April 2008 - sr-users - kamailio.org

[OpenSER-Users] OpenSer closes TCP connection if receives wrong data
by Iñaki Baz Castillo 16 Apr '08

16 Apr '08

Hi, if OpenSer receives a request via TCP with wrong data (for example Via missing) it closed automatically the connection. I'm just experimenting with SIP TCP and I'd like to know the reason for this behaviour, I know that many SIP TCP UAS's have not this behaviour. I also know that other UAS's dissallow empty lines before the message (so disracrd the message) while OpenSer do allow them. Any explanation/recomendation is welcome. Are those recommendation for a SIP TCP stack written in some RFC? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] Openser problem
by Bogdan-Andrei Iancu 16 Apr '08

16 Apr '08

Hi Norberto, most sure the problem is in routing the CANCEL request. On this topic, please see: http://www.voice-sistem.ro/downloads/2007.08.29-Admin-Course/ There is detailed explanation about CANCEL routing. Regards, Bogdan Norberto Monteiro wrote: > > Hello, > > I’m having a problem that is the following: > > I’m using rewriteport to send traffic to another proxy. Then When I > make a call and if I hung up the other phone still rings… > > What could be the problem? > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] Openser and Portabilling100
by Jaas 16 Apr '08

16 Apr '08

Hi all, I would like to know if there could be same problems with OpenSER and PortaBilling100 because I have to choose a billing system for the ISP I work. I don't see any document with interaction between these two systems... Alternatively, can you give me hints about a good and complete billing system? Thank you in advance

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[OpenSER-Users] FreeSWITCH and OpenSER
by Mikael A. Bjerkeland 16 Apr '08

16 Apr '08

Hi, I am trying to look up information on the use of FreeSWITCH together with OpenSER. I am doing a configuration on my own without much experience on the configuration of OpenSER. In my setup OpensSER acts as a registrar and proxy. I have multiple FreeSWITCH servers for media, routing and PSTN termination, but I am not sure if my approach is the "right way". All SIP traffic goes to the OpenSER proxy which in turn routes the call with the LCR module to a FreeSWITCH server. This has proven to work fine in my limited testing environment. I am using ENUM on the FreeSWITCH server to look up dialed numbers. If the dialed number was found in our ENUM registry and belongs to the proxy, we call back to OpenSER: SIP UA (subscriber 56195122) -> INVITE 25929511 -> FreeSWITCH -> ENUM lookup -> Found -> Routed back to OpenSER -> SIP UA (subscriber (25929511) If the call is not found: SIP UA (subscriber 56195122) -> INVITE 95018615 -> FreeSWITCH -> ENUM lookup -> Not found -> Routed to PSTN gateway or SIP peer I am not using aliases in OpenSER as it seems that ENUM provides the same functionality, but please correct me if I'm wrong. I also made this decision because it seems easier to implement features like call forwarding (unconditional, busy, no answer and timed) in FreeSWITCH rather than in OpenSER. This brings me back to my current issue: Call transfer Outbound and inbound calling with ENUM lookups in FreeSWITCH to my test phones works fine. So does calls to the PSTN gateway. My problem has to do with transfers. I can do a an attended transfer from my Polycom phone when I'm in a call. The second call is set up and I can talk to the person I want to transfer to. I hit the transfer button. Nothing happens, and FS and OpenSER try to look up number_to_transfer_to@openser-registrar. The call is routed to OpenSER which complains that it cannot find the subscriber/alias, which makes sense as there isn't configured an alias. But this also happens when I transfer to a PSTN number. Do I need to take special actions on REFERs? Are there any examples of a working OpenSER + FreeSWITCH setup out there? I believe FreeSWITCH and OpenSER complement each other great but I found the lack of information on the two applications used together a showstopper. Has anyone implemented phone service based on OpenSER and FreeSWITCH who can explain their setup a bit? Regards, Mikael

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[OpenSER-Users] Question about installing openser
by Pete Kay 15 Apr '08

15 Apr '08

Hi all, I am trying to get Openser 1.3.1 installed on Debian. It gave me a lot of errors when I try to "make" from the source. Could someone please tell me the list of packages I should install before building Openser? By the way, I haven't been able to find a step-by-step guides on installing Openser from scratch on Debian. If there is one exists somewhere on the Internet, I would greatly appreciate it if you can point me to it. I am currently working through a few of the how-tos trying to put pieces together. Thanks in advance for your help. Regards, Pete

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[OpenSER-Users] Problem with redirection (302 Moved temporarly)
by Antonio Reale 15 Apr '08

15 Apr '08

Hi all, I have a Linksys SPA2102 with 2 phones configured with the same account/number. If I call that number both phones ring and all is OK. The problem is that if I set a call forwarding on one of the two line on the SPA only one phone rings. Openser receives the 302 message from the SPA but the failure route that contains t_check_status("302") is not called. The forwarding works if only one line is connected. How can I handle the 302 in this situation? Thanks in advance. Regards, Antonio.

