment the next line to disable parallel forking via location */ # modparam(registrar, append_branches, 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) # - usrloc params - #modparam(usrloc, db_mode, 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam(usrloc, db_mode, 2) modparam(usrloc, db_url, mysql://opensips:opensip...@localhost/opensips) # - uri_db params - /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam(uri_db, use_uri_table, 0) modparam(uri_db, db_url, ) # - acc params - /* what sepcial events should be accounted ? */ modparam(acc, early_media, 1) modparam(acc, report_ack, 1) modparam(acc, report_cancels, 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be
Re: [OpenSIPS-Users] mysql problem on 1.5
Hi Brett, thanks to your logs, I spoted the problem. The fix is available on SVN. Thanks and regards, Bogdan Brett Nemeroff wrote: Bogdan, For what it's worth, I've updated to latest 1_5 tonight (about 20 minutes ago) and I still am having problems. Full out crashes as well. I rewrote my queries so I'd have a bunch of little (select * from acc where callid=X) kinds of queries. Of course, there is a lot of DB activity while this happens. Crashes start to happen within seconds of the DB activity ramping up. For grins, I slowed my queries down to ensure I only did one query per second (in my database, not opensips).. after about 15-20 queries (different each time really) opensips would just crash. I have acc and sip_trace loaded up, sip_trace isn't active for these calls. Also potentially relevant, my acc table is an InnoDB table. Now if I slowed my call volume to 1CPS and keep the queries at 1 QPS, it seemed to be happier, but still crashes eventually. -Brett On Mon, Apr 6, 2009 at 11:27 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Brett, it looks like the DB connections are dropped and reconnect is taking place (this are the errors about). But to find out the real cause, I can enable some more logs to spot the reason for re-connect... I will do it later as right now I'm in the middle of some DB debugging and I'm afraid of mixing different patches and what goes on SVN :) Regards, Bogdan Brett Nemeroff wrote: Hi All, So I'm doing some load testing with sipp on my opensips 1.5 system. I just checked out (like 2 hours ago, the 1.5 branch from SVN). Everything works just fine, until I run some rating scripts on my database (perl scripts accessing the mysql db directly). While my scripts are running, I see the UAS in sipp retransmitting the 200 OKs and the following gets printed to the syslog: http://www.pastebin.ca/1381169 As soon as my perl script is done, the 200OKs stop retransmitting... My PERL script isn't doing anything terribly unusual, however, it is performing the queries inside of a transaction, including a SELECT/DELETE * FROM acc WHERE kind of clause. Any ideas as to what is causing this? I'm afraid I may be losing call records.. -Brett ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] no audio from caller when using nathelper
Hi Gabriel, Gabriel Bermudez wrote: Thanks for your answer Bogdan Bogdan-Andrei Iancu escribió: Hi Gabriel, So you are not using rtpproxy, but rely on the fact that * is all the time public. In this case, audio from caller to * should work all the time as the destination is public (of course, if the caller does send RTP). For the other way around, you can be sure * sends RTP to the public IP of the NAT (of the client) by doing fix_nated_sdp(1) for the invite - this will force the COMEDIA support in *. Yes, some phones set their contact header with the correct public IP address, that's when rptproxy is not used. In this case they use * directly (but using opensips as a proxy). I do use fix_nated_sdp(1), but only when the nat_uac_test(3) gets passed. maybe the test 3 is not enough to detect all the NAT casestry to use more tests to see if makes a difference. Anyhow, for RTP nat traversal to work, it is mandatory for the party behind the nat to start sending RTP (to open the NAT). If the natted party will send no RTP, there will be no audio at all. And that's exactly what wasn't happening, *sometimes* (the sometimes was the one bugging me really). It seemed not a opensips nat issue but a asterisk nat issue. So I setted up asterisk's realtime with the following view CREATE OR REPLACE VIEW sipfriends AS SELECT subscriber.username AS name, 'friend'::character varying AS type, subscriber.username, subscriber.password AS secret, 'dynamic'::character varying AS host, 'rfc2833'::character varying AS dtmfmode, 'all'::character varying AS disallow, 'g729'::character varying AS allow, 'no'::character varying AS canreinvite, 'yes'::character varying AS nat, 'from-ser'::character varying AS context, ''::character varying AS regserver, 0 AS regseconds FROM subscriber; As you can see, nat=yes always. Seems that this solved the problem, I'll do some more testing tomorrow ;) Cool :) Regards, Bogdan Thanks for your help. Regards, Regard, Bogdan Gabriel Bermudez wrote: Hi everyone, I'm using the nathelper and dispatcher module to send calls to an Asterisk server. I'm using the Asterisk as a SIP to H.323 converter because our PSTN gateway only speaks H.323 For some reason *sometimes* the caller does not send RTP traffic to the opensips (one way audio). The caller's UA is behind a NAT, but it doesn't gets detected as a nated UA, so the RTP flow is between the client's public IP and the Asterisk public IP (rtpproxy is not used). I'm not sure if this problems happens also with UAs that get NAT detected (not seen it happen). I used tshark to capture the invite from an undetected NAT UA (changed the UA ip with *uac_public_ip* and opensip's ip with *opensips_public_ip*) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mysql problem on 1.5
