sr-users November 2005

sr-users@lists.kamailio.org

245 participants
570 discussions

[Serusers] avp_write/avp_pushto in database/memory?

by Ramin Nikaeen

Valued Colleagues, Do the avp_write and avp_pushto involve Database access operations?! For example: avp_write("sip:22@10.1.10.60:5060", "s:callfwd"); avp_pushto("$ruri", "s:callfwd"); Does avp_write store the string "sip:22@10.1.10.43:5060" for "s:callfwd" in database and avp_pushto looks up the string related to "s:callfwd" from the database or is the information stored in memory?! I want to avoid database access to ensure improved performance. Thanks ramin

19 years

[Serusers] call forwarding and *21

by Kyriakos Mavromichalis

Can someone point me to any resources on how to implement services like callforwarding by pressing *21 through the users' UA? Basically I need my users to be able to activate different services via the phone's keys. thank you -- Kyriakos Mavromichalis Otenet Telecom

19 years

[Serusers] Problem with running SER with mysql module

by Murděj Ukrutný

Hi, I am a beginner here. Could anyone please answer my question? I have downloaded the SER 0.9.4 and compile with mysql module it on SUSE Linux Enterprise Server 9. Mysql 4.1.15 compiled from source. I have problem whith running SER. Command 'serctl start' finished with message 'Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed' Message in /var/logs/messages is 'Nov 16 05:34:34 suse1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.14: cannot open shared object file: No such file or directory' 'libmysqlclient.so.14' is in '/usr/local/lib/mysql/' Any idea ? Thank. -- Murdej Ukrutny

19 years

[Serusers] Mediaproxy High Load

by Adrian Georgescu

The CPU will show in Linux high usage as soon as you get over 20 calls but you will be able to load the box up to 200 streams, the CPU will show the same high value while the proxy works fine. Use the rtpgenerator to test after what amount of calls your box will start loosing packets we reckon 70 streams per 1 GHz. Regards, Adrian <<<<<< Hi All, I tested the load of the mediaproxy (ver 1.4.2) and found that it is quite high. To give an idea, on a P4 2.4GHz machine, 20 sessions give a reading of about 35%, and it will go to almost 90% for 50 sessions. If we have a system that say can accommodate 10000 users (behind nat) or concurrent calls, that probably means we need about 10000/50=200 machines. That is way too many boxes. Does anybody has a solution for this? Thanks for any help available. Regards, TC Chan

19 years

[Serusers] difference between callsetup asterisk & ser and related security risks

by Arne Van Theemsche

Hi, I've been spending days on tcpdumping callsetups with ser OR asterisk as registar. This is my conclusion, can somebody confirm this? On asterisk: -asterisk maintains 2 independendly sipsessions between 2 clients, where asterisks stays the contact in the sipheaders. After picking up the call, asterisk sends an re-invite in the sip-body (address), and remains the contact for sipmessages. If someone hangs up, asterisk sends an re-invite to the other party, for redirecting stream back to asterisk, and then sending the bye to the other client (the one who didn't hang up). on ser: ser receives the invite, looking up the address of the callee, and sending the invite through to the callee, the contact header allways stays the caller's uri, but ser put's a record-route in the sipmessage, so that the bye first is send to the ser machine, and then to the other party. Am I seeing things correct here? What are the pro/contra's of both procedures? Imo gives the asterisk way a lot more overhead, but is more secure (2 clients keep communicating with asterisk, and are unable to fake things (or less easely), there were an not compliant client can bypass the record-route of ser, and so avoiding accounting. Again, am I correct here? thx Arne

19 years

[Serusers] ERROR: exec_str:

by davor jovanovic

Hi all, I'm using SER 0.9.3 and trying to do LDAP query but I keep getting this message: 0(17313) INFO: dont_fork turned on, living on 0(17313) exec_str: rtrim 0(17313) ERROR: exec_str: cmd /etc/ser/sipldap failed. exit_status=-1, errno=10: No child processes can someone explain to me why is that and how can I fix it? when I'm doing lookup with: (!lookup("location")) { if (!exec_dset("/etc/ser/sipldap")) { sl_send_reply("404", "Not Found"); break; } else { log(1," sipldap call"); }; }; I'm getting "Not Found", but when I'm doing (transferring to PSTN route without sl_send_reply) (!lookup("location")) { if (!exec_dset("/etc/ser/sipldap")) { #sl_send_reply("404", "Not Found"); route(5); break; } else { log(1," sipldap call"); }; }; route[5] { rewritehostport(xxxxxxxxxxx); I can extract LDAP telephone number and do get call transferred to PSTN GW but still getting "ERROR: exec_str:" thanx davor