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[OpenSER-Users] sipwise config error - save location?
by Robert McNaught 14 Apr '08

14 Apr '08

Please discard - fix for this was a curly bracket on line 313 - not sure if that was sipwise or me! Robert

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[OpenSER-Users] sipwise config error - save location?
by Robert McNaught 14 Apr '08

14 Apr '08

Hi, Has anyone ever seen this error before? I created a config file using the sipwise utility: [root@ser1 openser]# openser -c 0(30469) parse error (314,1-3): syntax error 0(30469) parse error (314,1-3): ERROR: bad config file (2 errors) 0(30469) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy [root@ser1 openser]# 314 if(!save("location")) 315 { 316 317 xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); 318 sl_reply_error(); 319 exit; 320 } It complains about line 314 having a syntax error - I have another openser machine running with the same exact line and it is ok? Has anyone seen this before? TIA Robert

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[OpenSER-Users] SIP Load Balanser for OpenSER
by Vladimir Romanov 14 Apr '08

14 Apr '08

Hi! I want test setup OpenSER in HA & performance cluster. For this I need SIP LoadBalancer. Anyone known were I can download vovida lbproxy? Vovida site not answer last time :( Maybe there exist other software SIP load balancers. -- Vladimir Romanov

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[OpenSER-Users] Avoid Loops between Openser and Asterisk
by Cosimo Fadda 13 Apr '08

13 Apr '08

Hi everybody, I'm using Opeser as Proxy and one Asterisk as Gateway and Voicemail server. Suppose that an incoming calls arrives to Ser from Asterisk and the user is offline; Then Openser sends back the Invite to Asterisk that should activate the voicemail application but unfortunately it detects a Loop (482) and rejects the invite. I think I have to mangle the SIP message in Openser before the send to Asterisk again. This is what I currently do: if(!lookup("location")) { #send to voicemail if active but not registerd xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if($avp(s:vmail) == "1") { route(11); } else { route(20); } } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(9); } exit; .... route[11] { xlog("L_INFO", "Forwarding request to VM\n"); prefix("vm"); sethostport("asterisk-gw:5060"); #append_branch(); if (!t_relay()) { xlog("L_INFO", "Unable TO Forward the request to VM\n"); route(20); exit; } What can I do to avoid Loop Detection? Thanks in advance, Cosimo Fadda ___________

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[Serusers] Asterisk with SER ...great solution
by satish patel 12 Apr '08

12 Apr '08

find solution on my website http://linuxbug.org/index_files/projects.html $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

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[OpenSER-Users] Openser problem
by Norberto Monteiro 11 Apr '08

11 Apr '08

Hello, I'm having a problem that is the following: I'm using rewriteport to send traffic to another proxy. Then When I make a call and if I hung up the other phone still rings. What could be the problem?

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[Serusers] Questions re. MySQL
by frq ser 11 Apr '08

11 Apr '08

Hi, Here's a set of basic questions for which I need your help. I want to play different announcements based on the caller's history. For simplicity, I want ann1_user1.wav to be played for the first call, and ann2_user1.wav for all subsequent calls from the 1st user. Similarly for 2nd user, the files would be ann*_user2.wav. 1) I want to have a basic record of the number of times a user has called the SER(ie, sent an INVITE). Which table and column in mySQL can I use for this purpose. Also, how can I write the integer using SER commands (ie something that I can put in ser.cfg file). There is an example ( http://www.iptel.org/write_custom_text_to_mysql_acc_table) which uses 'acc_db_request', but it can only write to sip_status column in acc table. How can I modify /write to to any general field in mysql? 2) Subsequently, how can I retreive this value from mysql through some command in ser.cfg file. I want to use this value in an 'if' statement and redirect the process flow accordingly. 3) I was thinking of rewriting the uri to make it 'ann1_user1@xxxxx' and sending the request to early_announce application so that the user hears different announcement based on username and their call history (ie whether this user has called before or not). However, I am not very clear about how to do this. Is there any simpler method to do this. The final aim of this whole exercise is to demonstrate call differentiation based on username and user history. So if someone has a better idea then please do share it. Thanks for the help and accept my apologies for squeezing so many questions into a single message. Regards, Frq

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[OpenSER-Users] My application destroys registration
by kriko 11 Apr '08

11 Apr '08

Hi! I have a problem with wesip, it works but it destroys client registration. I added simple if statement, which executes app on wesip server (only on INVITE): #app server conn if(!as_relay_t("examplesvr")){ t_reply("500","No App Server"); } If you look at wesip log: http://files.myopera.com/kriko/code/seas_log.txt You can see that wesip doesn't process this request, since the app only responds to INVITE, looking closer at this line: --- SIP/2.0 481 Call Leg/Transaction Does Not Exist --- If you would like to look at app it is here (really simple thing): http://files.myopera.com/kriko/code/sample.java What am I doing wrong? Is it because I'm using t_reply in wrong place or there is something else. I don't think it is my app, since it is not being executed. -- (\__/) (='.'=) (")_(")

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[OpenSER-Users] Openser And DISC Connection close
by Dilip 11 Apr '08

11 Apr '08

Hello Friends, When Openser want to connect with disc client at that time disc client close the connection and gives the folowing error. DEBUG:peer_state_machine: peer 0x2aaaaaaf33a8, state = 3, event=0 ERROR:peer_state_machine: event - state mismatch : peer=0x2aaaaaaf33a8, state=3, event=0 INFO:do_receive: timeout received Actually Openser and DISC can not ESTABLISH the connection and gives CLOSEWAIT in the netstat command. So what will be problem ,please guide me. Regards, Dilip

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[Serusers] Problem related to exec_msg
by frq ser 11 Apr '08