Both. Brett Nemeroff wrote: Is that on the 1_5 branch or trunk? On Wed, Apr 8, 2009 at 7:45 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Brett, thanks to your logs, I spoted the problem. The fix is available on SVN. Thanks and regards, Bogdan Brett Nemeroff wrote: Bogdan, For what it's worth, I've updated to latest 1_5 tonight (about 20 minutes ago) and I still am having problems. Full out crashes as well. I rewrote my queries so I'd have a bunch of little (select * from acc where callid=X) kinds of queries. Of course, there is a lot of DB activity while this happens. Crashes start to happen within seconds of the DB activity ramping up. For grins, I slowed my queries down to ensure I only did one query per second (in my database, not opensips).. after about 15-20 queries (different each time really) opensips would just crash. I have acc and sip_trace loaded up, sip_trace isn't active for these calls. Also potentially relevant, my acc table is an InnoDB table. Now if I slowed my call volume to 1CPS and keep the queries at 1 QPS, it seemed to be happier, but still crashes eventually. -Brett On Mon, Apr 6, 2009 at 11:27 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Brett, it looks like the DB connections are dropped and reconnect is taking place (this are the errors about). But to find out the real cause, I can enable some more logs to spot the reason for re-connect... I will do it later as right now I'm in the middle of some DB debugging and I'm afraid of mixing different patches and what goes on SVN :) Regards, Bogdan Brett Nemeroff wrote: Hi All, So I'm doing some load testing with sipp on my opensips 1.5 system. I just checked out (like 2 hours ago, the 1.5 branch from SVN). Everything works just fine, until I run some rating scripts on my database (perl scripts accessing the mysql db directly). While my scripts are running, I see the UAS in sipp retransmitting the 200 OKs and the following gets printed to the syslog: http://www.pastebin.ca/1381169 As soon as my perl script is done, the 200OKs stop retransmitting... My PERL script isn't doing anything terribly unusual, however, it is performing the queries inside of a transaction, including a SELECT/DELETE * FROM acc WHERE kind of clause. Any ideas as to what is causing this? I'm afraid I may be losing call records.. -Brett ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org mailto:Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] New MediaProxy release 2.3.3 (amd64)
Hello, amd64 packages have been uploaded to the repository. To upgrade your debian installation for 64 bit architectures: apt-get update apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- sessions Kind regards, Adrian Georgescu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ACC Problem. to_did blown away.
Hi Brett, The acc module will log the last RURI of the failed branch...so what was the RURI before sending the INVITE outbecause if I look at the ACK, I guess the INVITE also has mod_sofia, right ? In failure route you cannot change the branch that was just completed If you want something custom to be logged, use the extra_accounting options. Regards, Bogdan Brett Nemeroff wrote: Hey All, I'm sure I'm doing someting stupid here.. In general, I set all the acc_db flags at the top of my script so everything gets logs. I'm getting 486 Busy from the far end and nice, pretty to_did from the original RURI is being blown away with 'sip:mod_sofia' Question is.. what do I need to do to get the 486 logged, but to have the right did in the record.. right now I use $oU. Maybe a revert_uri? (am I making that up?) Here's the 486 from my upstream: U 5.6.239.142:5060 http://5.6.239.142:5060 - 1.2.204.8:5060 http://1.2.204.8:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 1.2.204.8;branch=z9hG4bK5a3.473e0af1.0. Via: SIP/2.0/UDP 2.3.72.138:5060;received=2.3.72.138;branch=z9hG4bK0bf20a91;rport=5060. From: custdomain.com http://custdomain.com sip:custdomain.com http://custdomain.com@2.3.72.138 http://2.3.72.138;tag=as0e9ef59a. To: sip:5212324399...@1.2.204.8 mailto:sip%3a5212324399...@1.2.204.8;tag=BeZcHaUmX1D8Q. Call-ID: 29728c50765bdc3275a6bd375a650...@2.3.72.138 mailto:29728c50765bdc3275a6bd375a650...@2.3.72.138. CSeq: 103 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12911. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Content-Length: 0. . U 2.3.72.138:5060 http://2.3.72.138:5060 - 1.2.204.8:5060 http://1.2.204.8:5060 ACK sip:mod_so...@63.211.239.143:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 2.3.72.138:5060;branch=z9hG4bK0bf20a91;rport. Route: sip:1.2.204.8;lr=on;did=bf5.880cbc94,sip:5.6.239.142;lr=on;ftag=as0e9ef59a. Max-Forwards: 70. From: custdomain.com http://custdomain.com sip:custdomain.com http://custdomain.com@2.3.72.138 http://2.3.72.138;tag=as0e9ef59a. To: sip:5212324399...@1.2.204.8 mailto:sip%3a5212324399...@1.2.204.8;tag=BeZcHaUmX1D8Q. Contact: sip:custdomain.com http://custdomain.com@2.3.72.138 http://2.3.72.138. Call-ID: 29728c50765bdc3275a6bd375a650...@2.3.72.138 mailto:29728c50765bdc3275a6bd375a650...@2.3.72.138. CSeq: 103 ACK. User-Agent: SS. Content-Length: 0. ___ 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] Modifying INVITE header to add phone-context
Hi Julian,
a much nicer option is the add_uri_param() function from URI module:
http://www.opensips.org/html/docs/modules/devel/uri.html#id228164
Regards,
Bogdan
Julian Yap wrote:
I pretty much solved the issue.
This is what I used:
subst_uri('/^sip:([0-9]+)@(.*)$/sip:\...@\2;phone-context=sip.server.com/i
On Tue, Apr 7, 2009 at 4:25 PM, Julian Yap julianok...@gmail.com wrote:
I have a PSTN gateway which requires a Phone-Context value in the
outgoing SIP INVITE message to further apply ISDN NPI/TON details.
Here's an example of what I currently have going out to the PSTN gateway:
INVITE sip:1...@sip.server.com:5060;user=phone SIP/2.0.
