sr-users May 2008

sr-users@lists.kamailio.org

121 participants
202 discussions

[OpenSER-Users] serweb-0.9.6 on openser-1.3.2

by Zen Kato

I installed serweb-0.9.6 on openser-1.3.2, but when I enter username and password, the following Warning comes out. How should I solve this problem? -------------------------------------------- http://localhost/openserweb/user/index.php username:0103@192.168.0.3 password:xxxxx --------------------------------------------- ----------------------------------------------------------------------------- http://localhost/openserweb/user/my_account.php?kvrk=483a8e78876b5 ( ! ) Warning: fopen(/tmp/openser_fifo) [function.fopen]: failed to open stream: Permission denied in /var/www/openserweb/serweb-0.9.6/html/functions.php on line 345 ----------------------------------------------------------------------------- /var/www/openserweb/serweb-0.9.6/html/functions.php (L:344-348) /* open fifo now */ $fifo_handle=fopen( $config->fifo_server, "w" ); if (!$fifo_handle) { $errors[]="sorry -- cannot open write fifo"; return false; } -- Zen Kato

16 years, 6 months

[OpenSER-Users] Possible problem with script presence-create.sql

by Florent Le Saout

Hello, I'm a new user of openser. (I use openser 1.3.1-3+b1 from package of debian testing) I want to use the module of presence. If I refer to documentation I read that I have to use presence_xml-create.sql to create the tables. But I can't find it. I use the script presence-create.sql which seems to replace it. I create a table named xcap but the module need xcaps, if I create the same table renamed xcaps that work well. I think there is a mistake of naming of the table xcap(s). I take a look to 1.3.x of your repository and I see the same thing in the sql script. Thank you for your job. Best regards. Florent le Saout. Créez votre adresse électronique prénom.nom(a)laposte.net 1 Go d'espace de stockage, anti-spam et anti-virus intégrés.

16 years, 6 months

[OpenSER-Users] Acceptor peer failed establishing state[3003], closing

by bhargav.b＠renovau.com

actually i am authenticating a user using openser and opendiameters NASREQ application as the client and server.at present the openser is sending the all needed avp's to opendiameter client.i can see all the avp code and avp data on the client side. my problem is that the on the diameter server side the avp,s are not transferred by the client. the client is giving something like (6372|1698151312) Acceptor peer failed establishing state [3003], closing i am confused about it.by reading some drafts i came to know that the error is due to realm mismatch.but i am sending the same realm as the opendiameter's realm.in both the cases it is [bhargav.s]. before i sent the avp's from openser side there was no such problem showing on the client side.if anybody know the solution please help me regards bhargav.b(a)renovau.com

16 years, 6 months

[OpenSER-Users] How to determine that a reply is a retransmission?

by Jon Steer

Environment: Openser 1.3.1-TLS Fedora 7 I have a module that performs a heavyweight operation on Invites and their respective Replys. I would prefer not to do the operation again if there is a retranmission of the Reply. How can I determine in a REPLY-ROUTE if a reply has already been seen? $T_reply_code would work, if on entry to the first reply route it was zero, but I suspect it is the value of the current reply. thanks, jon

16 years, 6 months

[OpenSER-Users] Centos 5.1 or Debian4.0.r0 for Openser server

by fatmeh fatah

Hello: May you comparison Centos 5.1 and Debian4.0.r0 for Openser server? Thank you.