11 Apr '08

Hi, I am trying to understand the 'exec_msg' command as described in Section 4 of SER Admin Guide. I am just trying to echo the user name (1001) of the caller on the screen. I use the config file given at the end. On receiving the INVITE message, SER DOES echo the user name, but I also get some warning and error messages as shown below: #> /opt/ser-sems/sbin/ser -f /xxx.../voip/ser_expt.cfg Listening on udp: 192.168.3.208 [192.168.3.208]:5070 tcp: 192.168.3.208 [192.168.3.208]:5070 Aliases: tcp: xxx:5070 udp: xxx:5070 WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing exec - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 2803 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(2801) Main Route... 0(2801) Inside search from-1001... 1001 0(2801) child process 2805 exited normally, status=0 0(2801) INFO: dont_fork turned on, living on 0(2801) WARNING: SIGPIPE received and ignored 0(2801) ERROR: exec_msg: cmd echo '1001' failed. exit_status=-1, errno=10: No child processes 0(2801) WARNING: receive_msg: error while trying script Can someone tell me how I can get rid of these messages. My aim was just to get the user name displayed, and this has been achieved. I just want to understand the reason for these warnings and error messages. Thanks Frq PS: This is my config file: # $Id: hello-world.cfg 46 2006-01-25 04:21:30Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=3 fork=no log_stderror=yes listen=192.168.3.208 #2.13 # put your server IP address here port=5070 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" unix_sock="/tmp/ser_sock" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/opt/ser-sems/lib/ser/modules/mysql.so" loadmodule "/opt/ser-sems/lib/ser/modules/sl.so" loadmodule "/opt/ser-sems/lib/ser/modules/tm.so" loadmodule "/opt/ser-sems/lib/ser/modules/rr.so" loadmodule "/opt/ser-sems/lib/ser/modules/maxfwd.so" loadmodule "/opt/ser-sems/lib/ser/modules/usrloc.so" loadmodule "/opt/ser-sems/lib/ser/modules/registrar.so" loadmodule "/opt/ser-sems/lib/ser/modules/textops.so" loadmodule "/opt/ser-sems/lib/ser/modules/exec.so" route { log(1, "Main Route...\n"); # sl_send_reply("404", "User Not Found...:-("); # if (uri=~"^sip:200@") { # exec_msg("echo 'Frq'"); # } if (search("(f|From): .*1002@")) { log(3, "Inside search from-1002...\n"); exec_msg("echo '1002'"); }; if (search("(f|From): .*1000@")) { log(3, "Inside search from-1000...\n"); exec_msg("echo '1000'"); }; if (search("(f|From): .*1001@")) { log(3, "Inside search from-1001...\n"); exec_msg("echo 1001"); }; }

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[OpenSER-Users] Recommendations for IM/Voice/Video client
by Adrian Georgescu 11 Apr '08

11 Apr '08

For those interested, you can monitor the progress of development for the multi party chat functionality based on MSRP here: http://www.msrprelay.org/ http://www.msrprelay.org/wiki/MSRPChatSoftwareDesign Adrian >>>>>>>>>>>> I've tryed Bria but in its web page it appears: IM Management - Send instant messages to groups - Use IM tabs for multiple IM conversations in one window - See IM alerts for incoming IM It's not as a IM conferencing in a server side, but maybe enought... Anyway, you can do a conference with IMC module and create contacts like "IM room 1", "IM room 2" in each client. Of course, for now there is no way to know which users are joining a conference. There are two RFC's about it: - SIP Event Package for Conference State http://tools.ietf.org/html/rfc4575 - SIP Call Control - Conferencing for User Agents http://tools.ietf.org/html/rfc4579 but AFAIK there are not yet client side implementation. -- Iñaki Baz Castillo

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[OpenSER-Users] (no subject)
by Hamed al-Suhli 10 Apr '08

10 Apr '08

Hi I'm new to openser and also Linux server install and configuration, I'm profissional at windows and its network, but need to install server for openser on local small lan. also will be used as backup for data. what is the easiest server to install? -- hamed suhli http://sourceforge.net/projects/tarmeez

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[OpenSER-Users] Recommendations for IM/Voice/Video client
by Robert McGilvray 10 Apr '08

10 Apr '08

This is a little off-topic, but figured it was a good place to ask. We're interested in deploying OpenSER coupled with a SIP client primarily used for IM, but also for voice and video conferencing. I'm looking for feedback and recommendations on clients from people that may do this already or have been involved with it. So far one of the better ones looks like Bria (a step up from Eyebeam) from CounterPath. One short coming though is the lack of IM conferencing. I could probably rig something up with IMC but our users would prefer to have a clicky GUI for it. This is for an enterprise and it'd be nice to be able to brand it. Thanks -Bob

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[OpenSER-Users] forwarding responses to early in TM
by Taisto Qvist 10 Apr '08

10 Apr '08

Hi folks, When I am doing forking I am getting a puzzling result concerning forwarding of responses, and I'm wondering how script needs tweaking. When recieving a 6xx on one branch, the TM-module automatically generates CANCEL to cancel the remaining branches(I wouldnt mind if this was configurable), but the puzzling thing is that it forwards the 603 _before_ it has terminated the remaining client transactions. (It sends cancel but it doesnt wait for final responses of the invites its cancelling, just forwards 6xx directly.) A similar scenario happends when I let Timer C pop. My TM-module will send a final response on the server-side at the same time as it starts to cancel the client side. But since one of the client txns just might result in a 2xx(INV) I cant forward the final response until ALL client txns are done, if I am going to follow the rfc properly. I thought the TM module would keep track of all client txns for "me", but it seems it isnt, so is this something I should "manually" handle in my script? Regards Taisto Qvist

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[Serusers] Automatic end of call
by inge 10 Apr '08

10 Apr '08

Hi all, In find some informations about an automatic end of call system. The aim is to cut the call after 2 hours. So it's not specificly link to a pre-paid system, but for all my ser user, I want to end the call after 2 hours. Is there a existing solution to do that ? Or maybe some idea to deal with this project ? Thank you !

