sr-users October 2006

sr-users@lists.kamailio.org

189 participants
345 discussions

RE: [Serusers] Presence
by senthil.nachimuthu＠bt.com 04 Oct '06

04 Oct '06

Hi Javier, Thanks for your information. I need to make SEP to send/proxy the message to my presence server. SEP and presence server are physically different entities. Would be of greater help if you could let me know how can i specify the IP address of my presence server in SEP so that the presence specific messages reaching SEP gets forwarded to my presence server. Thanks and Regards, Senthil Kumar, ________________________________ From: Javier Ramirez [mailto:jvr_78@yahoo.com.ar] Sent: 25 September 2006 15:03 To: Nachimuthu,SK,Senthil,XSD4C C Subject: Re: [Serusers] Presence if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); }; handle_subscription("registrar"); }; if (method=="PUBLISH") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_publish("registrar"); break; }; ----- Original Message ----- From: senthil.nachimuthu(a)bt.com To: serusers(a)lists.iptel.org Sent: Monday, September 25, 2006 8:28 AM Subject: [Serusers] Presence Hi, I want to configure SER <file://ser/about> 0.9.6 in such a way that It can proxy presence messages(SUBSCRIBE,PUBLISH) to presence server. Can you let me know how can I do that? Thanks and Regards, Senthil Kumar, Presence Capability, BT-Service Execution Platform, B62-MH,Orion Building,Adastral Park,Ipswich Work:+44 (0)1473 605 668 Mobile:+44 (0)7772287447 __________ NOD32 1.1768 (20060922) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers __________ NOD32 1.1768 (20060922) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com

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[Users] rtpproxy has been disabled temorarily
by Ronald Wiplinger 04 Oct '06

04 Oct '06

If I use # -- nathelper -- modparam("nathelper", "rtpproxy_sock", "udp:59.124.210.41:22222") # default port 22222 #modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:59.124.210.41") # ----------------------------------------------------------------- # NAT detection # ----------------------------------------------------------------- route(2); route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } I get in the log file: Oct 4 09:19:01 taipei Taipei-openser[4083]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4085]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4087]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4089]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4091]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4050]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4067]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily Oct 4 09:19:01 taipei Taipei-openser[4069]: WARNING: rtpp_test: support for RTP proxy <udp:59.124.210.41:22222>has been disabled temporarily with: modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") this error does not appear. In neither way it work for me (ring but no audio) Can anybody help me please? bye Ronald Wiplinger

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[Users] NAT: STUN, nathelper, rtpproxy, mediaproxy
by Ronald Wiplinger 04 Oct '06

04 Oct '06

We have STUN, nathelper with rtpproxy or mediaproxy, ... maybe even more. Which one is good for which situation? What are the differences? How can I cover each situation? Can I combine different methods? bye Ronald

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[Users] SER with multiple asterisks
by Adi Simon 04 Oct '06

04 Oct '06

Hi, I have one SER and two asterisk boxes. I'm trying to duplicate each register message from the client so it will reach each of the asterisks. I tried the following in my openser.cfg: if ( (is_method("SUBSCRIBE")) || (is_method("REGISTER")) ) { seturi("sip:nobody@10.0.0.1:5060"); append_branch("sip:parallel@10.0.0.2:5060"); if (!t_relay()) { sl_reply_error(); }; exit; } The first asterisk server registers the UA successfully. The second one gives this error: Oct 3 12:55:48 NOTICE[32364]: chan_sip.c:6395 check_auth: stale nonce received from 'sip:1234566@foo.bar' Transmitting (no NAT) to asterisk2:5060: I googled a bit and I understand I cannot fully duplicate a sip message since the nonce cannot be identical. Is there a way to recreate the sip message with another nonce? Is there any other solution to this scenario? Thanks, Adi.

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[Serusers] SRV usage with SER
by Paul Antinori (pantinor) 04 Oct '06

04 Oct '06

Does SER support the SRV mechanism, sending a request to a URL with a SRV domain name? Also, if the first priority destination fails due to a timeout, will it failover to the second priority? Thanks for help, Paul

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[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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sr-users October 2006