sr-users December 2006

sr-users@lists.kamailio.org

152 participants
257 discussions

[Serusers] Work with a new column in SER.subscriber

by inge

Hi all, I add a new column named "client" in subscriber table. Next, I want to do a test wich seems like that in SQL : "SELECT client FROM subscriber WHERE username = "actual calling number" => equivalent of avp_db_load I think ? Then, I think I must do a test on the previous values with an if and avp_check I think. But my problem is to do this both requests with SER syntax in ser.cfg. Someone can help me ? Thanks

17 years, 10 months

[Users] forking call to another user on same proxy

by Christian Schlatter

I try to fork a call to another user registered at the same proxy as the caller. I figured out that the easiest method would be to use the append_branch() function, something like route[0] { if (method=="REGISTER") { save("location"); exit; } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; } # if someone calls test2@domain, fork call to test4@domain # (test2 and test4 are both registered at domain) if (avp_check("$rU", "eq/test2")) { append_branch("sip:test4@127.0.0.1"); } if (!t_relay()) { sl_reply_error(); }; exit; } I know that this could also be achieved by adding permanent location records, but I'd like to load the forking targets from a DB in the future. So my first question would be if using append_branch("sip:test4@127.0.0.1"); is the correct way to do parallel forking to a local user. I'm asking because this method does not work for me as expected. It seems that the two UAC state machines running on the same proxy host that are involved in a forking call do somehow not work well together. Forking the INVITE to both test2 and test4 works well, but when I pick up test2 the call gets terminated after a few seconds. Further investigations have revealed that canceling the second call leg does not work correctly. There is a message flow diagram available at http://www.unc.edu/~cschlatt/openser.html . For some reason, the proxy process sending from 127.0.0.1 is retransmitting the 487 "Request Terminated" although it receives ACKs from the upstream proxy. And this causes the upstream proxy to send a 487 back to the caller (time 15.360), which then terminates the call established between caller and test2. And this only happens if I fork to the same proxy, if I use two proxy hosts everything works fine (??). Any advice is highly appreciated, Christian

17 years, 10 months

Re: [Users] LCR db_only mode problem

by Jayesh Nambiar

Hi Juha, Thanks for your reply. First prominent doubt that I have is; when I am using LCR in db_only mode why does openser try to load the table while starting up? The scenario is as follows: I had 50 rows in LCR table and started openser and it started properly and also routed the calls properly according to the values in lcr and gw table. Now I just appended more 40,000 rows to this table and again the routing worked properly as per my lcr table. This behaviour was correct as I was using query only mode. I did not need any restart or any fifo reload command. Only when I restarted my openser, the openser again died as it was trying load the lcr table while starting. Could this be a bug in lcr module as I am not sure whats happening. I tried increasing MAX_NO_OF_LCRS but even that did not help. Somebody please help if poosible. Thanks, w/regards, jayesh ----- Original Message ---- From: Juha Heinanen <jh(a)tutpro.com> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Tuesday, 26 December, 2006 10:58:01 PM Subject: [Users] LCR db_only mode problem X-VM-v5-Data: ([nil nil nil nil nil nil nil nil nil] the error messages you get are coming from this statement: if (lcr_dbf.query(dbh, NULL, 0, NULL, lcr_cols, 0, 4, 0, &res) < 0) { LOG(L_ERR, "lcr_reload_gws(): Failed to query lcr data\n"); lcr_dbf.close(dbh); return -1; } i don't know which kind of memory db queries reserve. if shared memory, you can start openser with larger -m parameter value. if pkg memory, you need to edit PKG_MEM_POOL_SIZE in config.h and recompile openser. with lots of lcr entries and no caching, performance may be bad. if you use caching then you need to increase value of #define MAX_NO_OF_LCRS 256 in lcr_mod.c and recompile lcr module. increasing MAX_NO_OF_LCRS means that more shared memory will be used so you may also need to start openser with larger -m parameter value. -- juha Send free SMS to your Friends on Mobile from your Yahoo! Messenger. Download Now! http://messenger.yahoo.com/download.php

17 years, 10 months

[Users] Forward to the "To" field

by Thomas Deillon

Hi all, Merry Xmas ! I'm just a small question if some people are still working :-) I try to make a load balancing solution with SER and Asterisk. I use the dispatcher module and I set on asterisks servers the default gw as SER. All requests go through the SER to the asterisks and on the SER I'm have a line like If source-ip = Asterisk source IP do ... And I want to do something like forward to the TO FIELD ... Do you know how can I do this ? Thanks a lot for your help, Thomas

17 years, 10 months

[Serusers] Ser, MediaProxy and Asterisk

by Gerardo Amaya

Hello all, I'm trying to use a SER server for the following: | | | | ISP #1 ISP #2 Private Link Private LINK NATed SIP SIP Clients 172.18.xxx.yyy 172.16.5.xx Clients | | | | | | | | | | | ---------------------------------------------------------------------------------------------------- | ---------------------------------------------------------------- SER with MEDIAPROXY ---------------------------------------------------------------- | | | ================ Asterisk (172.16.5.1) ================= | | PSTN I need asterisk to handle all the calls to the PSTN and event the SIP calls between them. I need to use the SER server as a tool to receive calls from any of the environments that I draw on top, and all communicating with Asterisk, even I also have a backup Asterisk box, that I can use in load balance with SER. I saw that there is a function "rewritehostport()" which I'm pretty sure it won't work since I will not keep the call in the SER server. I have made the NAT configuration of the Getting Started Guide, but I now that this is just part of what I need to build the complete solution. Can anyone please give me a hint in what is that I need to acomplish such task Thanks in advance Gerardo Amaya