19 years

[Serusers] Inaccurate Radius Accounting

by Lenir Santiago

Hey guys, I'm using Radius to do pretty much everything regarding my users, i.e. registrations, authorization, features and preferences. I implemented my own way of doing speeddial by using AVPs provided by radius and AVPOPs module, Here is the snippet: AVP-SIP = speedial:30:15615551212 avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); #15615551212 avp_printf("i:98","$rU:*"); #30 avp_printf("i:97","sip:$avp($temp2)@$td"); # sip:15615551212@xx.xx.xx.xx avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # <sip:15615551212@xx.xx.xx.xx>. xlog("L_N"," SPEEDIALING: $fU is speed-dialing $hdr(To)->$avp($temp2)\n"); if (avp_check("s:caller_speedial", "fm/$temp1/g")) { #avp_pushto("$To","i:96"); #avp_pushto("$ruri","i:97"); #avp_pushto("$duri","i:97"); #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" <$avp($temp2)@\2/'); #append_hf("To: \"5615551212\" <15615551212(a)xx.xx.xx.xx>;user=phone"); #subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); #subst_user('/(.*)/$avp($temp2)/'); #remove_hf("To"); rewriteuser("15615551212"); xlog("L_N"," SPEEDIALING: $fU is dialing $hdr(To) $avp($temp2)\n"); avp_delete("$temp1/g"); avp_delete("$temp2/g"); }; In this scenario, im forcing it instead of getting the value from AVPs just for ease of showing my point. The problem that I'm having is that no matter what I do to change the To header or change the R-URI, radius ALWAYS sends the accounting records with the original R-URI, in my case 30(a)xx.xx.xx.xx, how can I get acc.so to send the final resulting URI (15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, length=298 Acct-Status-Type = Start Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = Invite User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway;user=phone" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "102" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 rlm_sql (sql): Reserving sql socket id: 12 rlm_sql (sql): Released sql socket id: 12 Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, length=292 Acct-Status-Type = Stop Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = 8 User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "103" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 Thanks in advance! Lenir

19 years

[Users] no voice when using nathelper with rtpproxy

by messa＠innsof.com

Hi all I install rtpproxy in the same host (with the public IP) than openser. the installation of rtpproxy put rtpproxy.sock in /var/run directory.But when i make a call between 2 clients (each client in a different LAN), i have no voice after signalisation. And openser give me these errors: [root@Is-dev01 openser-1.0.0-tls]# /usr/local/sbin/openser Listening on udp: 62.128.173.101 [62.128.173.101]:5060 tcp: 62.128.173.101 [62.128.173.101]:5060 Aliases: tcp: 62-128-173-101.iwayafrica.com:5060 udp: 62-128-173-101.iwayafrica.com:5060 WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 2(0) INFO: fifo process starting: 20621 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) INFO: no fifo_db_url given - fifo DB commands disabled! 0(20619) ERROR: force_rtp_proxy2: no available proxies 0(20619) ERROR: force_rtp_proxy2: no available proxies 0(20619) ERROR: unforce_rtp_proxy: no available proxies What's the matter? Best regards Serge

19 years

[Serusers] Mediaproxy High Load

by tcchan

Hi All, I tested the load of the mediaproxy (ver 1.4.2) and found that it is quite high. To give an idea, on a P4 2.4GHz machine, 20 sessions give a reading of about 35%, and it will go to almost 90% for 50 sessions. If we have a system that say can accommodate 10000 users (behind nat) or concurrent calls, that probably means we need about 10000/50=200 machines. That is way too many boxes. Does anybody has a solution for this? Thanks for any help available. Regards, TC Chan

19 years

[Serusers] Incoming message

by Lazy.charles

19 years

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sr-users November 2005