16 years, 6 months

[OpenSER-Users] Click to Call functionality

by Darren Sessions

Quick question . . >From what I've been able to gather, the fifo db functionality has been removed since 1.2.x. What is another easy way to do click-to-call functionality now that it is gone? Thanks, - Darren

16 years, 6 months

Re: [OpenSER-Users] Problems with sl module in openser1.3.1

by Arturo Ochoa

On Friday 23 May 2008, arturo arturo wrote: > Could some help me with this problem, I'm trying to run openser 1.3.1, but > it dies complaining about the sl module. I issued just 'make all' command > before this > > root@zeus:/home/aochoa/openser/openser-1.3.1-notls > # ./openser > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/mysql.so > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/sl.so > May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags > mismatch for* /usr/local/lib/openser/modules/sl.so > core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, > SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, > USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Hi Arturo, the version of the sl module installed on your machine don't match the one of your server in the home directory. Do a make distclean; make all in the source directory, and install again. Cheers, Henning Thanks for your response Henning, I've already do what you mention, but the problem persists. I tried also with openser 1.3.2, and the same thing. Also I noted that there are different linked libs on the same module but compiled on different machines: Fedora: [root@apolo] /usr/local/lib/openser/modules # ldd sl.so linux-gate.so.1 => (0x0028b000) libc.so.6 => /lib/libc.so.6 (0x00111000) /lib/ld-linux.so.2 (0x0028c000) Centos: (this is the one with the problem) root@zeus:/home/aochoa/openser # ldd /usr/local/lib/openser/modules/sl.so libc.so.6 => /lib/tls/libc.so.6 (0x00149000) /lib/ld-linux.so.2 (0x003f7000) Any comments?

16 years, 6 months

[OpenSER-Users] CANCEL not always beeing relayed

by Tobias Lindgren

Hi all, I have some problems with CANCELs not always beeing relayed from A to B via an OpenSER 1.3.1. When I try to make a call and CANCEL it with my own equipment, everything works just fine. OpenSER receives CANCEL and passes it on to the PBX. However, some customer calls that are beeing CANCELed are not always relayed by OpenSER and I cannot find out why. They are received by OpenSER but never sent forward. I have sip-logs and OpenSER debugs, are there someone with more understanding of the OpenSER-code that can take a look at them? Br, /Tobias