17 years, 10 months

[Users] Issues with calls using openser.

by Shane Burrell

I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the far end is left off-hook. This worked before but I did modify my script to work with mediaproxy. Below is the wireshark decode of the sip messagining. Any help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway. |Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 | |22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@152.53.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060 | |(5060) ------------------> (5060) | | |22.031 | 100 Giving a try | |SIP Status | |(5060) <------------------ (5060) | | |22.031 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |22.040 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |22.042 | | 180 Ringing |SIP Status | | |(5060) <------------------ (5060) | |22.042 | 180 Ringing | |SIP Status | |(5060) <------------------ (5060) | | |25.244 | | 200 OK SDP ( telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |25.245 | 200 OK SDP ( telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |25.269 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |25.269 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060 | |(5060) ------------------> (5060) | | |25.270 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | | |25.291 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |25.291 | BYE | | |SIP Request | |(5060) ------------------> (5060) | | |25.293 | | BYE | |SIP Request | | |(5060) ------------------> (5060) | |25.326 | | 200 OK | |SIP Status | | |(5060) <------------------ (5060) | |25.327 | 200 OK | | |SIP Status | |(5060) <------------------ (5060) | | Shane

17 years, 10 months

[Users] several presence question

by Klaus Darilion

Hi! I read the docs of the presence module and have some question. Maybe someone can enlighten me ;-) 1. If I understand correctly, the pua module acts as a "UAC". Thus, it can be used to send PUBLISH or SUBSCRIBE. I wonder what is the usecase for sending SUBSCRIBE? Usually, a client which sends a SUBSCRIBE receives subsequent NOTIFY messages. How can these NOTIFY messages be handled? 2. If an external application sends PUBLISH via the uac_mi interface, how can be distinguished between a "new" PUBLISH or an "update" PUBLISH? For example if the MI interface returns the E-Tag from the 200 Ok response this could be used in the next request to group identify PUBLISH reqeusts which belong together. 3. There is a bug in the pua module docs: the db_url(str) parameter is described twice. 4. The following module documentations lack of default values for the module parameters: pua, pua_usrloc, presence (maybe some example show the default value, but IMO the default value should be explicitely stated) 5. How can I update the presence status from a third party application? Is it possible to write directly into the tables of the presence modules? Or can I fake an update using the pua_mi to loop a PUBLISH to itself for handling it afterwards with handle_publish()? May this cause problems because it interferes the subsequent PUBLISH request from the client (etag)? regards Klaus

17 years, 10 months

[Users] Problem registering the UA with openSER(tls enabled)

by Ncheeku Baranov

Hi, I am trying to make my non-TLS/TLS UA register with my TLS enabled openSER. Currently I am just working on my local machine with the client UAs on the same subnet,(so there is only one domain, but its not named). Below is my configuration file: disable_tls = 0 listen = tls:10.30.100.41:5061 tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user- cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user- privkey.pem" tls_ca_list = "usr/local/etc/openser/tls/user/user-calist.pem" However, with the above configuration the client UAs couldnot register and I got 408 Request Time out Message. Is there any field that is missing to make this simple scenario work? What should be the values of "tls_client_domain" and "tls_server_domain" fields in this case? I noticed that when I start the openSER without TLS support using "openserctl start" and do "ps -e" after that, there are more openSER processes running than if I start openSER with TLS support in which case I see very few of these processes running. Your help is much appreciated.... Best regards, NCheeku

17 years, 10 months

[Users] Accounting using dialog and sst module

by Asterisk Expert

Hello, I am currently trying to do accurate billing with openser. I found that dialog and sst modules can be used to determine failed transactions. i.e. If UA crashed and no BYE received at all. I tried to configure dialog and sst modules as specified in the documentation. But active dialogs keep incrementing even after hangup. It would be nice if someone can post a good example. I am using cvs head of openser. Snippet of config is as below: ###################### DIALOG ###################### modparam("dialog", "dlg_flag", 13) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "use_tight_match", 1) modparam("dialog", "enable_stats", 1) ###################### SST ###################### modparam("sst", "enable_stats", 1) modparam("sst", "timeout_avp", "$avp(i:10)") modparam("sst", "sst_flag", 14) route { ........ ........ if(method=="INVITE") { setflag(13); # set the dialog flag setflag(14); # Set the sst flag avp_write("15","$avp(i:10)"); }; if (method=="INVITE") { record_route_preset("192.168.1.2:5060;nat=yes;did=1"); }; ......... ......... } -- Regards Asterisk Expert

17 years, 10 months

[Users] Presence and Location

by Klaus Darilion

Hi! What kind of presence does the presence module support? Only the "basic" presence format PIDF (RFC 3863) or also PIDF extensions like the GEOPRIV Location Object Format RFC 4119? Or in other words - is the presence module aware of the XML payload or is it agnostic to the presence payload? regards klaus

17 years, 10 months

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sr-users December 2006