This is what I require:
INVITE sip:1...@sip.server.com:5060;phone-context=sip.server.com;user=phone
SIP/2.0.
Any clues on how to add the Phone-Context value?
Thanks,
Julian
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Re: [OpenSIPS-Users] ACC Problem. to_did blown away.
No, actually, the original invite, had a perfectly valid RURI in it.. but the contact had sip:mod_sofia in it.. I am using extra_accounting and I'm looking $oU, which is why I'm confused since the actual $oU is valid and not sip:mod_sofia On Wed, Apr 8, 2009 at 8:40 AM, Bogdan-Andrei Iancu bog...@voice-system.rowrote: Hi Brett, The acc module will log the last RURI of the failed branch...so what was the RURI before sending the INVITE outbecause if I look at the ACK, I guess the INVITE also has mod_sofia, right ? In failure route you cannot change the branch that was just completed If you want something custom to be logged, use the extra_accounting options. Regards, Bogdan Brett Nemeroff wrote: Hey All, I'm sure I'm doing someting stupid here.. In general, I set all the acc_db flags at the top of my script so everything gets logs. I'm getting 486 Busy from the far end and nice, pretty to_did from the original RURI is being blown away with 'sip:mod_sofia' Question is.. what do I need to do to get the 486 logged, but to have the right did in the record.. right now I use $oU. Maybe a revert_uri? (am I making that up?) Here's the 486 from my upstream: U 5.6.239.142:5060 http://5.6.239.142:5060 - 1.2.204.8:5060 http://1.2.204.8:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 1.2.204.8;branch=z9hG4bK5a3.473e0af1.0. Via: SIP/2.0/UDP 2.3.72.138:5060 ;received=2.3.72.138;branch=z9hG4bK0bf20a91;rport=5060. From: custdomain.com http://custdomain.com sip:custdomain.com http://custdomain.com@2.3.72.138 http://2.3.72.138;tag=as0e9ef59a. To: sip:5212324399...@1.2.204.8 sip%3a5212324399...@1.2.204.8 mailto: sip%3a5212324399...@1.2.204.8 sip%253a5212324399...@1.2.204.8 ;tag=BeZcHaUmX1D8Q. Call-ID: 29728c50765bdc3275a6bd375a650...@2.3.72.138 mailto: 29728c50765bdc3275a6bd375a650...@2.3.72.138. CSeq: 103 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12911. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Content-Length: 0. . U 2.3.72.138:5060 http://2.3.72.138:5060 - 1.2.204.8:5060 http://1.2.204.8:5060 ACK sip:mod_so...@63.211.239.143:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 2.3.72.138:5060;branch=z9hG4bK0bf20a91;rport. Route: sip:1.2.204.8;lr=on;did=bf5.880cbc94,sip:5.6.239.142;lr=on;ftag=as0e9ef59a. Max-Forwards: 70. From: custdomain.com http://custdomain.com sip:custdomain.com http://custdomain.com@2.3.72.138 http://2.3.72.138;tag=as0e9ef59a. To: sip:5212324399...@1.2.204.8 sip%3a5212324399...@1.2.204.8 mailto: sip%3a5212324399...@1.2.204.8 sip%253a5212324399...@1.2.204.8 ;tag=BeZcHaUmX1D8Q. Contact: sip:custdomain.com http://custdomain.com@2.3.72.138 http://2.3.72.138. Call-ID: 29728c50765bdc3275a6bd375a650...@2.3.72.138 mailto: 29728c50765bdc3275a6bd375a650...@2.3.72.138. CSeq: 103 ACK. User-Agent: SS. Content-Length: 0. ___ 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] dispatcher and attended transfers
Hi Stan, if I got it right, you want to have a kind of dispatching to guarantee that all in or out calls for user A are going through the same PBX. Correct? And the problem is when you to a REFERyou have A talking to PBX1 and it wants to do a transfer ? Or? Regards, Bogdan Stanisław Pitucha wrote: 2009/4/7 Adrian Georgescu a...@ag-projects.com: You cannot do this reliable the way you propose. The only reliable way is to sit behind a PBX/B2BUA that your control and behaves in a consistent and reliable way. Otherwise you are at the mercy at the combinations of the SIP User Agents that are involved in the call transfer operation. There is only one specific scenario I want to support: - phone has a dialog already open to a PBX - phone sends an new call INVITE to a PBX - phone joins the call legs with a REFER I think, this is the PBX/B2BUA situation you're talking about? I'm not sure what you mean by the combinations of the SIP User Agents that are involved. I didn't have any problems with this setup as long as the same phone always uses the same pbx. If you will try to fix incrementally every problem your discover in the SIP Proxy for call transfer you will be busy forever solving this because is end-point implementation dependent. I'm only trying to solve failover + distribution over PBXes in the proxy. Transfers are properly handled by N asterisk hosts. To be specific - my network looks like this: UAs - openser (with dispatcher) - N identical asterisk boxes All calls go through one of the asterisk boxes. Stan ___ 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] generate key
Be sure to read through the links that you are given so that you have a good understanding, but here are the steps I always take Now we need to create the TLS certifications and Keys (http://www.imacat.idv.tw/tech/sslcerts.html Read Create a Server Certificate) openssl genrsa -des3 -out /etc/ssl/private/openxcap.key 2048 -- Set the password to whatever you want chmod og-rwx /etc/ssl/private/openxcap.key openssl req -new -key /etc/ssl/private/openxcap.key -out /tmp/openxcap.req US State City Home Home openxcap01.blahblah.com CA openssl x509 -req -days 7305 -sha1 \ -extfile /etc/ssl/openssl.cnf -extensions v3_ca \ -signkey /etc/ssl/private/openxcap.key \ -in /tmp/openxcap.req -out /etc/ssl/certs/openxcap.crt rm -f /tmp/openxcap.req openssl genrsa -out /etc/openxcap/tls/openxcapserver.key 2048 chmod og-rwx /etc/openxcap/tls/openxcapserver.key openssl req -new -key /etc/openxcap/tls/openxcapserver.key -out /tmp/openxcapserver.req BE SURE NOT TO SET A PASSWORD** US State City Home Home openxcap01.blahblah.com openssl x509 -req -days 3650 -sha1 \ -extfile /etc/ssl/openssl.cnf -extensions v3_req \ -CA /etc/ssl/certs/openxcap.crt -CAkey /etc/ssl/private/openxcap.key \ -CAserial /etc/ssl/openxcap.srl -CAcreateserial \ -in /tmp/openxcapserver.req -out /etc/openxcap/tls/openxcapserver.crt openxcap.crt is the key that needs to be given out to the clients (Bria) - Copy it to the desktop, open IE and click on Tools - Internet Options - Content Tab - Certifications Button - Import - And select Automatically select the certificate store based on the type of certificate Then configure Bria with the following Presence Tab - Mode = Presence Agent Storage Tab - Storage Method = XCAP Root URL: https://openxcap01.blahblah.com/xcap-root/ Good Luck On Apr 8, 2009 4:28am, Uwe Kastens ki...