[OpenSIPS-Users] [OpenSIPS] sample script that works with cdrtools, freeradius, nat, drouting
Hi, There: It looks ag-projects is maintaining the cdrtools, media proxy. but I searched around and didn't find anywhere there is a script that supports all the needed feature: cdrtools, mediaproxy, nat_traversal, and drouting. so now I'm trying to be a little brave and post my script that includes all above. this script doesn't handle instance message, but only voice calls. can any body spot problems with this script ? The goal of the script is to let locally registered user to use gateway to make outgoing call, and receive incoming call. the numbering plan is for US. free radius should have good authenticaing and accounting for different messages, and some special DID are mapped to several numbers and routed to asterisk. Hopefully this script will be useful for a general VOIP carrier. I try to paste the document to be comment. Hopefully, by going through this exercise, we can get a good starting script for people to use as a model starting script. Jimmy ### # # $Id: opensips.cfg,v 1.13 2009/05/11 06:06:00 jinsong Exp $ # # OpenSIPS basic configuration script # by Anca Vamanu a...@voice-system.ro # # Please refer to the Core CookBook at http://www.opensips.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # #INVITE :Invites a user to a call #ACK : Acknowledgement is used to facilitate reliable message exchange for INVITEs. #BYE :Terminates a connection between users #CANCEL :Terminates a request, or search, for a user. It is used if a client sends an INVITE and then changes its decision to call the recipient. #OPTIONS :Solicits information about a server's capabilities. #REGISTER :Registers a user's current location #INFO :Used for mid-session signaling #MESSAGE : IMS send message #SUBSCRIBE : IMS presence subscribe message #PUBLISH: IMS publish message #1xx: Provisional -- request received, continuing to process the request; #2xx: Success -- the action was successfully received, understood, and accepted; #3xx: Redirection -- further action needs to be taken in order to complete the request; #4xx: Client Error -- the request contains bad syntax or cannot be fulfilled at this server; #5xx: Server Error -- the server failed to fulfill an apparently valid request; #6xx: Global Failure -- the request cannot be fulfilled at any server. #This function sets the value of the flag given as parameter to 1 (true). The value of the parameter must be an integer between 0 and 31. ### Global Parameters # debug=3 log_stderror=no 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 #disable dns to scale dns=no rev_dns=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no alias=machinename.somedomain.com /* 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=5060 /* 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:192.168.1.2:5060 ### Modules Section #set module path mpath=/usr/lib/opensips/modules/ /* uncomment next line for MySQL DB support */ loadmodule db_mysql.so loadmodule mi_fifo.so loadmodule sl.so loadmodule tm.so loadmodule rr.so loadmodule maxfwd.so loadmodule usrloc.so loadmodule signaling.so loadmodule registrar.so loadmodule textops.so loadmodule uri_db.so loadmodule uri.so loadmodule xlog.so loadmodule acc.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
[OpenSIPS-Users] Rebooting a Linksys through OpenSIPS
Hi all, I'm still dipping my toes in deep waters with OpenSips 1.5... And I've come across a small problem. I would like to reboot Linksys ATA (PAP2 etc...) using OpenSIPS. Can this maybe be done? Basically I use XML-RPC FIFO to send the SIP command. The problem is that Linksys replies with 401 challenge. Can I, or is it at all possible to reply to this challenge using OpenSIPS? Below is my PHP for this request. Note that I have coded end user IP and no in the PHP as this gives me one less place for problems. J ? # Using the XML-RPC extension to format the XML package echo(plaintext); #For pretty browser output $params[]= NOTIFY; $params[]= 'sip:1...@172.16.245.128'; $params[]= '.'; $params[]= '.'; $params[]= From: sip:xxx.xx.xxx.xxx\r\nTo: sip:1...@xxx.xx.xxx.xxx\r\nEvent: reboot_now\r\nContact: sip:dae...@!!\r\nContent-Length: 0\r\nContent-Type: text/plain\r\n; $request = xmlrpc_encode_request(t_uac_dlg, $params); #$request = xmlrpc_encode_request(which, NULL); # For testing of XMLRPC # Using the cURL extension to send it off, first creating a custom header block $header[] = Host: 127.0.0.1; $header[] = Connection: close; $header[] = User-Agent: OpenSIPg XML_RPC Client; $header[] = Content-type: text/xml; print_r($request); #debug echo(\n\n); #debug print_r($header); #debug $ch = curl_init(); curl_setopt( $ch, CURLOPT_URL, http://127.0.0.1/RPC2;); # URL to post to curl_setopt( $ch, CURLOPT_PORT, 8080); # URL to post to curl_setopt( $ch, CURLOPT_RETURNTRANSFER, 1 ); # return into a variable curl_setopt( $ch, CURLOPT_HTTPHEADER, $header ); # custom headers, see above curl_setopt( $ch, CURLOPT_POSTFIELDS, $request ); curl_setopt( $ch, CURLOPT_CUSTOMREQUEST, 'POST' ); # This POST is special, and uses its specified Content-type $result = curl_exec( $ch ); # run! curl_close($ch); echo $result; ? Best Regards, Matti Zemack, Stockholm, Sweden ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] All debugs to set a diagnostic of why dialog are not removed in my OpenSIPs 1.5.1 (SIP trace + opensips debug 4 + opensipsctl fifo dlg_list )