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[OpenSER-Users] Accounting Time on dialogue SIP
by Marcio 10 Apr '08

10 Apr '08

Hi... Anybody could me help, I need account time within dialogue SIP( from reply of the Invite -> OK until the BYE-> finish dialogue) and to save for a SGBD (MySQL - That's autenthication's responsible)....... Does anyone know how...? Any hints.....?? thx... Marcio

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[OpenSER-Users] Can't register on OpenSER using SIPp
by Vladimir Romanov 10 Apr '08

10 Apr '08

Hello All! I can't register in OpenSER using SIPp. SIPp versions tested 2.0.1 Windows (TLS), Latest from SVN on linux X-Lite registers w/o any problems. In pcap file & messages you can see 2 registration attempts. first form x-lite (successful) and second from SIPp. This looks like OpenSER and SIPp have some differences in MD5 authentication. -- Vladimir Romanov

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[OpenSER-Users] Can't register on OpenSER using SIPp
by Vladimir Romanov 10 Apr '08

10 Apr '08

I can't register in OpenSER using SIPp. SIPp versions tested 2.0.1 Windows (TLS), Latest from SVN on linux X-Lite registers w/o any problems. In pcap file & messages you can see 2 regsitration attempts. first form x-lite (sucessfull) and second from SIPp. This looks like OpenSER and SIPp have some differences in MD5 authentification. -- Vladimir Romanov

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[OpenSER-Users] Sip server tutorial
by kriko 10 Apr '08

10 Apr '08

Hi! I was assigned to explore OpenSer, however I'm more green on this stuff. Is there a good basic documentation/tutorial on how sip servers and systems around works (basic stuff like routing)? -- (\__/) (='.'=) (")_(")

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[OpenSER-Users] Wants to reopen auth_diameter module
by Dilip 10 Apr '08

10 Apr '08

Hello Friends, There is a auth_diameter module in the openser,But now no one is working on that. I want to make it compatible with the opendiameter. So that we can implement AAA protocol in the complete manner. For that i want your help for the completion of this module. Regards, Dilip

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[OpenSER-Users] survey: Who is using serweb?
by Klaus Darilion 10 Apr '08

10 Apr '08

Which version do you use? Does it work at all with current version of openser (1.3)? PS: this is to find out if serweb supported can be removed from openser regards klaus

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[Serusers] good news for German ser fans: book
by Jiri Kuthan 09 Apr '08

09 Apr '08

I got today this book on my desk, courtesy of one of its authors. https://www.opensourcepress.de/index.php?26&backPID=178&tt_products=194 Whereas I was busy with a bunch of unrelated things, the book was circulating in the iptel office all day and has found its keen readers. Unfortunately still only in German lanugage. -jiri

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[Serusers] select for canonical username (alias)?
by Stefan Sayer 09 Apr '08

09 Apr '08

Hello, is there a select to get the canonical username? or is 'canonical username' now a deprecated concept? I know there is the uid, but the problem is, with 36 characters uid is too long for my purpose (user id for voicebox). thanks Stefan -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[Serusers] about URI request
by liu york 09 Apr '08

09 Apr '08

hi i want to distribute tow alieas users use different pstn gw with same area code. i add below in my cfg.but it doesnt work.any advice for me ? many many thx. if (uri=~"^sip:0060.*@(my\.test\.com)"){ strip(2); prefix("21"); rewritehostport("1.2.3.4:5060"); # This is where PSTN Gateway installed route(1); return; }; if (uri=~"^sip:0060.*@(your\.test\.com)"){ strip(2); prefix("22"); rewritehostport("5.6.7.8:5060"); # This is where PSTN Gateway installed route(1); return; };

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[OpenSER-Users] Invalid cseq for aor - OpenSER stuck?
by Adrian A 09 Apr '08

09 Apr '08

I am running OpenSER 1.3.1 on a Debian 4.0 system. It seems that there's some sort of bug with OpenSER getting stuck and stops processing requests at least once a day. The logs only show the following line repeated every minute: ERROR:registrar:update_contacts: invalid cseq for aor <1302> At 1302 is a Cisco 7905 phone running the latest firmware. If I restart OpenSER then things get back to normal. Does anyone know what the problem might be?

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[Serusers] SER 0.9.7-pre3 and DNS SRV
by Steven C. Blair 09 Apr '08

09 Apr '08

I have a SER 0.9.7-pre3 system which is working fine except for call transfers. It turns out that when the NOTIFY is sent after the REFER method the proxy performs a queried for only A records not SRV. This results in an IP address being returned which is a non-SIP host. This is because the domain name is being used by different services some of which use A records for resolving the name and some, like SIP, are suppose to use SRV. How can I make this version of SER perform an SRV lookup instead of an A record lookup when initiating SIP messages? Thanks, Steve

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[OpenSER-Users] OpenSer and sip calls
by kriko 09 Apr '08

09 Apr '08

Hi! I'm trying to get openser working properly - it starts up, I can even regiter from softphone (linphone / twinkle) but I cannot make calls. OpenSer is running on 10.99.8.103:5060 (this is my machine, IP is bound to eth0), when I try to call to e.g.: sip:demovoice@10.99.8.48 I get some errors as in this log: http://files.myopera.com/kriko/code/openser.tar.gz Configuration is included too. Softphone clients are properly configured - it is the proxy that is causing me problems. Any kind of information is appreciated. -- kriko

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[OpenSER-Users] Thoughts on openSER + Asterisk scenarios
by Gabriel Ortiz Lour 09 Apr '08

09 Apr '08

Hi all, We are working here on a solution for a large scale VoIP system to cover our entire state, using both openSER and Asterisk. The project is on an initial stage, basically deciding architecture, so I would like to share our thoughts here in the mailing list and hopefully get some feedback to get starting on the right foot. We have our own private network (internet only on a further first step, no NAT), and we think about two possible scenarios: 1: The SIP accounts would register on the local Asterisk and those would talk to a SER core to connect them all, something like "Asterisk: Oh! this account is not mine so I'll send it to the core to take care for me" and the SER core would try to reach the other endpoint or return some error code. 2: All SIP accounts would be registered on the SER core, using the local asterisk servers as mediaservers and backup SIP proxies, where the SIP accounts would be registering on them only when there is some problem on the link with the SER core (and here they would have to have their own local PSTN gateways) This is the two basic ideas, so we would like to have any thoughts about the "pros and cons" of them. We have also planned many PSTN gateways that would be used for LCR (making all PSTN calls = local) for the entire system, that should be considered on a failure event. Thanks, Gabriel Ortiz PS.: Any help/thoughts are very welcome, as we will share all the experience and development on this project with the FOSS community. We are a government IT company from Paraná, Brasil aligned with the Free Software movement, and the aim is to document and package every aspect of this project.