@kiste.org wrote: Hi Michael, Try searching for openssl. http://sial.org/howto/openssl/self-signed/ BR Uwe Hello, I will generate a certificate and a private key for my server (openxcap) - tls/server.crt - tls/server.key i dont know how to generate this files. regards michael ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ 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] New MediaProxy release 2.3.3 (amd64)
2009/4/8 Adrian Georgescu a...@ag-projects.com: Hello, amd64 packages have been uploaded to the repository. To upgrade your debian installation for 64 bit architectures: apt-get update apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- sessions pbx:/etc/mediaproxy/tls# apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: mediaproxy-common The following NEW packages will be installed mediaproxy-common mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions 0 upgraded, 4 newly installed, 0 to remove and 0 not upgraded. Need to get 124kB/200kB of archives. After this operation, 942kB of additional disk space will be used. Do you want to continue [Y/n]? y Get: 1 http://ag-projects.com unstable/main mediaproxy-dispatcher 2.3.3 [15.6kB] Get: 2 http://ag-projects.com unstable/main mediaproxy-relay 2.3.3 [15.7kB] Get: 3 http://ag-projects.com unstable/main mediaproxy-web-sessions 2.3.3 [92.7kB] Fetched 124kB in 1s (97.3kB/s) Selecting previously deselected package mediaproxy-common. (Reading database ... 26719 files and directories currently installed.) Unpacking mediaproxy-common (from .../mediaproxy-common_2.3.3_amd64.deb) ... Selecting previously deselected package mediaproxy-dispatcher. Unpacking mediaproxy-dispatcher (from .../mediaproxy-dispatcher_2.3.3_all.deb) ... dpkg: error processing /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb (--unpack): trying to overwrite `/usr/bin/media-dispatcher', which is also in package mediaproxy-common Selecting previously deselected package mediaproxy-relay. Unpacking mediaproxy-relay (from .../mediaproxy-relay_2.3.3_all.deb) ... dpkg: error processing /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb (--unpack): trying to overwrite `/usr/bin/media-relay', which is also in package mediaproxy-common Selecting previously deselected package mediaproxy-web-sessions. Unpacking mediaproxy-web-sessions (from .../mediaproxy-web-sessions_2.3.3_all.deb) ... Errors were encountered while processing: /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb E: Sub-process /usr/bin/dpkg returned an error code (1) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mysql problem on 1.5
Bogdan,I no longer get crashes. However the opensips process hangs pretty badly while the DB operations are going on. I've tried to rewrite my queries to do more small queries rather than longer slow ones. So what I'm doing, I'm using sipp performing calls at 30CPS lasting 10 seconds (to generate a lot of call records). While this is running, I run my rating script, which gathers unique callid. smashes records together into a cdr record. My database engine is InnoDB and I'm using transactions. I'm not actually getting to a commit in any of this. So while my script is running. I see on the UAS side of sipp, it stops receiving calls, and starts performing retransmissions. I've verified with tshark that packets are hitting opensips, but not getting a reply. I have 20 children running. Am I doing something wrong? Thanks for your help, Brett On Wed, Apr 8, 2009 at 7:54 AM, Bogdan-Andrei Iancu bog...@voice-system.rowrote: Both. Brett Nemeroff wrote: Is that on the 1_5 branch or trunk? On Wed, Apr 8, 2009 at 7:45 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Brett, thanks to your logs, I spoted the problem. The fix is available on SVN. Thanks and regards, Bogdan Brett Nemeroff wrote: Bogdan, For what it's worth, I've updated to latest 1_5 tonight (about 20 minutes ago) and I still am having problems. Full out crashes as well. I rewrote my queries so I'd have a bunch of little (select * from acc where callid=X) kinds of queries. Of course, there is a lot of DB activity while this happens. Crashes start to happen within seconds of the DB activity ramping up. For grins, I slowed my queries down to ensure I only did one query per second (in my database, not opensips).. after about 15-20 queries (different each time really) opensips would just crash. I have acc and sip_trace loaded up, sip_trace isn't active for these calls. Also potentially relevant, my acc table is an InnoDB table. Now if I slowed my call volume to 1CPS and keep the queries at 1 QPS, it seemed to be happier, but still crashes eventually. -Brett On Mon, Apr 6, 2009 at 11:27 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Brett, it looks like the DB connections are dropped and reconnect is taking place (this are the errors about). But to find out the real