When monitoring a REGISTER dialog, I have found that is creates 2 dialogs, here is an example, it creates 2 dialogs (first REGISTER / 401 Unauthorized) with dialog 3102:436706855 or 0x2b2cc4ab4b58 and after (REGISTER / 200 OK) with dialog 3102:436706856 or 0x2b2cc4ab52d8 The first one is correctly removed, but not the second one! I have to wait 3 hours before its removal Here is the second dialog that is not removed: dialog:: hash=3102:436706856 state:: 3 user_flags:: 0 timestart:: 1242030852 timeout:: 10915 callid:: jjolrfphvjri...@leonhart.interne.smart-telecom.ch from_uri:: sip:0213115...@212.147.46.91 from_tag:: ypyiu caller_contact:: sip:0213115...@194.38.160.113:5070 caller_cseq:: 604 caller_route_set:: caller_bind_addr:: udp:212.147.46.91:5060 to_uri:: sip:0213115...@212.147.46.91 to_tag:: as6c579ae5 callee_contact:: sip:0213115...@212.147.46.81:5060 callee_cseq:: 604 callee_route_set:: callee_bind_addr:: udp:212.147.46.91:5060 Here is my opensips.log file: May 11 10:34:12 vp-pro-01 opensips[3265]: DBG:dialog:build_new_dlg: new dialog 0x2b2cc4ab4b58 (c=jjolrfphvjri...@leonhart.interne.smart-telecom.ch ,f=sip:0213115097.46.91,t=5...@212.147.46.91,ft=ypyiu) on hash 3102 May 11 10:34:12 vp-pro-01 opensips[3265]: DBG:dialog:populate_leg_info: route_set , contact sip:0213115...@194.38.160.113:5070, cseq 603 and bind_ addr udp:212.147060 May 11 10:34:12 vp-pro-01 opensips[3265]: DBG:dialog:dlg_set_leg_info: set leg 0 for 0x2b2cc4ab4b58: tag=ypyiu rr= ct=sip:0213115...@194.38.1 60.113:5070 cse May 11 10:34:12 vp-pro-01 opensips[3265]: DBG:dialog:link_dlg: ref dlg 0x2b2cc4ab4b58 with 3 - 3 May 11 10:34:12 vp-pro-01 opensips[3265]: DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout May 11 10:34:12 vp-pro-01 opensips[3265]: DBG:dialog:unref_dlg: unref dlg 0x2b2cc4ab4b58 with 1 - 2 May 11 10:34:12 vp-pro-01 opensips[3263]: DBG:dialog:next_state_dlg: dialog 0x2b2cc4ab4b58 changed from state 1 to state 5, due event 4 May 11 10:34:12 vp-pro-01 opensips[3263]: DBG:dialog:dlg_onreply: dialog 0x2b2cc4ab4b58 failed (negative reply) May 11 10:34:12 vp-pro-01 opensips[3263]: DBG:dialog:unref_dlg: unref dlg 0x2b2cc4ab4b58 with 1 - 1 May 11 10:34:12 vp-pro-01 opensips[3270]: DBG:dialog:build_new_dlg: new dialog 0x2b2cc4ab52d8 (c=jjolrfphvjri...@leonhart.interne.smart-telecom.ch ,f=sip:0213115097.46.91,t=5...@212.147.46.91,ft=ypyiu) on hash 3102 May 11 10:34:12 vp-pro-01 opensips[3270]: DBG:dialog:populate_leg_info: route_set , contact sip:0213115...@194.38.160.113:5070, cseq 604 and bind_ addr udp:212.147060 May 11 10:34:12 vp-pro-01 opensips[3270]: DBG:dialog:dlg_set_leg_info: set leg 0 for 0x2b2cc4ab52d8: tag=ypyiu rr= ct=sip:0213115...@194.38.1 60.113:5070 cse May 11 10:34:12 vp-pro-01 opensips[3270]: DBG:dialog:link_dlg: ref dlg 0x2b2cc4ab52d8 with 3 - 3 May 11 10:34:12 vp-pro-01 opensips[3270]: DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout May 11 10:34:12 vp-pro-01 opensips[3270]: DBG:dialog:unref_dlg: unref dlg 0x2b2cc4ab52d8 with 1 - 2 May 11 10:34:12 vp-pro-01 opensips[3269]: DBG:dialog:next_state_dlg: dialog 0x2b2cc4ab52d8 changed from state 1 to state 3, due event 3 May 11 10:34:12 vp-pro-01 opensips[3269]: DBG:dialog:dlg_onreply: dialog 0x2b2cc4ab52d8 confirmed May 11 10:34:12 vp-pro-01 opensips[3269]: DBG:dialog:dlg_onreply: 0x2b2cc4ab52d8 totag in rpl is as6c579ae5 (10) May 11 10:34:12 vp-pro-01 opensips[3269]: DBG:dialog:populate_leg_info: route_set , contact sip:0213115...@212.147.46.81:5060, cseq 604 and bind_a ddr udp:212.147.60 May 11 10:34:12 vp-pro-01 opensips[3269]: DBG:dialog:dlg_set_leg_info: set leg 1 for 0x2b2cc4ab52d8: tag=as6c579ae5 rr= ct=sip:0213115...@212 .147.46.81:506004 May 11 10:34:12 vp-pro-01 opensips[3269]: DBG:dialog:insert_dlg_timer_unsafe: inserting 0x2b2cc4ab5310 for 10915 May 11 10:34:12 vp-pro-01 opensips[3269]: DBG:dialog:ref_dlg: ref dlg 0x2b2cc4ab52d8 with 1 - 3 May 11 10:34:17 vp-pro-01 opensips[3280]: DBG:dialog:next_state_dlg: dialog 0x2b2cc4ab4b58 changed from state 5 to state 5, due event 1 May 11 10:34:17 vp-pro-01 opensips[3280]: DBG:dialog:unref_dlg: unref dlg 0x2b2cc4ab4b58 with 1 - 0 May 11 10:34:17 vp-pro-01 opensips[3280]: DBG:dialog:unref_dlg: ref =0 for dialog 0x2b2cc4ab4b58 May 11 10:34:17 vp-pro-01 opensips[3280]: DBG:dialog:destroy_dlg: destroing dialog 0x2b2cc4ab4b58 May 11 10:34:17 vp-pro-01 opensips[3280]: DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x2b2cc4ab4b58 [3102:436706855] with clid 'jjol rfphvjri...@leonerne.smarth' and tags 'ypyiu' '' May 11 10:34:17 vp-pro-01 opensips[3280]: DBG:dialog:next_state_dlg: unref dlg 0x2b2cc4ab52d8 with 1 - 2 May 11 10:34:17 vp-pro-01 opensips[3280]: DBG:dialog:next_state_dlg: dialog 0x2b2cc4ab52d8 changed from state 3 to state 3, due event 1 you can see
[OpenSIPS-Users] CANCEL with a To tag.
