sr-users September 2005

sr-users@lists.kamailio.org

231 participants
508 discussions

[Serusers] installation & first run problem (I am new, ...)
by Ronald Wiplinger 29 Sep '05

29 Sep '05

I am new to SER, forgive me if I use "strange terms" or ask stupid or obviously things ;-) I downloaded SER and compiled from source. I tried to use the ser_mysql.sh script to create the database and the permittion. For me unknown reasons, the permissions for all ser(ro) are set to 'N' in the table mysql.user I tried to change it to 'Y' with GRANT ALL PRIVILEGES ON ser.* to 'ser'@'localhost' IDENTIFIED BY 'password' but that does not change anything in the mysql.user Strange is also that the database ser does not show up at all in phpMyAdmin and also the 4 records in mysql.user. Before I changed /usr/local/etc/ser/ser.cfg to reflect the mysql I tried to start ser with /usr/local/sbin/serctl start and I had many ser process in my ps list, but now it says: :/usr/local # /usr/local/sbin/serctl start Starting SER : PID file exists! (/var/run/ser.pid) already running? What have I done wrong? bye Ronald Wiplinger

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[Users] asterisk
by Urban 29 Sep '05

29 Sep '05

Hi, I'm new to the list and didn't find this information in the archives so bare with me if this has already been discussed in the list. I'm trying to setup openser toghether with asterisk on the same machine. I don't get it to work using the basic information I found on the web with rewritehostport("localhost:5090") and t_relay(). openser is running on port 5060 and * on 5090. When using tcpdump on the machine openser is not even contacting asterisk on port 5090 instead it returns "100 trying..." to the phone. openser will just work as a proxy to * and add some extra security but most of the pbx routing logic and voicemail will be handled by *. Anyone having this up and running and willing to share the required configuration? thanks urban

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[Users] avp error
by jayesh nambiar 29 Sep '05

29 Sep '05

hi bogdan, Thanks for the help. atleast I got rid of that error but got a new error. I am trying to implement the ACL Bitmap example provided in the avp_docs. The idea is as follows: # 1 - free destination # 2 - local call # 3 - long distance call # 4 - international call # if(uri=~"sip:0900[0-9]+") /* free destinations */ { if (!avp_check("$acl", "and/i:0x01") { sl_send_reply("403","Forbidden - Free Destinations Not Allowed"); return; }; } else if(uri=~"sip:0[1-9][0-9]+") { if (!avp_check("$acl", "and/i:0x04") { sl_send_reply("403","Forbidden - Long Distance Calls Not Allowed"); return; }; } else if(uri=~"sip:00[1-9][0-9]+") { if (!avp_check("$acl", "and/i:0x08") { sl_send_reply("403","Forbidden - International Calls Not Allowed"); return; }; } else { if (!avp_check("$acl", "and/i:0x02") { sl_send_reply("403","Forbidden - Local Calls Not Allowed"); return; }; }; # authorized # rewritehostport("gateway_ip:gateway_port"); rewritehostport("10.10.10.10:5060"); But now I get the following error: ERROR:avpops:parse_check_value: parse error in <and/i:0x01> pos 3 Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: ERROR:avpops:fixup_check_avp: failed to parse checked value Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: ERROR: fix_expr : fix_actions error Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: Binding '2204','sip:2204@192.168.2.20:8484' has expired Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: use_table(): Invalid parameter value Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: db_del_ucontact: Error in use_table Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: wb_timer(): Can't delete contact from the database Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: Binding '2207','sip:2207@192.168.2.18:8467' has expired Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: use_table(): Invalid parameter value Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: db_del_ucontact: Error in use_table Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: wb_timer(): Can't delete contact from the database Is there anything wrong with the expression syntax or there could be problem with the entry in the table. Can you pls help me figure it out. thanx in advance. jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

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[Serusers] Mediaproxy works only for the caller
by Felipe Louback 29 Sep '05