cause, I can enable some more logs to spot the reason for re-connect... I will do it later as right now I'm in the middle of some DB debugging and I'm afraid of mixing different patches and what goes on SVN :) Regards, Bogdan Brett Nemeroff wrote: Hi All, So I'm doing some load testing with sipp on my opensips 1.5 system. I just checked out (like 2 hours ago, the 1.5 branch from SVN). Everything works just fine, until I run some rating scripts on my database (perl scripts accessing the mysql db directly). While my scripts are running, I see the UAS in sipp retransmitting the 200 OKs and the following gets printed to the syslog: http://www.pastebin.ca/1381169 As soon as my perl script is done, the 200OKs stop retransmitting... My PERL script isn't doing anything terribly unusual, however, it is performing the queries inside of a transaction, including a SELECT/DELETE * FROM acc WHERE kind of clause. Any ideas as to what is causing this? I'm afraid I may be losing call records.. -Brett ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org mailto:Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dialog profiles, DBs and restarts
Bogdan, No problem. Where does one do that? - Jeff On 4/8/09 10:26 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, It is a know limitation that I what to address in the next version - please open a feature request for it. Regards, Bogdan Jeff Pyle wrote: Hello, It appears that while dialogs themselves survive opensips restarts (with db_mode=1), the profile/value associations do not. Is this configurable? If not, is there a formal process to submit a feature request? Thanks, Jeff ___ 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] dispatcher and attended transfers
2009/4/8 Bogdan-Andrei Iancu bog...@voice-system.ro:
if I got it right, you want to have a kind of dispatching to guarantee that
all in or out calls for user A are going through the same PBX. Correct?
In short:
Yes. But internal calls should use only one PBX in the cluster, not two.
Long explanation:
Actually, we don't care where the first call is exactly (can be
randomised / load balanced). Just that if some user is already talking
to someone on PBX X, we want his next call to route via X too.
We've got that already working via a hack - we're getting the caller /
callee contact from the dialog table into an avp. Then if
$avp(s:contact){uri.host} matches any of our pbxes, we use it for
routing; otherwise we use a dispatcher. That works relatively well,
but breaks when one of the PBXes dies for example. Dialog stays in the
database forever and that user can't dial out anyone, because he
always gets routed to the dead IP.
We can't simply route all the calls to/from user A to the same PBX,
because we'd like to use only one PBX when doing internal calls. If we
want internal calls to use only one PBX, hashing just on From: and
To: headers is not enough, because we have 2 transfer scenarios:
A calls B, B calls C, transfers
- (two different destinations - B, C and sources - A, B)
A calls B, A calls C, transfers
- (two different destinations - B, C, same sources)
Good enough solution would be, to set the $avp(s:contact){uri.host} as
the first destination, with failover positions filled with the result
of ds_select_domain().
Ok... I hope that explained the situation :)
Thanks,
Stan
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Re: [OpenSIPS-Users] New MediaProxy release 2.3.3
Hello, I attempted an upgrade from 2.3.2, recompiling from source. Everything seemed to build and install okay. But, when running media-dispatcher on 2.3.3, I receive the following error: Traceback (most recent call last): File /usr/bin/media-dispatcher, line 32, in ? log.level.current = config_file.get_option(Dispatcher, 'log_level', default=log.level.DEBUG, type=datatypes.LogLevel) AttributeError: 'module' object has no attribute 'level' I reinstalled 2.3.2 and the error went away. - Jeff On 4/8/09 5:08 AM, Adrian Georgescu a...@ag-projects.com wrote: Hello, There is a new release of MediaProxy available, it contains various bug fixes. To upgrade your debian installation: apt-get update apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions Or download the tar file from: http://download.ag-projects.com/MediaProxy/ The changelog since 2.3.2 is below: mediaproxy (2.3.3) unstable; urgency=low * Re-raise the exception on failing to read RADIUS config file so we get a full traceback * Have dispatcher close TLS connection cleanly when relay has duplicate IP * Added log_level to both Relay and Dispatcher configuration sections * Improved reconnection behaviour in relay to dispatcher When the connection from the relay to the dispatcher is lost, first retry in 1 second, then retry in 10 second on subsequent attempts if it loses the connection again. * Fix bug where relay connects needlessly to previously removed dispatcher * Implemented a keepalive mechanism from relay to dispatcher * On relay reconnect don't have dispatcher query expired sessions * Removed superfluous datatype declaration * Only allow positive integers for time intervals and delays * Updated version dependency for python-application * In dispatcher, replace old connection from relay with new one instead of giving an error * In the dispatcher, check if the reported expired session belongs to the relay that reported it * Improved log messages when a relay reconnects to the dispatcher * In dispatcher, break the connection to a relay if a request times out * In dispatcher check if we know about the session that expired at relay * Use a more robust strategy to disconnect an unresponsive relay Kind regards, Adrian Georgescu ___ 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] New MediaProxy release 2.3.3 (amd64)