Hi,
I would like to ask what would be the best way to handle CANCEL
request with a To tag. I know such a CANCEL request is not RFC
compatible, but unfortunately I came across some buggy UAs doing this.
What I did is:
if (has_totag())
{
/**
* sequential request withing a dialog should
* take the path determined by record-routing
*/
if (loose_route())
{
route(T_RELAY);
}
else
{
if (is_method(SUBSCRIBE) is_uri_host_local())
{
/* in-dialog subscribe requests */
route(PRESENCE);
}
if (is_method(ACK))
{
if (t_check_trans())
{
/**
* non loose-route, but stateful ACK;
* must be an ACK after a 487 or e.g. 404 from upstream server
*/
t_relay();
exit;
}
else
{
/* ACK without matching transaction ... ignore and discard. */
exit;
}
}
if (is_method(CANCEL))
{
t_relay();
exit;
}
sl_send_reply(404,Not here);
exit;
}
}
Would the above be OK? Or is it any better way of handling CANCEL with a To tag?
Any suggestions very much appreciated.
Regards,
Chris
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Re: [OpenSIPS-Users] CANCEL with a To tag.
2009/5/11 Chris Maciejewski ch...@wima.co.uk: Hi, I would like to ask what would be the best way to handle CANCEL request with a To tag. I know such a CANCEL request is not RFC compatible CANCEL is hop-by-hop. This means that when OpenSIPS receives a CANCEL, it *doesn't* route it, but it generates a new one (this occurs when you do t_relay() for a CANCEL). It's impossible to add To tag to a CANCEL generated by OpenSIPS (expect if the CANCEL occurs for a re-INVITE being into an already established dialog, so arriving CANCEL has To tag and OpenSIPS routes it as any other in-dialog request). but unfortunately I came across some buggy UAs doing this. What do you mean with it? what does this UAS? -- 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] Question about dialog max timeout...
Hi Marc, Marc Leurent wrote: Good morning everybody, Thank you for the information!! So my first problem is that dialogs are built even if I do not call the create_dialog() function or the setflag(4) that I have defined in dialog params.. to solve this mystery...can you grab a full debug log (debug=6) from your opensips ? I will try to see what is triggering the dialog creation - if no issue you can send me offlist the script to be able to correlate with the logs. To answer your question, yes dialogs are being removed, but some of them are not, I think because of reasons below: Each time I REGISTER: If Contact header is used to match the dialog, if I rewrite it because of routing specific architecture in fix_nated_contact();, dialogs won't be removed even if I use modparam(dialog, dlg_match_mode, 0) ?? Because dialogs are still there after the 200 OK like this dialog for REGISTER: dialog:: hash=3917:871117747 state:: 3 user_flags:: 0 timestart:: 1242029446 timeout:: 11025 callid:: jjolrfphvjri...@leonhart.interne from_uri:: sip:0213115...@212.147.46.91 from_tag:: pnvkk caller_contact:: sip:02131...@194.38.160.113:5070 caller_cseq:: 554 caller_route_set:: caller_bind_addr:: udp:212.147.46.91:5060 to_uri:: sip:0213115...@212.147.46.91 to_tag:: as54366440 callee_contact:: sip:02131...@212.147.46.81:5060 callee_cseq:: 554 callee_route_set:: callee_bind_addr:: udp:212.147.46.91:5060 Each time I unregister,I got an error: ERROR:dialog:populate_leg_info: bad sip message or missing Contact hdr Indeed, my Asterisk servers are responding without a Contact header to unregister, keeping dialog data until timeout and I thought dialog module didn't need it for matching the dialog! You should not use dialog module for register - the dialog module works only for INVITE ! REgards, Bogdan Thank you ++ -- -- Marc LEURENT lf...@leurent.eu Le Saturday 09 May 2009 10.59:18 Bogdan-Andrei Iancu, vous avez écrit : Hi Marc, It is normal to increase as it is an absolute timestamp - see my prev email on this thread. Regarding the shm mem - do you see the dialogs actually being removed? Regards, Bogdan Marc Leurent wrote: Sorry for the 3rd email.. but I think it's interesting, because the dialog timeout of my transactions seems to increase with OpenSIPs uptime! So my shm memory increase all the time. Does any one have the same problem with OpenSIPs 1.5.1? Thanks opensipsctl fifo dlg_list | grep timeout | sort -nr timeout:: 20395 timeout:: 20340 timeout:: 20285 timeout:: 20230 timeout:: 20175 timeout:: 20120 timeout:: 20064 timeout:: 20009 timeout:: 19954 timeout:: 19899 ... and just after opensipsctl fifo dlg_list | grep timeout | sort -nr timeout:: 20450 timeout:: 20395 timeout:: 20340 timeout:: 20285 timeout:: 20230 timeout:: 20175 timeout:: 20120 timeout:: 20064 timeout:: 20009 timeout:: 19954 timeout:: 19899 ... PS: I'm using opensips -V version: opensips 1.5.1-notls (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 5469 2009-03-18 12:43:10Z bogdan_iancu $ main.c compiled on 19:03:00 Apr 30 2009 with gcc 4.1.2 Le Friday 08 May 2009 13.06:18 Marc Leurent, vous avez écrit : Hello all, I have juste a small question about the dialog module, I have set the parameters below for OpenSIPs dialog module with a default_timeout, to 3 hours, # - dialog params - modparam(dialog, enable_stats, 1) # If the statistics support should be enabled or not. #modparam(dialog, hash_size, 4096) # The size of the hash table internally used to keep the dialogs. modparam(dialog, rr_param, did) # Name of the Record-Route parameter to be added with the dialog cookie. modparam(dialog, dlg_flag, 4) # Flag to be used for marking if a dialog should be constructed for the current request. #modparam(dialog, buy_on_timeout_flag, 6) # Message falg to be set if you want the dialog module to automatically send BYE requests (in both directions) when the dialog give timeout. #modparam(dialog, timeout_avp, $avp(i:10))# The specification of an AVP to contain a custom timeout (in seconds) for the dialog. modparam(dialog,
Re: [OpenSIPS-Users] Help with MediaProxy running under OpenVZ...
