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39 participants
37513 discussions

[Users] different between "silo" and "voice_silo" table
by cv aik 04 Oct '06

04 Oct '06

Hi, Thanx for the answer last time... I have add configuration of MSILO module in openser.cfg. The problem is I don't know how to test it (how to leave offline voice message between clients). If there is any application I have to use to save voice message, please let me know. And about the silo table, I've checked OPENSER database and I 've found that silo table is already there without I've to create it. But, when I configure my serweb, I found in my /var/www/html/serweb/config.php, there is another table named "voice_silo". There is no table named "voice_silo" in my OPENSER database. I'd like to know what's the different between "silo" and "voice_silo" (maybe in storing message (voice or text)). Thanx Regards, Aldi --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

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[Serusers] authenticating forwarded calls
by Mark Price 04 Oct '06

04 Oct '06

Is it possible to set up SER with asterisk so that any INVITE that is forwarded from SER to asterisk does not need to be authenticated by asterisk? Thanks, Mark

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[Users] Who want to take this challenge (20US$)
by Ronald Wiplinger 04 Oct '06

04 Oct '06

Hi, I want to get a success feeling, ... and with your help I would get my openser installation running with: a. mysql (done) b. nat traversal (not working yet) c. ENUM (not working yet) d. sipbroker (not started yet) e.g., *777xxxxx should go to Sipgate.de .... If you want to help me to set it up, please send me first an email and I am happy to reserve time for your set-up try. If you succeed I pay you US$ 20 to your PayPal account !!!! Environment: 1. Eyebeam is on my internal network 2. several video phones are available outside of my network (one on a public IP: Grandstream 3000 video phone) and others behind a nat 3. several ENUM numbers are available, which are going to my asterisk server (different machine, different domain name) 4. sipbroker can also reach our asterisk server 71xx IP range 192.168.10.x -------------- NAT 7111 (public IP) | | ---------------------------------------INTERNET -------------------------------------------- | | | extern: 203.70.36.126 | | second ADSL on 192.168.250.250 | public ip address | taipei.elmit.net (Openser server) other servers with 2 Ethernet | | IP range 192.168.250.x | | | | | --------------------------------------------------------------------------------------------- | | | | phones and desk top computers bye Ronald Wiplinger

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[Users] ENUM dialing
by Ronald Wiplinger 03 Oct '06

03 Oct '06

I try to call an ENUM number. I use this settings for it: mpath="/usr/local/lib64/openser/modules/" loadmodule "enum.so" # -- enum params -- modparam("enum", "domain_suffix", "e164.arpa.") route{ # ----------------------------------------------------------------- # initial sanity checks -- messages with # max_forwards==0, or excessively long requests # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 4096 ) { sl_send_reply("513", "Message too big"); exit; }; # ----------------------------------------------------------------- # NAT detection # ----------------------------------------------------------------- log("I am checking for NAT now\n"); route(2); # ----------------------------------------------------------------- # if you want to make ENUM work with numbers starting with "00", # use the following to convert "00" it into a "+" # ----------------------------------------------------------------- log("I am checking for ENUM now\n"); if (uri=~"^sip:00[1-9][0-9]*@") { # tried before: taipei\.elmit\.net") { # strip leading "00" # (change example.net to your domainname or skip the stuff after the "@") strip(2); # (adjust, if your international prefix is something else than "00") prefix("+"); }; # check if request uri starts with an international phone # number (+X.), if yes, try to ENUM resolve in e164.arpa. # if no result, try in nrenum.net if (uri=~"sip:\+[0-9]+@") { # tried before: taipei\.elmit\.net") { # (change example.net to your domainname or skip the stuff after the "@") ####if ( !enum_query("e164.arpa.") ) { #### enum_query("nrenum.net."); #### }; if (enum_query()) { t_relay(); exit; } }; I tried to add some log statements, but I never see them in /var/log/messages either. Below is the ngrep output. It seems that it does not go anywhere in route() Any ideas how to track it down? bye Ronald Content-Type: application/sdp. User-Agent: eyeBeam release 1005g stamp 32576. Content-Length: 809. . v=0. o=- 2 2 IN IP4 192.168.250.108. s=CounterPath eyeBeam 1.5. c=IN IP4 192.168.250.108. t=0 0. m=audio 48702 RTP/AVP 107 100 106 6 0 105 18 3 5 101. a=alt:1 2 : O5/HQNzE iLXVkGqH 5.32.86.166 48702. a=alt:2 1 : ok7vH1EF jO6dCAlm 192.168.250.108 48702. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=x-rtp-session-id:E4797B0FC1A140648811BD63523C3B70. m=video 40488 RTP/AVP 125. a=alt:1 2 : urVBRqXw O9lI2kQp 5.32.86.166 40488. a=alt:2 1 : LCXVKz4X nxDzxq4V 192.168.250.108 40488. a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800. a=rtpmap:125 H264/90000. a=sendrecv. a=x-rtp-session-id:B8E2EE0EA3944D6DA47E5D98AABCD52B. # U 2006/09/28 16:57:11.567171 192.168.250.41:5060 -> 192.168.250.108:40430 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.250.108:40430;branch=z9hG4bK-d87543-0c396723de34cf5a-1--d87543-;rport=40430. To: "00886228357765"<sip:00886228357765@192.168.250.41>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e6c9. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=3f7a5072. Call-ID: YmQ0YzNiYmU4MDJjMjQ4YzAyZjJjMjZiYzA3ZGYxYTk.. CSeq: 1 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.250.41:5060 "Noisy feedback tells: pid=11197 req_src_ip=192.168.250.108 req_src_port=40430 in_uri=sip:00886228357765@192.168.250.41 out_uri=sip:+886228357765@192.168.250.41 via_cnt==1". . # U 2006/09/28 16:57:11.569258 192.168.250.108:40430 -> 192.168.250.41:5060 ACK sip:00886228357765@192.168.250.41 SIP/2.0. Via: SIP/2.0/UDP 192.168.250.108:40430;branch=z9hG4bK-d87543-0c396723de34cf5a-1--d87543-;rport. To: "00886228357765"<sip:00886228357765@192.168.250.41>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e6c9. From: "Ronald-taipei @ video "<sip:7311@192.168.250.41>;tag=3f7a5072. Call-ID: YmQ0YzNiYmU4MDJjMjQ4YzAyZjJjMjZiYzA3ZGYxYTk.. CSeq: 1 ACK. Content-Length: 0. .