29 Sep '05

Hey all, I am using ser-0.9.4 and mediaproxy-1.4.2 but it seems that mediaproxy only works for the caller. During a call, if I use sessions.py, i see: Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------------------------- 150.164.x.x:8000 - 127.0.0.1:35000 - ?.?.?.?:? inactive 0'26" Unknown Audio 0/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 Proxy version: 1.4.2 The status is inactive and after 60 seconds it ends. The called part receives audio but the caller receives nothing. It seems that the called part is not passing through media proxy. Here it is my ser.cfg: route{ if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too many hops"); }; break; }; if (msg:len >= max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; record_route(); if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; t_relay(); break; }; if (method=="REGISTER") { # Mark as NAT'ed if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; if (!www_authorize("mydomain", "subscriber")) { www_challenge("mydomain", "0"); break; } if (!save("location")) { sl_reply_error(); }; break; }; if (method == "BYE" || method == "CANCEL") { end_media_session(); }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } How do I do for the called part to pass through mediaproxy? I've searched about this, but I found nothing. Initially I was using ser0.8.14 and I thought it was a problem with it and I am now using 0.9.0. It seems that the problem wasnt with ser version 0.8.14. Could anyone give me some advice or tell me where I can find more information about it? Thanx, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "For God so loved the world that he gave his one and only Son, that whoever believes in him shall not perish but have eternal life." John 3:16

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[Serusers] Re: RTPproxy and SER
by Alexandre Aractingi 29 Sep '05

29 Sep '05

Hi again, I made the same test with mediaproxy instead of rtpproxy, the problem is exactly the same. In my config there is no NAT at all, private IP addresses are routed via an IP tunnel to my SER proxy, and I'd like to use the media proxy to communicate with an external gateway. The SDP within the 200OK is left intact, so a private IP address is given and the call fails... Any ideas? Thanks! > Le mardi 27 septembre 2005 à 18:30 +0200, Alexandre Aractingi a écrit : > > Hi all, I'm trying to use RTPproxy (CVS) and SER to enable > > locally-routed endpoints to talk to an external public gateway. When a > > call comes from the external gateway, the SDP in the INVITE gets > > rewritten properly with SER's IP address, but then when my phone hangs > > off, the SDP in the 200 OK doesn't get rewritten, so an RFC1918 IP > > address is passed to the external gateway (so I get no audio). > > > > I'm using nathelper with: > > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") > > > > and for my tests I put "force_rtp_proxy()" at the very beginning of the > > routing logic in order to make sure it gets invoked. > > > > But the SDP in the 200 OK is not rewritten no matter what. Is this a > > know issue? Is there a known workaround? > > > > Thanks in advance for any help, > > > Alexandre Aractingi > Net-tone / Active Telecom > Direct IP : +33 1 72 74 70 02 > Standard : +33 1 49 23 76 59 -- Alexandre Aractingi <aaractingi(a)libertysurf.fr>

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[Serusers] timestamp field in ACC table
by Ryan Pagquil 29 Sep '05