On 08 May 2009, at 20:14, Totally Useless wrote: I'm not sure if this is possible, maybe that's why I am having such a hard time with it.. I am trying to run MediaProxy on Debian 5.0 in an OpenVZ container. On my OpenVZ controller, I added iptable_nat to the IPTABLES list and /etc/init.d/vz restarted When trying to start MediaProxy, I get: May 8 17:59:23 mediaproxy01 media-relay[472]: Starting MediaProxy Relay 2.3.4 May 8 17:59:23 mediaproxy01 media-relay[472]: Set resource limit for maximum open file descriptors to 11000 May 8 17:59:23 mediaproxy01 media-relay[472]: fatal error: failed to create MediaProxy Relay: No such file or directory May 8 17:59:23 mediaproxy01 media-relay[472]: Traceback (most recent call last): May 8 17:59:23 mediaproxy01 media-relay[472]: --- exception caught here --- May 8 17:59:23 mediaproxy01 media-relay[472]: File /usr/bin/media-relay, line 58, in module May 8 17:59:23 mediaproxy01 media-relay[472]: relay = MediaRelay() May 8 17:59:23 mediaproxy01 media-relay[472]: File /var/lib/python-support/python2.5/mediaproxy/relay.py, line 340, in __init__ May 8 17:59:23 mediaproxy01 media-relay[472]: self.session_manager = SessionManager(self, Config.port_range.start, Config.port_range.end) May 8 17:59:23 mediaproxy01 media-relay[472]: File /var/lib/python-support/python2.5/mediaproxy/mediacontrol.py, line 601, in __init__ May 8 17:59:23 mediaproxy01 media-relay[472]: self.watcher = _conntrack.ExpireWatcher() May 8 17:59:23 mediaproxy01 media-relay[472]: mediaproxy.interfaces.system._conntrack.Error: No such file or directory Any idea what this is caused from? Not much help on Google for that error... Indeed, it looks like it has a problem opening the netfilter-conntrack socket. Could you try using the userspace conntrack tool, for example performing sudo conntrack -L to list the connection tracking entries? Perhaps some modules are not installed automatically. Also, I'm not entirely sure if you should be running the relay inside a virtual machine. For testing purposes it should be fine, but realtime media traffic probably does not flow well through a virtualized host. Ruud Klaver AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ACK timout OpenSIPS MediaProxy Integration using MySQL db
Hi, On 09 May 2009, at 04:59, Khan wrote: Sorry forgot the attachment of WS... Please Help... On Fri, May 8, 2009 at 9:56 PM, Khan khansfri...@gmail.com wrote: Hi everyone, I am having trouble running OpenSIPS, while back I had same problem, this problem is surfaced back again, I am sure there is something I am missing in my configuration. Currently i am trying to do followings: UAC1 MyOpenSIPS Server MySQL auth MediaProxy UAC2 At this point, I don't have anything else running except OpenSIPS, MySQL, and MediaProxy. My connections are such... ISP Modem -- *MyRouter --- MyServerBox * MyRouter has open ports, 80,22,5060, 1-13000, 5-6 My Goals is to Integrate OpenSIPS. MySQL, and MediaProxy and make it functional... I started mediaproxy as follows: root# media-dispatcher --no-fork Starting MediaProxy Dispatcher 2.3.4 Twisted is using epollreactor mediaproxy.dispatcher.RelayFactory starting on 50100 mediaproxy.dispatcher.OpenSIPSControlFactory starting on '/var/run/mediaproxy/dispatcher.sock' mediaproxy.dispatcher.ManagementControlFactory starting on 25061 Problem is that MediaProxy is not working with my configuration, I don't know what is going on. I can see both MediaProxy and OpenSIPS is running on my server BUT UAC outside my network still giving me the same problem as in the beginning. I constantly receive OPTIONS/ SUBSCRIBE from softphone outside my network since it doesn't received ACK, thus generate error... I made a call which lasted 35 seconds and got cut off giving UAC of other party an error of network failure. I have produced WS trace, please look at it and guide me what is wrong with this situation, also my configuration is on the following link as of today... http://pastebin.com/m3cf2769e Thanks for all your help, -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... -- Khan VoIP Rookie Every beginning has an end regardless we believe it or not... 0507WStrace.pcap I don't see how this has anything to do with mediaproxy, but you should be very suspicious about the fact that both the 180 and the 200 OK in response to the INVITE contain a Record-Route with a private IP address (192.168.1.9), to which the ACK is subsequently sent. Ruud Klaver AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CANCEL with a To tag.
You can see a SIP flow before I added CANCEL to a lose routing
section of my Opensips config here:
http://wima.co.uk/sip/2009-05-11_10-18-39-test-call_index.html
Note: F23 is rejected by OpenSIPs as it got tag in a To: header.
And after I added:
if (is_method(CANCEL))
{
t_relay();
exit;
}
to my lose routing logic, OpenSIPs generates CANCEL and sends it to
the next hop:
http://wima.co.uk/sip/2009-05-11_10-46-46-test-call_index.html
2009/5/11 Iñaki Baz Castillo i...@aliax.net:
2009/5/11 Chris Maciejewski ch...@wima.co.uk:
Hi,
I would like to ask what would be the best way to handle CANCEL
request with a To tag. I know such a CANCEL request is not RFC
compatible
CANCEL is hop-by-hop. This means that when OpenSIPS receives a CANCEL,
it *doesn't* route it, but it generates a new one (this occurs when
you do t_relay() for a CANCEL).