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[Users] Digest authentication
by Artur Hayne 03 Oct '06

03 Oct '06

Hi, Can I change the authentication protocol for any other or the Openser authenticate only with the digest? The Digest needs that the password in the DB stay in clear text and its not good. Thanks. --------------------------------- O Yahoo! está de cara nova. Venha conferir!

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[Users] Who is the rtpproxy current maintainer ?
by Jeremie Le Hen 03 Oct '06

03 Oct '06

Hi there, sorry for bugging you with such a trifling question, but I tried to send a patch to the documented RTPProxy maintainer (Maxim Sobolev) twice (@PBXPress.com and @portaone.com) but I got a mail from MAILER-DAEMON each time, claiming the user doesn't exist. Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Serusers] Re: sems pthread issue
by jorain 03 Oct '06

03 Oct '06

Hi everyone, I'm doing some heavy tests on sems, which involves making lots of calls to ivr & isdngw modules. I noticed that the sems threads size (seen from process table or system monitor) keep increases with every call made. It wasn't release even when the call is already hangup, until it reach like 2GB, then I start getting pthread error code 12 whenever I tried calling. I have to restart sems to release the threads. I'm no good in C language and not sure how I can handle the pthread error. Can anyone give me some advice as how to utilize the memory usage for sems, so that it can keep on running like SER? Best Regards, Jorain

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[Users] problem to create DB in openser cvs (1.2)
by Rafael J. Risco G.V. 03 Oct '06

03 Oct '06

Hi I've installed openser dev 1.2 to test new acc module in my lab but cant create DB in mysql server using ist openser_mysql, please see this error: ]# openser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': 10.0.1.231 creating database openser ... /usr/local/sbin/openser_mysql.sh: line 262: method: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found /usr/local/sbin/openser_mysql.sh: line 262: cseq_no: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_code: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_reason: command not found real 0m0.000s user 0m0.000s sys 0m0.000s /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found /usr/local/sbin/openser_mysql.sh: line 262: method: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: from_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_uri: command not found /usr/local/sbin/openser_mysql.sh: line 262: to_tag: command not found /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found /usr/local/sbin/openser_mysql.sh: line 262: cseq_no: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_code: command not found /usr/local/sbin/openser_mysql.sh: line 262: sip_reason: command not found real 0m0.000s user 0m0.000s sys 0m0.000s /usr/local/sbin/openser_mysql.sh: line 262: callid: command not found ERROR 1064 (42000) at line 80: You have an error in your SQL syntax; check the manual that corresponds to your MySQL serverversion for the right syntax to use near '=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wh'at line 2

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[Users] rewritehost failure route
by Cseke Tamas 03 Oct '06

03 Oct '06

Hello, I 've problem with rewritehost function in failure_route with openser 1.1.0 In core functions i don't know in which route can i use them, unlike functions in modules. I 'd like to make a failover system, In route block rewritehost is successfull but on failure route there is no effect i think, i pass the host to t_relay, so it is relayed to correct host, but in r-uri, there is incorrect host, and sip proxy reply with (SIP/2.0 400 Bad Request - 'Invalid IP Address) route{ ...... t_on_failure("1"); rewritehost("xxx.xxx.xxx.90"); t_relay(); ..... } failure_route[1] { t_on_failure("2"); append_branch(); rewritehost("xxx.xxx.xxx.91"); t_relay("xxx.xxx.xxx.91"); } failure_route[2] { # failover cisco (zajcek) append_branch(); rewritehost("xxx.xxx.xxx.31"); t_relay("xxx.xxx.xxx.31"); } Thanks any help, Tamas

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[Users] SER / Load Balancing
by Thomas Deillon 03 Oct '06

03 Oct '06

Hi everybody, I saw this: http://www.iptel.org/faq/how_can_i_get_load_balancing_failover_with_ser. On asterisk, you can put and share on DB the dial plan and the sip definition but not the information relative to the REGISTER message (localisation). So, you cannot make perfect load-balanced servers. We have here a lot of clients and asterisk boxes in 3 towns with a very good connections between them. I wondered if it's possible with ser to put and share in the database the "REGISTER" information and so able any phone to connect to any server ? Thanks a lot for your help, Best Regards, Thomas

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