Similarly, ... Processing triggers for man-db ... Errors were encountered while processing: /var/cache/apt/archives/mediaproxy-common_2.3.3_amd64.deb E: Sub-process /usr/bin/dpkg returned an error code (1) - Jeff On 4/8/09 10:23 AM, Gavin Henry gavin.he...@gmail.com wrote: 2009/4/8 Adrian Georgescu a...@ag-projects.com: Hello, amd64 packages have been uploaded to the repository. To upgrade your debian installation for 64 bit architectures: apt-get update apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- sessions pbx:/etc/mediaproxy/tls# apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: mediaproxy-common The following NEW packages will be installed mediaproxy-common mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions 0 upgraded, 4 newly installed, 0 to remove and 0 not upgraded. Need to get 124kB/200kB of archives. After this operation, 942kB of additional disk space will be used. Do you want to continue [Y/n]? y Get: 1 http://ag-projects.com unstable/main mediaproxy-dispatcher 2.3.3 [15.6kB] Get: 2 http://ag-projects.com unstable/main mediaproxy-relay 2.3.3 [15.7kB] Get: 3 http://ag-projects.com unstable/main mediaproxy-web-sessions 2.3.3 [92.7kB] Fetched 124kB in 1s (97.3kB/s) Selecting previously deselected package mediaproxy-common. (Reading database ... 26719 files and directories currently installed.) Unpacking mediaproxy-common (from .../mediaproxy-common_2.3.3_amd64.deb) ... Selecting previously deselected package mediaproxy-dispatcher. Unpacking mediaproxy-dispatcher (from .../mediaproxy-dispatcher_2.3.3_all.deb) ... dpkg: error processing /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb (--unpack): trying to overwrite `/usr/bin/media-dispatcher', which is also in package mediaproxy-common Selecting previously deselected package mediaproxy-relay. Unpacking mediaproxy-relay (from .../mediaproxy-relay_2.3.3_all.deb) ... dpkg: error processing /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb (--unpack): trying to overwrite `/usr/bin/media-relay', which is also in package mediaproxy-common Selecting previously deselected package mediaproxy-web-sessions. Unpacking mediaproxy-web-sessions (from .../mediaproxy-web-sessions_2.3.3_all.deb) ... Errors were encountered while processing: /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb E: Sub-process /usr/bin/dpkg returned an error code (1) ___ 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] New MediaProxy release 2.3.3 (amd64)
And trying to create the package from source gives the following error: Now signing changes and any dsc files... signfile mediaproxy_2.3.3.dsc Dan Pascu d...@ag-projects.com gpg: skipped Dan Pascu d...@ag-projects.com: secret key not available gpg: [stdin]: clearsign failed: secret key not available debsign: gpg error occurred! Aborting debuild: fatal error at line 1250: running debsign failed - Jeff On 4/8/09 10:57 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Similarly, ... Processing triggers for man-db ... Errors were encountered while processing: /var/cache/apt/archives/mediaproxy-common_2.3.3_amd64.deb E: Sub-process /usr/bin/dpkg returned an error code (1) - Jeff On 4/8/09 10:23 AM, Gavin Henry gavin.he...@gmail.com wrote: 2009/4/8 Adrian Georgescu a...@ag-projects.com: Hello, amd64 packages have been uploaded to the repository. To upgrade your debian installation for 64 bit architectures: apt-get update apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- sessions pbx:/etc/mediaproxy/tls# apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: mediaproxy-common The following NEW packages will be installed mediaproxy-common mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions 0 upgraded, 4 newly installed, 0 to remove and 0 not upgraded. Need to get 124kB/200kB of archives. After this operation, 942kB of additional disk space will be used. Do you want to continue [Y/n]? y Get: 1 http://ag-projects.com unstable/main mediaproxy-dispatcher 2.3.3 [15.6kB] Get: 2 http://ag-projects.com unstable/main mediaproxy-relay 2.3.3 [15.7kB] Get: 3 http://ag-projects.com unstable/main mediaproxy-web-sessions 2.3.3 [92.7kB] Fetched 124kB in 1s (97.3kB/s) Selecting previously deselected package mediaproxy-common. (Reading database ... 26719 files and directories currently installed.) Unpacking mediaproxy-common (from .../mediaproxy-common_2.3.3_amd64.deb) ... Selecting previously deselected package mediaproxy-dispatcher. Unpacking mediaproxy-dispatcher (from .../mediaproxy-dispatcher_2.3.3_all.deb) ... dpkg: error processing /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb (--unpack): trying to overwrite `/usr/bin/media-dispatcher', which is also in package mediaproxy-common Selecting previously deselected package mediaproxy-relay. Unpacking mediaproxy-relay (from .../mediaproxy-relay_2.3.3_all.deb) ... dpkg: error processing /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb (--unpack): trying to overwrite `/usr/bin/media-relay', which is also in package mediaproxy-common Selecting previously deselected package mediaproxy-web-sessions. Unpacking mediaproxy-web-sessions (from .../mediaproxy-web-sessions_2.3.3_all.deb) ... Errors were encountered while processing: /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb E: Sub-process /usr/bin/dpkg returned an error code (1) ___ 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] New MediaProxy release 2.3.3 (amd64)