It's impossible to add To tag to a CANCEL generated by OpenSIPS
(expect if the CANCEL occurs for a re-INVITE being into an already
established dialog, so arriving CANCEL has To tag and OpenSIPS routes
it as any other in-dialog request).
but unfortunately I came across some buggy UAs doing this.
What do you mean with it? what does this UAS?
--
Iñaki Baz Castillo
i...@aliax.net
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Re: [OpenSIPS-Users] CANCEL with a To tag.
2009/5/11 Chris Maciejewski ch...@wima.co.uk:
You can see a SIP flow before I added CANCEL to a lose routing
section of my Opensips config here:
http://wima.co.uk/sip/2009-05-11_10-18-39-test-call_index.html
Note: F23 is rejected by OpenSIPs as it got tag in a To: header.
As you can see, that CANCEL has To tag but no Route so it is
correctly rejected:
CANCEL sip:somepst...@sip.domain.com SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP {UA_SafeCom_IP}:5060;branch=z9hG4bK4d054e
To: sip:somepst...@sip.domain.com;tag=as7fa158c0
From: John Smith sip:10...@sip.domain.com;tag=1294d4
Call-ID: 91b140a4-26ca446b-56590615-dc8b1...@{ua_safecom_ip}
CSeq: 1879 CANCEL
User-Agent: IP SIP Phone/2.0.6
Max-Forwards: 70
Contact: sip:10...@{ua_safecom_ip}:5060
And after I added:
if (is_method(CANCEL))
{
t_relay();
exit;
}
to my lose routing logic, OpenSIPs generates CANCEL and sends it to
the next hop:
http://wima.co.uk/sip/2009-05-11_10-46-46-test-call_index.html
Yes, it's a correct workaround to allow CANCEL from that broken UA.
However, note that, even if CANCEL F23 has To tag, the CANCEL
generated by OpenSIPS doesn't it (F25).
--
Iñaki Baz Castillo
i...@aliax.net
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Re: [OpenSIPS-Users] CANCEL with a To tag.
It sounds like the CANCEL with the To-tag should have a Route header as
well in order for it to be processed like any other sequential/in-dialog
request -- that is to say, under loose_route().
Or, the CANCEL is intended for OpenSIPS itself, in which case it should
not have a To-tag.
I would not try to accommodate this broken UA if I were you. When
breakage is so fundamental, this way lies madness.
Chris Maciejewski wrote:
You can see a SIP flow before I added CANCEL to a lose routing
section of my Opensips config here:
http://wima.co.uk/sip/2009-05-11_10-18-39-test-call_index.html
Note: F23 is rejected by OpenSIPs as it got tag in a To: header.
And after I added:
if (is_method(CANCEL))
{
t_relay();
exit;
}
to my lose routing logic, OpenSIPs generates CANCEL and sends it to
the next hop:
http://wima.co.uk/sip/2009-05-11_10-46-46-test-call_index.html
2009/5/11 Iñaki Baz Castillo i...@aliax.net:
2009/5/11 Chris Maciejewski ch...@wima.co.uk:
Hi,
I would like to ask what would be the best way to handle CANCEL
request with a To tag. I know such a CANCEL request is not RFC
compatible
CANCEL is hop-by-hop. This means that when OpenSIPS receives a CANCEL,
it *doesn't* route it, but it generates a new one (this occurs when
you do t_relay() for a CANCEL).
It's impossible to add To tag to a CANCEL generated by OpenSIPS
(expect if the CANCEL occurs for a re-INVITE being into an already
established dialog, so arriving CANCEL has To tag and OpenSIPS routes
it as any other in-dialog request).
but unfortunately I came across some buggy UAs doing this.
What do you mean with it? what does this UAS?
--
Iñaki Baz Castillo
i...@aliax.net
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--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [OpenSIPS-Users] CANCEL with a To tag.
2009/5/11 Alex Balashov abalas...@evaristesys.com: It sounds like the CANCEL with the To-tag should have a Route header as well in order for it to be processed like any other sequential/in-dialog request -- that is to say, under loose_route(). But it would be incorrect anyway. A CANCEL for an initial-INVITE shouldn't have To tag since the CANCEL must end the whole UAC transaction, not just an early-dialog. Or, the CANCEL is intended for OpenSIPS itself, in which case it should not have a To-tag. The CANCEL is always for OpenSIPS since CANCEL is hop by hop. I would not try to accommodate this broken UA if I were you. When breakage is so fundamental, this way lies madness. I agree. -- 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] CDRTool - MaxSessionTime and Debit balance cannot longer find the account info
Adrian, I have finally got the time to troubleshoot further my issue and found the problem. I was using as my cdr_source for the RatingEngine a CDRS_cisco like source, therefore I think this was the one messing up some things when starting up the RatingEngine. I don't expect you supporting cisco style cdrs for prepaid accounts, so I will define the opensips datasource just for the sake of prepaid accounts to work proper. Again, thanks for the time looking on my issue. Cheers, DanB On Tue, 2009-05-05 at 12:57 +0200, Adrian Georgescu wrote: Dan, I do not know what is wrong with your setup. I did the same and for me it works just fine: addbalance from=...@mydomain.com value=10.00 1 getbalance from=...@mydomain.com 10. MaxSessionTime callid=6432622...@1 From=sip:d...@mydomain.com To=sip:00497751800...@mydomain.com Duration=7200 Gateway=10.0.0.1 7200 May 5 12:54:34 ws1 cdrtool[17327]: MaxSessionTime callid=6432622...@1 From=sip:d...@mydomain.com To=sip:00497751800...@mydomain.com Duration=7200 Gateway=10.0.0.1 May 5 12:54:34 ws1 cdrtool[17327]: MaxSessionTime=7200 Type=prepaid callid=6432622...@1 billingparty=...@mydomain.com DestId=49 Balance=10. Spans=1 Regards, Adrian On May 1, 2009, at 3:51 PM, Dan-Cristian Bogos wrote: Guys, I did a CDRTool upgrade to the latest 6.7.8 and it looks like there is something wrong with the account matching. Both MaxSessionTime and DebitBalance are failing (in mysql it looks like the account is queried as empty string). Let me know if the same scenario works for you. Ta, DanB Log from CDRTool console: version CDRTool version 6.7.8 addbalance from=...@mydomain.com value=10.00 1 getbalance from=...@mydomain.com 20. MaxSessionTime callid=6432622...@1 From=sip:d...@mydomain.com To=sip:00497751800...@mydomain.com Duration=7200 Gateway=10.0.0.1 Lock=1 none Log in syslog: May 1 15:28:51 framdsrv01 cdrtool[5908]: addbalance from=...@mydomain.com value=10.00 May 1 15:28:51 framdsrv01 cdrtool[5908]: Prepaid account d...@mydomain.com credited with 10.00 May 1 15:29:06 framdsrv01 cdrtool[5908]: getbalance from=...@mydomain.com May 1 15:29:31 framdsrv01 cdrtool[5908]: MaxSessionTime callid=6432622...@1 From=sip:d...@mydomain.com To=sip:00497751800...@mydomain.com Duration=7200 Gateway=10.0.0.1 Lock=1 May 1 15:29:31 framdsrv01 cdrtool[5908]: MaxSessionTime=unlimited Type=postpaid callid=6432622...@1 BillingParty= ___ 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] CANCEL with a To tag.