29 Sep '05

Hi guys, I've just found out that the timestamp field in ACC table doesn't contain the real timestamp relative to 1970. It is just like the time field in ACC table but without the ":" and the "-" delimiters. So when I computed for the duration it doesn't give me the correct duration time. Here is the query I used to extract for duration: SELECT t1.from_uri as CLR, t1.to_uri as CLD, (t2.time - t1.time) as Duration, t1.time as Setup_Time, t2.time as Release_Time FROM acc t1, acc t2 WHERE t1.sip_callid = t2.sip_callid AND ((t1.fromtag = t2.fromtag and t1.totag = t2.totag) OR (t1.fromtag = t2.totag and t1.totag = t2.fromtag)) AND t1.sip_method='ACK' AND t2.sip_method='BYE'; attached is a text file that contains one of the record in acc table. Thanks, Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com +-------------------------------------------------------------------------------------+---------------------------------------------------------+------------+------------+-----------------------------------------------------------------------------+------------------------------+----------------------------------+-------------------------------+--------------------------------------------------------+-----------+---------------+----------------------------------+------------+---------------------+----------------+----------------+----------------+ | sip_from | sip_to | sip_status | sip_method | i_uri | o_uri | from_uri | to_uri | sip_callid | username | domain | fromtag | totag | time | timestamp | caller_deleted | callee_deleted | +-------------------------------------------------------------------------------------+---------------------------------------------------------+------------+------------+-----------------------------------------------------------------------------+------------------------------+----------------------------------+-------------------------------+--------------------------------------------------------+-----------+---------------+----------------------------------+------------+---------------------+----------------+----------------+----------------+ | "5997" <sip:5997@202.147.52.22:5060>;tag=5deb812f096b2f10914b000000000000 | "632108" <sip:632108@202.84.24.107:5060>;tag=3332695765 | 200 | INVITE | sip:632108@202.84.24.107:5060;transport=UDP | sip:user1@10.0.1.144:5060 | sip:5997@202.147.52.22:5060 | sip:632108@202.84.24.107:5060 | 000d2858-7ff00284-462b78c8-5c0771e7(a)202.147.52.22 | 5997 | 202.147.52.22 | 5deb812f096b2f10914b000000000000 | 3332695765 | 2005-09-28 06:51:11 | 20050928145111 | 0 | 0 | | "5997" <sip:5997@202.147.52.22:5060>;tag=5deb812f096b2f10914b000000000000 | "632108" <sip:632108@202.84.24.107:5060>;tag=3332695765 | 200 | ACK | sip:202.84.24.107;ftag=5deb812f096b2f10914b000000000000;lr=on | sip:user1@202.84.24.126:1470 | sip:5997@202.147.52.22:5060 | sip:632108@202.84.24.107:5060 | 000d2858-7ff00284-462b78c8-5c0771e7(a)202.147.52.22 | 5997 | 202.147.52.22 | 5deb812f096b2f10914b000000000000 | 3332695765 | 2005-09-28 06:51:11 | 20050928145111 | 0 | 0 | | "5997" <sip:5997@202.147.52.22:5060>;tag=5deb812f096b2f10914b000000000000 | "632108" <sip:632108@202.84.24.107:5060>;tag=3332695765 | 200 | BYE | sip:202.84.24.107;transport=UDP;ftag=5deb812f096b2f10914b000000000000;lr=on | sip:user1@202.84.24.126:1470 | sip:5997@202.147.52.22:5060 | sip:632108@202.84.24.107:5060 | 000d2858-7ff00284-462b78c8-5c0771e7(a)202.147.52.22 | 5997 | 202.147.52.22 | 5deb812f096b2f10914b000000000000 | 3332695765 | 2005-09-28 06:52:13 | 20050928145213 | 0 | 0 |

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[Users] B2bua +Ser +Asterisk
by harry gaillac 29 Sep '05

29 Sep '05

Hello, I wish to set b2bua-1.5.0 (Vovida) between Ser and Asterisk. Ser-------B2bua-------Asterisk---Pstn | Billing How may I set b2bConfig.xml in order to send calls to pstn via b2bua (billing) and asterisk ? Can I receive calls from pstn to ser via b2bua if asterisk is able to send call to b2bua ? Can we use b2bua for both incoming/outgoing calls ? Regards Harry .......... <Proxy_Server> <Address>xxx.xxx.xxx.xxx</Address> <Port>xxxx</Port> </Proxy_Server> .......... ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Users] voicemail howto
by Taymour A. El Erian 29 Sep '05

29 Sep '05

Hi. Is there any how to on how to configure OpenSER, SEMS for voicemail ?, the one with SEMS does not work -- Taymour A El Erian System Division Manager RHCE, LPIC, CCNA, MCSE, CNA TE Data E-mail: taymour.elerian(a)tedata.net Web: www.tedata.net Tel: +(202)-4166600 Fax: +(202)-4166700 Ext: 1101

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[Users] access control list using avps
by jayesh nambiar 29 Sep '05

29 Sep '05

hi daniel, the $avp is an alias to i:800. My main doubt is where in the table should i mention the values 1,2,3 and 4 for the users who got the appropriate privileges. I am sorry, for i get confused manipulating the values of avp. pls help in this. thanks a lot jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

4 3

[Serusers] about my previous question [please help me about how to config radiusclient!]
by zhangshuai 29 Sep '05

29 Sep '05

Dear all, Sorry to bother you again. After googled the error messages: Sep 29 17:11:06 localhost ser[4408]: rc_ip_hostname: couldn't look up host by addr: DA61FC29 Sep 29 17:11:06 localhost ser[4408]: rc_send_server: no reply from RADIUS server unknown:1812 I found this: http://forums.staros.com/showthread.php?p=21895 "The key thing most people miss is that the IP you enter in the radius server for the remote that will talk to it has to be the IP of the device that the server is listening on. Most people simply use the IP of the gateway device and that is not correct unless that device also has Hotspot or PPPoE or some other service that talks to radius." I still don't know how to check his words are right. What do you think? Shuai http://www.goldentek.biz

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sr-users September 2005