2009/4/8 Jeff Pyle jp...@fidelityvoice.com: And trying to create the package from source gives the following error: Now signing changes and any dsc files... signfile mediaproxy_2.3.3.dsc Dan Pascu d...@ag-projects.com gpg: skipped Dan Pascu d...@ag-projects.com: secret key not available gpg: [stdin]: clearsign failed: secret key not available debsign: gpg error occurred! Aborting debuild: fatal error at line 1250: running debsign failed The INSTALL guide mentions this and that it's ok due to you not having the signing key. You'll find the deb in the ../ directory. Cheers. -- http://www.suretecsystems.com/services/openldap/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] New MediaProxy release 2.3.3 (amd64)
Understood. I¹m rather new to the deb package process, and I did discover the deb packages one level up. Is it possible to run a 2.3.2 dispatcher and a 2.3.3 relay, or do the versions needs to match exactly? - Jeff On 4/8/09 11:11 AM, Adrian Georgescu a...@ag-projects.com wrote: You obviously do not have the key of the developer who made the package but you still have the package built. This is not an error per se. Adrian On Apr 8, 2009, at 5:06 PM, Jeff Pyle wrote: And trying to create the package from source gives the following error: Now signing changes and any dsc files... signfile mediaproxy_2.3.3.dsc Dan Pascu d...@ag-projects.com gpg: skipped Dan Pascu d...@ag-projects.com: secret key not available gpg: [stdin]: clearsign failed: secret key not available debsign: gpg error occurred! Aborting debuild: fatal error at line 1250: running debsign failed - Jeff On 4/8/09 10:57 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Similarly, ... Processing triggers for man-db ... Errors were encountered while processing: /var/cache/apt/archives/mediaproxy-common_2.3.3_amd64.deb E: Sub-process /usr/bin/dpkg returned an error code (1) - Jeff On 4/8/09 10:23 AM, Gavin Henry gavin.he...@gmail.com wrote: 2009/4/8 Adrian Georgescu a...@ag-projects.com: Hello, amd64 packages have been uploaded to the repository. To upgrade your debian installation for 64 bit architectures: apt-get update apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- sessions pbx:/etc/mediaproxy/tls# apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: mediaproxy-common The following NEW packages will be installed mediaproxy-common mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions 0 upgraded, 4 newly installed, 0 to remove and 0 not upgraded. Need to get 124kB/200kB of archives. After this operation, 942kB of additional disk space will be used. Do you want to continue [Y/n]? y Get: 1 http://ag-projects.com unstable/main mediaproxy-dispatcher 2.3.3 [15.6kB] Get: 2 http://ag-projects.com unstable/main mediaproxy-relay 2.3.3 [15.7kB] Get: 3 http://ag-projects.com unstable/main mediaproxy-web-sessions 2.3.3 [92.7kB] Fetched 124kB in 1s (97.3kB/s) Selecting previously deselected package mediaproxy-common. (Reading database ... 26719 files and directories currently installed.) Unpacking mediaproxy-common (from .../mediaproxy-common_2.3.3_amd64.deb) ... Selecting previously deselected package mediaproxy-dispatcher. Unpacking mediaproxy-dispatcher (from .../mediaproxy-dispatcher_2.3.3_all.deb) ... dpkg: error processing /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb (--unpack): trying to overwrite `/usr/bin/media-dispatcher', which is also in package mediaproxy-common Selecting previously deselected package mediaproxy-relay. Unpacking mediaproxy-relay (from .../mediaproxy-relay_2.3.3_all.deb) ... dpkg: error processing /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb (--unpack): trying to overwrite `/usr/bin/media-relay', which is also in package mediaproxy-common Selecting previously deselected package mediaproxy-web-sessions. Unpacking mediaproxy-web-sessions (from .../mediaproxy-web-sessions_2.3.3_all.deb) ... Errors were encountered while processing: /var/cache/apt/archives/mediaproxy-dispatcher_2.3.3_all.deb /var/cache/apt/archives/mediaproxy-relay_2.3.3_all.deb E: Sub-process /usr/bin/dpkg returned an error code (1) ___ 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] CDRTool - ShowPrice - No match for gateway parameter
Guys, some strange thing I noticed in the last versions of CDRTool related to usage of the Gateway parameter in ShowPrice. Based on logs it looks like the gateway parameter is somehow faked (or perhaps wrongly converted). 1. On ShowPrice commands: * Using default dataset, I have replaced the default entry (gateway, domain, subscriber empty) with (gateway=10.0.0.1 , subscriber and domain empty). Reloaded the cdrtool from console and executed: ShowPrice From=sip:1...@example2.com To=sip:0031650222...@example.com Gateway=10.0.0.1 Duration=59 The answer was: 0 In the syslog I could find: Apr 8 17:12:54 DellLaptop cdrtool[11081]: ShowPrice From=sip:1...@example2.com To=sip:0031650222...@example.com Gateway=10.0.0.1 Duration=59 Apr 8 17:12:54 DellLaptop cdrtool[11081]: Error: no customer found in billing_customers table for billing party=...@example2.com, domain=example2.com, gateway=0.0.0.0 In the mysql table I have: mysql select * from billing_customers; ++--+-+---+---+---+---+---+--+---+--+--+ | id | gateway | domain | subscriber| profile_name1 | profile_name1_alt | profile_name2 | profile_name2_alt | timezone | increment | min_duration | country_code | ++--+-+---+---+---+---+---+--+---+--+--+ | 4 | 10.0.0.1 | | | 441 | | 442 | | Europe/Amsterdam | 0 | 0 | | | 5 | | example.com | | 441 | | 442 | | Europe/Amsterdam | 0 | 0 | | | 6 | | | al...@example.com | 441 | | 442 | | Europe/Amsterdam | 0 | 0 | | ++--+-+---+---+---+---+---+--+---+--+--+ 3 rows in set (0.01 sec) Ta, DanB ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RtpRroxy sturtup (init.d) script for redhat