Iñaki Baz Castillo wrote: 2009/5/11 Alex Balashov abalas...@evaristesys.com: It sounds like the CANCEL with the To-tag should have a Route header as well in order for it to be processed like any other sequential/in-dialog request -- that is to say, under loose_route(). But it would be incorrect anyway. A CANCEL for an initial-INVITE shouldn't have To tag since the CANCEL must end the whole UAC transaction, not just an early-dialog. Agreed, but I think the more harmless approach would be for the To tag issue to be ignored by the proxy and passed to the receiving UA to deal with. Or, the CANCEL is intended for OpenSIPS itself, in which case it should not have a To-tag. The CANCEL is always for OpenSIPS since CANCEL is hop by hop. Well, true. I meant a stateless vs. stateful CANCEL -- which also changes the domain destination of the RURI. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CANCEL with a To tag.
Alex Balashov wrote: Iñaki Baz Castillo wrote: 2009/5/11 Alex Balashov abalas...@evaristesys.com: It sounds like the CANCEL with the To-tag should have a Route header as well in order for it to be processed like any other sequential/in-dialog request -- that is to say, under loose_route(). But it would be incorrect anyway. A CANCEL for an initial-INVITE shouldn't have To tag since the CANCEL must end the whole UAC transaction, not just an early-dialog. Agreed, but I think the more harmless approach would be for the To tag issue to be ignored by the proxy and passed to the receiving UA to deal with. Although, since the has_totag() check is done first and loose_route() second in stock configs from which people derive theirs, I guess that really wouldn't work... -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CANCEL with a To tag.
I also agree with Alex.
I've tried to accommodate broken UAs in the past and I'll tell you, you're
just seeing a part of what's broken. If you allow for this, what else are
you going to find out down the road that isn't going to work.
As they say, it's the Tip, of the iceburg
-Brett
On Mon, May 11, 2009 at 6:40 AM, Alex Balashov abalas...@evaristesys.comwrote:
It sounds like the CANCEL with the To-tag should have a Route header as
well in order for it to be processed like any other sequential/in-dialog
request -- that is to say, under loose_route().
Or, the CANCEL is intended for OpenSIPS itself, in which case it should
not have a To-tag.
I would not try to accommodate this broken UA if I were you. When
breakage is so fundamental, this way lies madness.
Chris Maciejewski wrote:
You can see a SIP flow before I added CANCEL to a lose routing
section of my Opensips config here:
http://wima.co.uk/sip/2009-05-11_10-18-39-test-call_index.html
Note: F23 is rejected by OpenSIPs as it got tag in a To: header.
And after I added:
if (is_method(CANCEL))
{
t_relay();
exit;
}
to my lose routing logic, OpenSIPs generates CANCEL and sends it to
the next hop:
http://wima.co.uk/sip/2009-05-11_10-46-46-test-call_index.html
2009/5/11 Iñaki Baz Castillo i...@aliax.net:
2009/5/11 Chris Maciejewski ch...@wima.co.uk:
Hi,
I would like to ask what would be the best way to handle CANCEL
request with a To tag. I know such a CANCEL request is not RFC
compatible
CANCEL is hop-by-hop. This means that when OpenSIPS receives a CANCEL,
it *doesn't* route it, but it generates a new one (this occurs when
you do t_relay() for a CANCEL).
It's impossible to add To tag to a CANCEL generated by OpenSIPS
(expect if the CANCEL occurs for a re-INVITE being into an already
established dialog, so arriving CANCEL has To tag and OpenSIPS routes
it as any other in-dialog request).
but unfortunately I came across some buggy UAs doing this.
What do you mean with it? what does this UAS?