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Re: [OpenSER-Users] [openser 1.3] loading module auth.so stop my server (bug?!)
by Pascal Maugeri 09 Apr '08

09 Apr '08

I'm still troubleshooting this issue and this morning I added more outputs in the source code of auth_mod.c. And I discovered: 1) when I set the configuration parameter "secret" of the module auth (modparam("auth", "secret", "secretphrase) ) in openser.cfg everything works fine ! 2) when I remove the configuration parameter "secret", the auth_mod.c calls the method generate_random_secret() as stated in the documentation. It enters in the loop for(i = 0; i < RAND_SECRET_LEN; i++) { LM_INFO("(3a)\n"); sec_rand[i] = 32 + (int)(95.0 * rand() / (RAND_MAX + 1.0)); } LM_INFO("(4)\n"); but never exit ! This is the reason why my server does not start and keeps blocked there. I modified the code above to print more logs, I can see in the logs the string (3a) once but I never (4): IMHO it is blocked somewhere in the first random computation. I'm running openser on Redhat Enterprise v4. Is there anything else I should configure in my environment ? Or I found a bug ? Regards, Pascal On Tue, Apr 1, 2008 at 12:45 PM, Pascal Maugeri <pascal.maugeri1(a)gmail.com> wrote: > Hi > > I'm still unable to have a working server 1.3 that loads the module > auth.so. > > It is weird because there is no error message in my log file also I set > the debug level to maximum. But after starting the openser service, it says > it is successful but there is no running process (pgrep openser does not > show anything, when it does if I remove the load auth.so)! > > How can I get more debugging info about the loading and initialization of > auth ? > > Can anybody share with me his or her configuration file ? I may miss > something in mine: > loadmodule "sl.so" > [...] > loadmodule "auth.so" > (I believe it is sufficient). > > Is there any known issue using Redhat Enterprise 4 ? > > Regards, > Pascal > > >

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Re: [OpenSER-Users] Specifying IP Address for SIP Responses
by Bogdan-Andrei Iancu 09 Apr '08

09 Apr '08

Hi Victor, if a UAC wants to have asymmetric signalling, just send from IP_A:PORT_A and put in top most via IP_B:PORT_B - via is used for routing back the reply. regards, Bogdan Victor Cartes wrote: > Thanks, Iñaki. But It is not what I really need. > > I need to send to an UAS a SIP Request, like OPTIONS, and tell it to > send the reply to a different transport address (IP Addr + Port) than > the source address of the request. > > This is, basically, in order to perform NAT discovering and typing > from the server. > > Thanks in advance, > > Víctor > > > > > 2008/4/7, Victor Cartes <ietf.sip(a)tivatechnologies.com > <mailto:ietf.sip@tivatechnologies.com>>: > > Actually not. What I need the UAS exactly to do is to send the SIP > reply to a different IP address than the source of the SIP Request > that it is replying to. Just how it sends the replies to the UDP > port specified in top most Via HF. > > > 2008/4/7, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>: > > Hi Victor, > > Not sure I got your question correctly, but you mean like > something like "received" via param used to route back the > replies? > > Regards, > Bogdan > > Victor Cartes wrote: > > Hello to all, > > About SIP/UDP: Is there any way to specify the IP address > for SIP responses, in a similar fashion as port specified > on top most Via HF? > > Thanks in advance, > > Víctor > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > >

3 3

[OpenSER-Users] Sametime
by r007b17 09 Apr '08

09 Apr '08

Hi all, i have a lotus sametime server (SIP) and a Jabber server (XMPP). Is it possible messaging with each other using xmpp Module ? Thanks in advance. ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 3

Re: [OpenSER-Users] CPL and Postgres problems.
by Bogdan-Andrei Iancu 09 Apr '08

09 Apr '08

Hmm....that is something strange - probably there is something mixed with the postgres data types..... Could you open a bug report for further investigation? Regards, Bogdan Ola Karlsson wrote: > It was TEXT and I changed it to BYTEA and that seemed to do the trick. > > /Ola > > On Mon, Apr 7, 2008 at 11:57 AM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > So, you only changed the column type (for cpl_bin) from BLOB to > BYTEA ? > > Regards, > Bogdan > > Ola Karlsson wrote: > > Bogdan.. > > > > I changed the column type in the database to bytea for cgi-bin and > > stuff started working , seems the decode only happened for BLOB type > > not for text. > > > > When debugging I got a \ instead of 0x01 så figured something was > > wrong getting stuff out of the database. > > > > /Ola > > > > On Mon, Apr 7, 2008 at 10:50 AM, Bogdan-Andrei Iancu > > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro> > <mailto:bogdan@voice-system.ro <mailto:bogdan@voice-system.ro>>> > wrote: > > > > Hello Ola, > > > > That is strange as the len of the script is taken from the > > postgres driver (via openser API) - the returned columns must be > > BLOB type. Check the types of the cpl_* columns; otherwise I > > suspect a postgres driver (in openser) issue. > > > > Best regards, > > Bogdan > > > > Ola Karlsson wrote: > > > > Hi Users! > > > > Have a problem with CPL scripts and postgres. > > I've tried to make the simplest possible script for > debugging > > which resulted in the following in the 'cpl_xml' column. > > > > '<cpl>\012 <incoming>\012 <reject status="reject" > > reason="I reject anonymous calls"/>\012 > > </incoming>\012</cpl>\012\012' > > > > Loading it with openserctl that gets translated to this > in the > > cpl_bin field. > > > > > '\001\001\000\000\000\006\002\001\000\000\000\006\026\000\002\000\000\000\002[\000\001\000\031I > > reject anonymous calls\000\000' > > > > The reason I figure that postgres could have something to do > > with this is that I've got this working before(before being > > quite a while ago though. ) with MySQL. > > > > Turning on debugging gives this information. > > > > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c B4script > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_dest_user: trying to get user from new_uri > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_dest_user: trying to get user from R_uri > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_user_script: fetching script for user <olak> > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_user_script: we got the script len=0 > > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c After Script > > > > OpenSER says that the length of the script is 0 whatever > > script I load?. > > I changed that in the CPL-C module so we get the right > length > > of the compiled script but then it falls over on the sanity > > check for NODE_CPL = 001 , and if you skip that check it > > starts complaining about unknown nodes ( in my case 92 ) ... > > > > So my question is if anyone uses postgres and CPL, if so can > > you give me some pointers? > > Would be good to know if this happens to be a bug or an > > untested area in that case I have no problem fixing the > > problem just don't want to burn alot of time on something > > already fixed or me doing something stupid with > configuration. > > > > Regards > > /Ola Karlsson > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > <mailto:Users@lists.openser.org <mailto:Users@lists.openser.org>> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Serusers] Adding authentication through mySQL
by frq ser 09 Apr '08