16 years, 6 months

Re: [OpenSER-Users] call failed 403 forbiden

by luzango mfupe

Hi mates, I still need your pointers regarding my problem in this post, today i have attached the routes suspected to be involved in this saga. From my config file plz see below. # ----------------------------------------------------------------- # Unauthorized relay # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local()) { # We prevent unauthorised relays "clever guys we got u" append_hf("P-hint: outbound\r\n"); route(10); # need to be authenticated } else { sl_send_reply("403", "Forbidden"); }; return; }; route[10] { #------------------------------------------------------------- # Default Message Handler with Proxy Authentication # ----------------------------------------------------------------- if(method=="ACK") { #these you never proxy authenticate route(1); return; }; if(method=="BYE" || method=="CANCEL") { #these you never proxy authenticate route(1); return; }; xlog("L_INFO", "Proxy auth $fd $dP destination:$du $dd $ds");# if (!route(7)) { #verify the user return(0); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; # consume_credentials(); }; # if (is_user_in("Credentials", "local")) { # Uncomment to use the group options route(1); # }else{ # sl_send_reply("403", "Busted!!!, you are not allowed this route"); # }; return; } With Best Regards, LU. > > Message: 3 > Date: Tue, 20 May 2008 17:38:50 +0200 > From: "luzango mfupe" <luzango.mfupe(a)gmail.com> > Subject: [OpenSER-Users] Call failed 403 Forbiden > To: users(a)lists.openser.org > Message-ID: > <9cdd611a0805200838oc11cfedg9762b7451bf543c4(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi mates, > Everytime i do try to make my 2 xlite clients talk (which i correctly added > them into the database), i encountered with the same problm, Openser > perfoms > authentication and return call failed 403 forbiden signal. > > My setup comprise of the first box with openser 1.3 and mediaproxy the > second box with Mysql and two Xlite clients in two other boxes all are in > an > internal network, as far as am concerned my NetAdmin have already opened > ports 5060 and 3306 for me. I need your right direction on this probm. > below is my ngrep snapshot > > mzee:/# ngrep -d eth1 -W byline port 5060 > interface: eth1 (168.172.200.0/255.255.255.0) > filter: (ip or ip6) and ( port 5060 ) > # > U 168.172.200.70:1824 -> 168.172.200.87:5060 > INVITE sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> < > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87> > >SIP/2.0. > Via: SIP/2.0/UDP 168.172.200.70:1824 > ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. > Max-Forwards: 70. > Contact: <sip:dreamteam@168.172.200.70:1824>. > To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>< > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>>>. > From: "dream"<sip:dreamteam@168.172.200.87<sip%3Adreamteam(a)168.172.200.87>< > sip%3Adreamteam(a)168.172.200.87 <sip%253Adreamteam(a)168.172.200.87>> > >;tag=af4bd714. > Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. > CSeq: 1 INVITE. > Session-Expires: 95. > Min-SE: 90. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, > INFO. > Content-Type: application/sdp. > Supported: timer. > User-Agent: X-Lite release 1011s stamp 41150. > Content-Length: 426. > . > v=0. > o=- 9 2 IN IP4 168.172.200.70. > s=CounterPath X-Lite 3.0. > c=IN IP4 168.172.200.70. > t=0 0. > m=audio 52166 RTP/AVP 107 119 100 106 0 105 98 8 101. > a=alt:1 1 : uZB2dYm+ NKBRK8Ep 168.172.200.70 52166. > a=fmtp:101 0-15. > a=rtpmap:107 BV32/16000. > a=rtpmap:119 BV32-FEC/16000. > a=rtpmap:100 SPEEX/16000. > a=rtpmap:106 SPEEX-FEC/16000. > a=rtpmap:105 SPEEX-FEC/8000. > a=rtpmap:98 iLBC/8000. > a=rtpmap:101 telephone-event/8000. > a=sendrecv. > # > U 168.172.200.87:5060 -> 168.172.200.70:1824 > SIP/2.0 403 Forbidden. > Via: SIP/2.0/UDP 168.172.200.70:1824 > ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. > To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>< > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>> > >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. > From: "dream"<sip:dreamteam@168.172.200.87<sip%3Adreamteam(a)168.172.200.87>< > sip%3Adreamteam(a)168.172.200.87 <sip%253Adreamteam(a)168.172.200.87>> > >;tag=af4bd714. > Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. > CSeq: 1 INVITE. > Server: OpenSER (1.3.0-notls (i386/linux)). > Content-Length: 0. > . > # > U 168.172.200.70:1824 -> 168.172.200.87:5060 > ACK sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> < > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>> > SIP/2.0. > Via: SIP/2.0/UDP 168.172.200.70:1824 > ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. > To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>< > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>> > >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. > From: "dream"<sip:dreamteam@168.172.200.87<sip%3Adreamteam(a)168.172.200.87>< > sip%3Adreamteam(a)168.172.200.87 <sip%253Adreamteam(a)168.172.200.87>> > >;tag=af4bd714. > Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. > CSeq: 1 ACK. > Content-Length: 0. > > WBR, > LU. > >

16 years, 6 months

[OpenSER-Users] Problem with BYE signaling during no answer call forward to PSTN

by Krunal Patel

Hi, Here is the scenario of the call. 12345 ---> 56565656 ----> PSTN I have registered user 12345 using sjphone. Dialing 56565656 which is registered to the same openser. Now i let the call ring & after timeout call is transffered to PSTN number. Which is terminated by PSTN gateway. I am using mediaproxy for audio signaling & getting audio properly both side. Now the problem is when PSTN number hangsup the call,it should send BYE signal to the domain name. But instead of sending BYE signal to domain name it sends it to the IP. & because of that the UA can't acknowledge the BYE signal & stay connected. & BYE keeps looping in openser. As the pstn number sends BYE to IP , I have tried by setting alias=xxx.xxx.xxx.xxx . Eventhough it was not solved. Thanks in advance -- Regards, Krunal Patel

16 years, 6 months

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sr-users May 2008