--
Iñaki Baz Castillo
i...@aliax.net
___
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--
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Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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[OpenSIPS-Users] opensips 1.5.1 - how to update the MySQL-DB if IP-address of UA changes
Dear sirs and madams, I'm a student from germany and currently I'm trying to install a SIP-server. This is my first time working with SIP (in general) and opensips. Hopefully I won't embarass myself too much. So here's my problme: Is there a way to implement a routine to update the MySQL-DB faster if the IP-address of a client changes? To be more specific: In the first step of my project I'm using only 192.168.1.x as my domain. The Proxy/ Registar can be found at 192.168.1.60 and I'm using two PCs as UACs. I can register a client with the server, if he's at e.g. 192.168.1.200. If I change its IP-address (to e.g .180), the changes take quite some time to manifest. If I re-register manually (ekiga sopftphone), there are two entries in the location-table, which I certainly don't want. On the one hand, how do I prohibid a UAC from registering with two different IP-addresses using the same username/ password. And on the other hand how can I speed up the process to update the database when a IP-address change takes place?! I'd really appreciate your help and thank you in advance. Best regards, Simon Witte -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss für nur 17,95 Euro/mtl.!* http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips is crashing when i load mi_xmlrpc module
hi all, when i load the module mi_xmlrpc.so the opensips is stopping with the following log. *debian:/etc/opensips# /etc/init.d/opensips start Starting opensips: opensipsListening on udp: 127.0.0.1 [127.0.0.1]:5080 udp: 192.168.1.129 [192.168.1.129]:5080 tcp: 127.0.0.1 [127.0.0.1]:5080 tcp: 192.168.1.129 [192.168.1.129]:5080 Aliases: tcp: debian.local:5080 tcp: localhost:5080 udp: debian.local:5080 udp: localhost:5080 May 11 18:19:32 [3156] INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) May 11 18:19:32 [3158] NOTICE:core:main: version: opensips 1.5.0dev5-notls (i386/linux) May 11 18:19:32 [3158] INFO:core:main: using 128 Mb shared memory May 11 18:19:32 [3158] INFO:core:main: using 1 Mb private memory per process May 11 18:19:32 [3158] NOTICE:signaling:mod_init: initializing module ... May 11 18:19:32 [3158] INFO:sl:mod_init: Initializing StateLess engine May 11 18:19:32 [3158] INFO:maxfwd:mod_init: initializing... . debian:/etc/opensips# May 11 18:19:32 [3158] INFO:tm:mod_init: TM - initializing... May 11 18:19:32 [3158] INFO:usrloc:ul_init_locks: locks array size 512 May 11 18:19:32 [3158] INFO:registrar:mod_init: initializing... May 11 18:19:32 [3158] INFO:textops:mod_init: initializing... May 11 18:19:32 [3158] WARNING:permissions:parse_config_file: file not found: //etc/opensips/permissions.allow May 11 18:19:32 [3158] WARNING:permissions:mod_init: default allow file (//etc/opensips/permissions.allow) not found = empty rule set May 11 18:19:32 [3158] WARNING:permissions:parse_config_file: file not found: //etc/opensips/permissions.deny May 11 18:19:32 [3158] WARNING:permissions:mod_init: default deny file (//etc/opensips/permissions.deny) not found = empty rule set May 11 18:19:32 [3158] INFO:avpops:avpops_init: initializing... May 11 18:19:32 [3158] INFO:xlog:mod_init: initializing... May 11 18:19:32 [3158] INFO:dialog:mod_init: Dialog module - initializing May 11 18:19:32 [3158] INFO:acc:mod_init: initializing... May 11 18:19:32 [3158] INFO:siptrace:mod_init: initializing... May 11 18:19:32 [3158] INFO:auth:mod_init: initializing... May 11 18:19:32 [3158] INFO:auth_db:mod_init: initializing... May 11 18:19:32 [3158] INFO:avp_radius:mod_init: initializing... May 11 18:19:32 [3158] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb May 11 18:19:32 [3158] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb May 11 18:19:32 [3172] INFO:mi_datagram:datagram_process: a new child 0/3172 May 11 18:19:32 [3173] INFO:mi_datagram:datagram_process: a new child 1/3173 May 11 18:19:32 [3158] INFO:core:handle_sigs: child process 3171 exited normally, status=1 May 11 18:19:32 [3158] INFO:core:handle_sigs: terminating due to SIGCHLD May 11 18:19:32 [3167] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3162] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3164] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3161] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3174] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3168] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3170] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3160] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3166] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3165] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3159] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3163] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3176] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3177] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3172] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3173] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3175] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3179] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3180] INFO:core:sig_usr: signal 15 received May 11 18:19:32 [3178] INFO:core:sig_usr: signal 15 received* can anyone help. -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS] sample script that works with cdrtools, freeradius, nat, drouting
Thanks Jimmy, I have been looking for some sample script for ages at every possible place. So, far no luck but its really nice of you to post your script here. I always wondered with open source software what is the deal of posting your functional script (hiding your confidential info) but never had an answer. Finally I see a brave person posting script... Would you mind if i try your script and see if it works for me??? Thanks, On Mon, May 11, 2009 at 1:31 AM, Jinsong Hu jinsong...@hotmail.com wrote: Hi, There: It looks ag-projects is maintaining the cdrtools, media proxy. but I searched around and didn't find anywhere there is a script that supports all the needed feature: cdrtools, mediaproxy, nat_traversal, and drouting. so now I'm trying to be a little brave and post my script that includes all above. this script doesn't handle instance message, but only voice calls. can any body spot problems with this script ? The goal of the script is to let locally registered user to use gateway to make outgoing call, and receive incoming call. the numbering plan is for US. free radius should have good authenticaing and accounting for different messages, and some special DID are mapped to several numbers and routed to asterisk. Hopefully this script will be useful for a general VOIP carrier. I try to paste the document to be comment. Hopefully, by going through this exercise, we can get a good starting script for people to use as a model starting script. Jimmy ### # # $Id: opensips.cfg,v 1.13 2009/05/11 06:06:00 jinsong Exp $ # # OpenSIPS basic configuration script # by Anca Vamanu a...@voice-system.ro # # Please refer to the Core CookBook at http://www.opensips.