09 Apr '08

Hi, I am trying to add user authentication through mySQL database to my SER (0.9.7-pre1). When I tried to run SER with the modifed ser.cfg file, it gave the error that it could not locate the 'mysql.so' module. Then I tried to 'make' the mysql module following the instructions given here: http://www.iptel.org/faq/why_does_ser_fail_to_start_with_an_error_load_modu… When I typed 'make modules=modules/mysql modules' at the SER source directory, it gives me the error message saying that it could not locate 'mysql.h' and 'errmsg.h' (complete log given at the end of this email). I've tried to locate these files on my computer but couldn't find them. Note that mySQL is working fine on its own. I need your suggestions as to how I should proceed. Thanks in advance for any help. Frq

2 3

[Serusers] ser sip-t or sip-i
by mashudi 09 Apr '08

09 Apr '08

Dear all, is ser can handle sip-t or sip-i invite message ? have some one got experience with sip-t or sip-i using ser ?

1 0

[OpenSER-Users] [Fwd: Re: Bug with OpenSER 1.3 Presence module : event handling.]
by Anca Vamanu 09 Apr '08

09 Apr '08

-------- Original Message -------- Subject: Re: [OpenSER-Users] Bug with OpenSER 1.3 Presence module : event handling. Date: Wed, 09 Apr 2008 11:20:04 +0300 From: Anca Vamanu <anca(a)voice-system.ro> To: Jeremy McNamara <jj(a)nufone.net> CC: Iñaki Baz Castillo <ibc(a)aliax.net>, users(a)openser.org References: <b769badb0710240220pe56ebe2mf617230fdb667586(a)mail.gmail.com> <200710241333.09880.ibc(a)aliax.net> <471F34B5.6040806(a)voice-system.ro> <47FBB0CE.4000909(a)nufone.net> Hi Jeremy, Firstly, no, the message: '"Missing or unsupported event header field value / ^Ievent= message-summary"' has nothing to do with presence. It is about another feature - notifications about whether the user has voice mail messages pending. Now, since presence does not work for you with xlite, I am wandering if you have clicked the option Show contact availability , or something like this, when editing the buddy details. It is at the end of the form. regards, Anca Vamanu Jeremy McNamara wrote: > Anca Vamanu wrote: > >> Hello, >> >> A major change in the presence structure occurred in openser 1.3. You >> now need to load two modules to have presence running: presence and >> presence_xml. >> Most probably you didn't load the presence_xml module and this is why >> the error message appears. >> >> > > Currently I am at Interop HotStage in Belmont, CA trying to deploy a > 'unified communication' situation to demonstrate for the Interop Vegas > show here in a few weeks. I am in charge of the OpenSER/Asterisk > components and presence is giving me a huge headache. > > Hopefully someone out there can show me what I am doing wrong. > > I am running Openser v1.3.1 from tarball on Ubuntu Gutsy, no xcap > server (haven't figured out if we really need one or not for our demo...) > > I have loaded both presence and presence_xml. As documented in the > reply (down below) if you notice there is a 'tab' character in the > event= presence, I too experience this, however mine is with > message-summary. > > I see the Subscribe/Publish messages coming into openser and they are > getting routed to the appropriate config block and processed as I have > seen documented, but every so often the "Missing or unsupported event > header field value / ^Ievent= message-summary" error pops up. > > Between two X-Lite softclients using "presence agent' method, we never > seem to get any presence notifications, I presume due to this error, > but I am unsure. > > All dependencies on both presence.so and presence_xml.so seem to be > met (verified with ldd) > > # ldd presence_xml.so > linux-gate.so.1 => (0xffffe000) > libxml2.so.2 => /usr/lib/libxml2.so.2 (0xb7e76000) > libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0xb7d2c000) > libdl.so.2 => /lib/tls/i686/cmov/libdl.so.2 (0xb7d27000) > libz.so.1 => /usr/lib/libz.so.1 (0xb7d12000) > libm.so.6 => /lib/tls/i686/cmov/libm.so.6 (0xb7ced000) > /lib/ld-linux.so.2 (0x80000000) > > (identical for presence.so) > > > Here are the relevant portions of my config: > > loadmodule "presence.so" > modparam("presence", "db_url", "mysql://XXX:XXX@localhost/openser") > modparam("presence", "max_expires", 60) > modparam("presence", "server_address", "sip:valid.ip.address:5060") > > loadmodule "presence_xml.so" > modparam("presence_xml", "db_url","mysql://XXX:XXX@localhost/openser") > modparam("presence_xml", "force_active", 1) > > ... > > if(is_method("PUBLISH|SUBSCRIBE")) { > route(2); > } > > ... > > route[2] { > xlog("L_INFO", "Dealing with $rm F:$fu T:$tu\n"); > sl_send_reply("100","trying"); > if (!t_newtran()) { > sl_reply_error(); > exit; > }; > > append_to_reply("Contact: <sip:valid.ip.address:5060>\r\n"); > if(is_method("PUBLISH")) { > if($hdr(Sender)!= NULL) { > handle_publish("$hdr(Sender)"); > } else { > handle_publish(""); > } > t_release(); > } else { > if( is_method("SUBSCRIBE")) { > handle_subscribe(); > t_release(); > } > } > exit; > } > > > > > If anyone wants to help me troubleshoot and make this work, I will > make sure you get credit for assisting us here at Interop. > > > > > Jeremy McNamara > > > >>> El Miércoles, 24 de Octubre de 2007, Antoine Fressancourt escribió: >>> >>> I've now installed SVN version and same erro occurs to me: >>> >>> Oct 24 13:32:37 [4653] ERROR:presence:handle_subscribe: Missing or >>> unsupported event header field value >>> Oct 24 13:32:37 [4653] ERROR:presence:handle_subscribe: >>> event= presence >>> >>> The client is Twinkle and presence worked well in 1.2.X. >> > > > > >