Bogdan-Andrei Iancu wrote: Hi Vladimir, really nice, indeed - I did this manually all the time :) Maybe Maxim can integrate this directly in the RTPproxy project Yes, I will do it. In fact we plan moving towards multi-threading design in the next release, which should make utilizing multi-core chips much easier. Regards, -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts T/F: +1-646-651-1110 Web: http://www.sippysoft.com MSN: sa...@sippysoft.com Skype: SippySoft ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RtpRroxy sturtup (init.d) script for redhat
Romanov Vladimir wrote: Hi! Could you please add command line option to change syslog FACILITY? Now I simply modify this in source and recompile. Vladimir, Can you please send a patch? Thanks! Regards, -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts T/F: +1-646-651-1110 Web: http://www.sippysoft.com MSN: sa...@sippysoft.com Skype: SippySoft ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] sst min-se problem
Hi Bogdan, If the current code is operating contrary to the RFC, how might one such as me request it be updated? - Jeff On 4/6/09 1:10 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Looking in the code, the 422 is sent only if the proxy min-se (1800) is smaller than the min(received-min_se(90), received-se(300)) - 1800 90 - false, no 422. But reading the RFC 4028, I would say the condition is the other way around - if the local min-se is higher than min(received-min_se(90), received-se(300)) , the 422 should be sent out. Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, Makes sense, but the why didn't the proxy reject the request with a 422 since the Session-Expires from the request is less than the proxy's Min-SE of 1800? - Jeff On 4/6/09 12:56 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, What you configure is the min-se of the proxy. (1800) In DBG:sst:sst_check_min: Session-Expires: 300; MIN-SE: 90 are the values from received from request. Regards, Bogdan Jeff Pyle wrote: Hello, I have the sst module configured as follows: loadmodule sst.so modparam(sst|dialog, timeout_avp, $avp(s:sst_timeout)) modparam(sst, sst_flag, 6) modparam(sst, enable_stats, 1) modparam(sst, min_se, 1800) modparam(sst, reject_to_small, 1) Opensips 1.5 receives an invite containing the following header: Session-Expires: 300 sstCheckMin(1) at debug=6 shows this: DBG:sst:sst_check_min: No MIN-SE header found. DBG:sst:sst_check_min: Session-Expires: 300; MIN-SE: 90 DBG:sst:sst_check_min: Done returning false (-1) Since the invite from my gateway didn't contain a MIN-SE, why doesn't it use the 1800 provided at the modparam? Thanks, Jeff ___ 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 1.5 with load_balancing
Hi Bogdan, Sorry, I need to clear up the configuration before trying use loadbalancer. The behaviour was every time I made a call a little bit strange - but every time in an different way. I will setup some virtual servers and play around with the configuration. Thanks Uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, But there is not ERROR (as you mentioned) in the log you sent. Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, Here we go. BR Uwe Bogdan-Andrei Iancu schrieb: HI Uwe, can you post a debug=6 log of the entire call? Thanks and regards, Bogdan Uwe Kastens wrote: Hi, I configured load_balancing following the tutorial. The call is relayed via t_relay to the 1st pstn gw. After that I will receive the following error: ERROR:load_balancer:do_load_balance: failed to create dialog and it looks like, that I am missing some answers. BR uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRTool - ShowPrice - No match for gateway parameter
More on the subject ... Just to be sure that I am not doing any mistake, the log of mysql for the same command (ShowPrice From=sip:1...@example2.com To=sip:0031650222...@example.com Gateway=10.0.0.1 Duration=59) shows the gateway parameter queried as suspected, faked: 090408 21:17:22 314 Init DB cdrtool 314 Query select * from billing_customers where subscriber = '1...@example2.com' or domain= 'example2.com' or gateway = '0.0.0.0' or (subscriber = '' and domain = '' and gateway = '') order by subscriber desc, domain desc, gateway desc limit 1 Ta, DanB On Wed, 2009-04-08 at 17:34 +0200, Dan-Cristian Bogos wrote: Guys, some strange thing I noticed in the last versions of CDRTool related to usage of the Gateway parameter in ShowPrice. Based on logs it looks like the gateway parameter is somehow faked (or perhaps wrongly converted). 1. On ShowPrice commands: * Using default dataset, I have replaced the default entry (gateway, domain, subscriber empty) with (gateway=10.0.0.1 , subscriber and domain empty). Reloaded the cdrtool from console and executed: ShowPrice From=sip:1...@example2.com To=sip:0031650222...@example.com Gateway=10.0.0.1 Duration=59 The answer was: 0 In the syslog I could find: Apr 8 17:12:54 DellLaptop cdrtool[11081]: ShowPrice From=sip:1...@example2.com To=sip:0031650222...@example.com Gateway=10.0.0.1 Duration=59 Apr 8 17:12:54 DellLaptop cdrtool[11081]: Error: no customer found in billing_customers table for billing party=...@example2.com, domain=example2.com, gateway=0.0.0.0 In the mysql table I have: mysql select * from billing_customers; ++--+-+---+---+---+---+---+--+---+--+--+ | id | gateway | domain | subscriber| profile_name1 | profile_name1_alt | profile_name2 | profile_name2_alt | timezone | increment | min_duration | country_code | ++--+-+---+---+---+---+---+--+---+--+--+ | 4 | 10.0.0.1 | | | 441 | | 442 | | Europe/Amsterdam | 0 | 0 | | | 5 | | example.com | | 441 | | 442 | | Europe/Amsterdam | 0 | 0 | | | 6 | | | al...@example.com | 441 | | 442 | | Europe/Amsterdam | 0 | 0 | | ++--+-+---+---+---+---+---+--+---+--+--+ 3 rows in set (0.01 sec) Ta, DanB ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