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # #INVITE :Invites a user to a call #ACK : Acknowledgement is used to facilitate reliable message exchange for INVITEs. #BYE :Terminates a connection between users #CANCEL :Terminates a request, or search, for a user. It is used if a client sends an INVITE and then changes its decision to call the recipient. #OPTIONS :Solicits information about a server's capabilities. #REGISTER :Registers a user's current location #INFO :Used for mid-session signaling #MESSAGE : IMS send message #SUBSCRIBE : IMS presence subscribe message #PUBLISH: IMS publish message #1xx: Provisional -- request received, continuing to process the request; #2xx: Success -- the action was successfully received, understood, and accepted; #3xx: Redirection -- further action needs to be taken in order to complete the request; #4xx: Client Error -- the request contains bad syntax or cannot be fulfilled at this server; #5xx: Server Error -- the server failed to fulfill an apparently valid request; #6xx: Global Failure -- the request cannot be fulfilled at any server. #This function sets the value of the flag given as parameter to 1 (true). The value of the parameter must be an integer between 0 and 31. ### Global Parameters # debug=3 log_stderror=no 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 #disable dns to scale dns=no rev_dns=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no alias=machinename.somedomain.com /* 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=5060 /* 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:192.168.1.2:5060 ### Modules Section #set module path mpath=/usr/lib/opensips/modules/ /* uncomment next line for MySQL DB support */ loadmodule db_mysql.so loadmodule mi_fifo.so loadmodule sl.so loadmodule tm.so loadmodule rr.so loadmodule maxfwd.so loadmodule usrloc.so loadmodule signaling.so loadmodule registrar.so loadmodule textops.so loadmodule uri_db.so loadmodule uri.so loadmodule xlog.so loadmodule acc.so /* uncomment next lines for MySQL based authentication support NOTE: a
Re: [OpenSIPS-Users] howto set 2 route entries with one server
Hi Uwe, never tried it, but using both normal and preset RR should do the job...have you tried it ? Regards, Bogdan Uwe Kastens wrote: Hello, I need to set 2 record_route entries for the following setup: a)asterisk1 ... asteriskn b)opensips opensips UA I need to set a) record_route_preset(asteriskx:5100) b) record_route() Any options to do this? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] howto set 2 route entries with one server
Hi Bogdan, never tried it, but using both normal and preset RR should do the job...have you tried it ? Yes I tried it. It won't work. rr_mod.c has some checks inside, that prevent doing this. I will try to play around with it. 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] howto set 2 route entries with one server
yeahI remember the checks...they should be on only if you try to do twice normal RR...but try to disable it and see it is worksI say it should :) Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, never tried it, but using both normal and preset RR should do the job...have you tried it ? Yes I tried it. It won't work. rr_mod.c has some checks inside, that prevent doing this. I will try to play around with it. BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS] sample script that works with cdrtools, freeradius, nat, drouting
Go ahead. I am not sure how good it is. if you find problems , please let me know. I am sure there must be problems there. But since I am also new, so I can't spot them all. that is why I posted it. it will be nice that there are people spot the problem and tell me. I can continue to improve it. when it is good enough, maybe this can be put in opensips repository for everybody to share. Jinsong - Original Message - From: Khan khansfri...@gmail.com To: Jinsong Hu jinsong...@hotmail.com Cc: users@lists.opensips.org Sent: Monday, May 11, 2009 11:55 AM Subject: Re: [OpenSIPS-Users] [OpenSIPS] sample script that works with cdrtools, freeradius, nat, drouting Thanks Jimmy, I have been looking for some sample script for ages at every possible place. So, far no luck but its really nice of you to post your script here. I always wondered with open source software what is the deal of posting your functional script (hiding your confidential info) but never had an answer. Finally I see a brave person posting script... Would you mind if i try your script and see if it works for me??? Thanks, On Mon, May 11, 2009 at 1:31 AM, Jinsong Hu jinsong...@hotmail.com wrote: Hi, There: It looks ag-projects is maintaining the cdrtools, media proxy. but I searched around and didn't find anywhere there is a script that supports all the needed feature: cdrtools, mediaproxy, nat_traversal, and drouting. so now I'm trying to be a little brave and post my script that includes all above. this script doesn't handle instance message, but only voice calls. can any body spot problems with this script ? The goal of the script is to let locally registered user to use gateway to make outgoing call, and receive incoming call. the numbering plan is for US. free radius should have good authenticaing and accounting for different messages, and some special DID are mapped to several numbers and routed to asterisk. Hopefully this script will be useful for a general VOIP carrier. I try to paste the document to be comment. Hopefully, by going through this exercise, we can get a good starting script for people to use as a model starting script. Jimmy ### # # $Id: opensips.cfg,v 1.13 2009/05/11 06:06:00 jinsong Exp $ # # OpenSIPS basic configuration script # by Anca Vamanu a...@voice-system.ro # # Please refer to the Core CookBook at http://www.opensips.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # #INVITE :Invites a user to a call #ACK : Acknowledgement is used to facilitate reliable message exchange for INVITEs. #BYE :Terminates a connection between users #CANCEL :Terminates a request, or search, for a user. It is used if a client sends an INVITE and then changes its decision to call the recipient. #OPTIONS :Solicits information about a server's capabilities. #REGISTER :Registers a user's current location #INFO :Used for mid-session signaling #MESSAGE : IMS send message #SUBSCRIBE : IMS presence subscribe message #PUBLISH: IMS publish message #1xx: Provisional -- request received, continuing to process the request; #2xx: Success -- the action was successfully received, understood, and accepted; #3xx: Redirection -- further action needs to be taken in order to complete the request; #4xx: Client Error -- the request contains bad syntax or cannot be fulfilled at this server; #5xx: Server Error -- the server failed to fulfill an apparently valid request; #6xx: Global Failure -- the request cannot be fulfilled at any server. #This function sets the value of the flag given as parameter to 1 (true). The value of the parameter must be an integer between 0 and 31. ### Global Parameters # debug=3 log_stderror=no 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 #disable dns to scale dns=no rev_dns=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no alias=machinename.somedomain.com /* 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=5060 /* uncomment
[OpenSIPS-Users] fix_nated_register replacement
Hi, I noticed that in nathelper module, there is a fix_nated_register() method to fix the nat for register . however, in nat_traversal module, there is no such method. so if I migrate from nathelper to nat_traversal, what do I do to fix the nat and save the registration to usrloc ? Jimmy ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