1 0

[OpenSER-Users] Missing or unsupported event header field value - presence woes
by Jeremy McNamara 08 Apr '08

08 Apr '08

I am at Interop HotStage in Belmont, CA trying to deploy a 'unified communication' situation to demonstrate for the Interop Vegas show here in a few weeks. I am in charge of the OpenSER/Asterisk components and presence is giving me a huge headache. > Apr 8 11:34:12 openser openser[23798]: > ERROR:presence:handle_subscribe: Missing or unsupported event header > field value Apr 8 11:34:12 openser openser[23798]: > ERROR:presence:handle_subscribe: ^Ievent= message-summary Hopefully someone out there can help me figure out what is going wrong. I am running Openser v1.3.1 from tarball on Ubuntu Gutsy, no xcap server (haven't figured out if we really need one or not for our demo...) I have loaded both presence and presence_xml. If you notice there is what I believe to be a 'tab' character in the event= message-summary error response. I wonder if this tab is causing the issue. I see the Subscribe/Publish messages coming into openser and they are getting routed to the appropriate config block and processed as I have seen documented, but every so often the "Missing or unsupported event header field value and ^Ievent= message-summary" error pops up. Between two X-Lite softclients using 'presence agent' method, we never seem to get any presence notifications, I presume due to the error above, but I am unsure. All dependencies on both presence.so and presence_xml.so seem to be met (verified with ldd) # ldd presence_xml.so linux-gate.so.1 => (0xffffe000) libxml2.so.2 => /usr/lib/libxml2.so.2 (0xb7e76000) libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0xb7d2c000) libdl.so.2 => /lib/tls/i686/cmov/libdl.so.2 (0xb7d27000) libz.so.1 => /usr/lib/libz.so.1 (0xb7d12000) libm.so.6 => /lib/tls/i686/cmov/libm.so.6 (0xb7ced000) /lib/ld-linux.so.2 (0x80000000) (identical for presence.so) Here are the relevant portions of my config: loadmodule "presence.so" modparam("presence", "db_url", "mysql://XXX:XXX@localhost/openser") modparam("presence", "max_expires", 60) modparam("presence", "server_address", "sip:valid.ip.address:5060") loadmodule "presence_xml.so" modparam("presence_xml", "db_url","mysql://XXX:XXX@localhost/openser") modparam("presence_xml", "force_active", 1) ... if(is_method("PUBLISH|SUBSCRIBE")) { route(2); } ... route[2] { xlog("L_INFO", "Dealing with $rm F:$fu T:$tu\n"); sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; append_to_reply("Contact: <sip:valid.ip.address:5060>\r\n"); if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) { handle_publish("$hdr(Sender)"); } else { handle_publish(""); } t_release(); } else { if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } } exit; } If anyone wants to help me troubleshoot and make this work, I will make sure you get credit for assisting us here at Interop. Jeremy McNamara > > > >

2 6

[Serusers] call recording app
by Mihaly Zachar 08 Apr '08

08 Apr '08

Hi all, I got a job now, I need to build a call recording application, what stores the rtp streams of the calls, what routed to the application. There is an existing routing logic (special sip proxy) what decides if the call should be recorded or not. If the call needs to be recorded, this proxy will route the call to this recorder application. This app needs to send through the call to the next sip hop, but it must store the RTP streams on the fly. What we have now: - we have our own rtpproxy, we can modify it easily if we need to store the streams. - we have our own B2BUA what controls the rtpproxy I just wondering if we can use SER + SEMS for this, or should we build our own system ? What should be the most effective way ? Can it be done with SEMS easily ? Thanks for the suggestions, recommendations. Misi

2 1

[OpenSER-Users] downgrading or removing openser
by Robert McNaught 08 Apr '08

08 Apr '08

Hi, I have googled and found some posts relating to this, but no definitive answers: how can I downgrade from openser 1.2.x to openser 1.1.x - do I just recompile openser 1.1 and it overwrites whatever is in 1.2? there does not seem to be a make uninstall option for make in 1.2. Recompiling does not give any errors, but wont there be modules, etc that are still left over from 1.2.x? Can anyone offer some advice? Robert

2 1

[OpenSER-Users] Specifying IP Address for SIP Responses
by Victor Cartes 08 Apr '08

08 Apr '08

Hello to all, About SIP/UDP: Is there any way to specify the IP address for SIP responses, in a similar fashion as port specified on top most Via HF? Thanks in advance, Víctor

5 12

[OpenSER-Users] ERROR:presence:publ_notify: Could not send notify for presence
by gregor.sp 08 Apr '08

08 Apr '08

Hi all, I am using OpenXcap (0.99) and OpenSER (1.3), presence authorization is working and users receive NOTIFYs when users subscribe to presence. But after updating presence information with PUBLISH no NOTIFY are sent to clients. The debug messages the server shows when the server receives a PUBLISH are: ERROR:presence_xml:get_final_notify_body: while extracting the transformation node ERROR:presence_xml:pres_apply_auth: in function get_final_notify_body ERROR:presence:send_notify_request: in function apply_auth_nbody ERROR:presence:notify: sending Notify not successful ERROR:presence:publ_notify: Could not send notify for presence Somebody could help me with this? Thanks. Gregorio -- View this message in context: http://www.nabble.com/ERROR%3Apresence%3Apubl_notify%3A-Could-not-send-noti… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 2

[OpenSER-Users] Session not stop
by Sergey Babichev 08 Apr '08

08 Apr '08

Hi all! I have found out a problem at end of a call from voip. If a call the user finishes voip that session is not broken off. If from PSTN that session comes to the end successfully. PSTN Gateway -> Openser with mediaproxy -> VoIP This is log (grep mediaproxy): call from pstn to voip, voip user ended call Apr 8 12:45:40 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:40 ser1-2 mediaproxy[15711]: session F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x: 529/302/831 packets, 105720/58476/164196 bytes (caller/called/relayed) Apr 8 12:45:40 ser1-2 mediaproxy[15711]: session F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x: ended. Apr 8 12:45:40 ser1-2 mediaproxy[15711]: execution time: 0.44 ms Apr 8 12:45:41 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:41 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:42 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:42 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:44 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:44 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:48 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:48 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:52 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:52 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:56 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:56 ser1-2 mediaproxy[15711]: execution time: 0.17 ms Apr 8 12:46:00 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:00 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:04 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:04 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:08 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:08 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:30 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:30 ser1-2 mediaproxy[15711]: execution time: 0.17 ms Apr 8 12:46:30 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:30 ser1-2 mediaproxy[15711]: execution time: 0.16 ms Apr 8 12:46:31 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:31 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:33 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:33 ser1-2 mediaproxy[15711]: execution time: 0.16 ms Apr 8 12:46:37 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:37 ser1-2 mediaproxy[15711]: execution time: 0.15 ms This is log (grep mediaproxy): call from pstn to voip, pstn user ended call Apr 8 12:53:26 ser1-2 mediaproxy[15711]: session 109131A1-48011DD-A33BB2F1-2DDABEF8(a)x.x.x.x: ended. Apr 8 12:53:26 ser1-2 mediaproxy[15711]: execution time: 0.40 ms What's problem maybe? -- Best regards, Sergey Babichev

1 0

[OpenSER-Users] SeasMessageEvent [ExpressMChannel[12]] - ERROR parsing a SEAS message event!!!
by bsb093 08 Apr '08

08 Apr '08

Hi all, I am trying to run OpenSER 1.2 with WeSIP. After the callee (shine_sip1) answer the call I get the following exception and the call hangs up after a BYE message from the caller to the calee. ExpressMessageChannel [ExpressMChannel[12]] - Done processing ACK sip:shine_sip1@192.168.2.101:5070 SIP/2.0 ExpressMessageChannel [ExpressMChannel[12]] - peerAddress = 192.168.2.100/5060 SeasMessageEvent [ExpressMChannel[12]] - ERROR parsing a SEAS message event!!! java.lang.StringIndexOutOfBoundsException: String index out of range: -243 at java.lang.String.checkBounds(Unknown Source) at java.lang.String.<init>(Unknown Source) at com.voztele.javax.sip.header.Via.<init>(Via.java:112) at com.voztele.javax.sip.header.ViaList.<init>(ViaList.java:43) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:247) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPRequest.<init>(SIPRequest.java:154) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:355) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Unknown Source) Exception in thread "ExpressMChannel[12]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Unknown Source) SeasMessageEvent [SeasRouter] - New message received, from 192.168.2.100:5060 to 192.168.2.100:5060 over UDP Please advice what can cause this exception. Thanks in advance, Shine -- View this message in context: http://www.nabble.com/SeasMessageEvent----------ExpressMChannel-12-----ERRO… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[OpenSER-Users] authentication using openser
by bhargav.b＠renovau.com 08 Apr '08

08 Apr '08

hi all, i am trying to do authentication using openser at the front end and diameter as the back end.i have installed openser and opendiameter and they are running fine independently.but when i send needed avp's from openser towards opendiameter,the opendiameter client is clsing the connection.i donot understand why this is happening.if any body have any idea or if anybody provide me the refference to any study material regarding the integration of openser and diameter client,it would really help me a lot. seeking ur help regards bhargav.b

1 0

[OpenSER-Users] tee with openser
by Robert McNaught 07 Apr '08

07 Apr '08

Hi, This is probably a simple one - I have tried googling, but am not getting the answer to this. I am trying to catch what is displayed on screen and put it into a text file using tee so I can look at it. I can do 'free -h | tee dump.txt' and dump.txt is populated with data I can do 'asterisk -cvvvv | tee dump.txt' and dump.txt is populated with data when i try 'openser | tee dump.txt' and dump.txt does not get populated with the screen info. I have changed permissions to 666 on the dump.txt file incase that was the issue. Am I doing some silly? I thought that tee only copied what was displayed on screen. Thanks Robert

2 1

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