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37573 discussions

[Kamailio-Users] avpops OR sqlops
by alex pappas 31 Dec '09

31 Dec '09

Hi all, I'm trying to make an sql query through my kamailio script in order to check for number portability. i have a table with 2 col (phonenumber,prefix). I'm trying both modules avpops and sqlops but I'm not able to retreive any data. modparam("avpops", "db_url","mysql://openser:openserrw@localhost/openser") avp_db_query("SELECT prefix FROM ported WHERE phonenumber='$rU'","$avp(s:lnp_prefix)") ;The query run perfectly in command line. The return code is always 1 but when I print $avp(s:lnp_prefix) I get no result. I have 2 questions : 1. Which is the best way to check portability in my case 2. What I'm doing wrong and I'm not getting any result from the query. Thank you ALL Alex

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[Kamailio-Users] 1.5.3 dialog module: dialogs stay in database when in "shutdown only" mode
by Nathan Angelacos 30 Dec '09

30 Dec '09

Hi, It appears that in "shutdown only" mode (modparm db_mode = 3 ) dialogs are not cleared from the database on startup: 1. kamctl start initiate call 2. kamctl stop 3. kamctl start terminate call (dialog is terminated correctly) 4. kamctl db show dialog shows the dialog record 5. kamctl stop 6. kamctl db show dialog shows the (now very dead) dialog record Perhaps the dialog table should be cleared right after step 3 or at the beginning of step 5 when the dialog module is in "shutdown only" mode.

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[Kamailio-Users] Problems with Sip Redirect Client
by Daniel Cardoso Alves 30 Dec '09

30 Dec '09

Hello... I have a SIP Redirect Serve with a kamailio and it works fine with others system(NexTone, Asterisk, etc), but I have a problem with a SIP redirect client with Kamailio. I am using http://kamailio.org/docs/modules/1.5.x/uac_redirect.html to try manipulate the 3xx message. I would like to send one destiny at a time, but the get_redirects("10:2","redirect") send only 2 contacts and at teh same time. I get from my SIP Redirect Server this message: SIP/2.0 301 Moved Permanently..Via: SIP/2.0/UDP 2XX.XX.XXX.XXX:5080;branch=z9hG4bK633e.a3e32ff2.0;rport=5080;received=2XX.XX.XXX.XXX..Via : SIP/2.0/UDP 192.168.0.133:7191;received=201.XX.X.XXX;branch=z9hG4bK-d87543-254f7f3a1058e232-1--d87543-;rport=7191..To: <sip:05433330000 @2XX.XX.XXX.XXX:5080>;tag=609b19d62bd6886602a6c902875525b0.c8b2..From: "1938"<sip:1938@2XX.XX.XXX.XXX:5080>;tag=187fec3c..Call-ID: 4959bc 623e556571@ZGVzZW4tcGMtMDguZ292b2lwLmNvbS5icg....CSeq: 1 INVITE..Contact: sip:55341#5433330000@192.168.0.2, q=1, sip:123#5433330000@1 92.168.0.3, q=2, sip:55341#5433330000@192.168.0.4, q=3, sip:55341#5433330000@192.168.0.5, sip:55341#5433330000@192.168.0.6, sip:553 41#5433330000(a)192.168.0.7, sip:55341#5433330000@192.168.0.8, sip:55341#5433330000@192.168.0.9, sip:55341#5433330000@192.168.0.10. .Server: Test Redirect..Content-Length: 0.... How I can configure to send fist to sip:55341#5433330000@192.168.0.10 and after if I not sucess it sends to sip:55341#5433330000@192.168.0.9, etc...? Thanks... ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

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[Kamailio-Users] [kamailio-users] Kamailio 1.5.0 + MediaProxy 2.3.8
by Chandrakant Solanki 30 Dec '09

30 Dec '09

Hi Kamailio - 1.5.0 MediaProxy - 2.3.8 CentOS 5.2 - 2.6.18-92.el5PAE My kamailio is running on *192.168.X.Y* ** *Here is my /etc/mediaproxy/config.ini* ------------------------------------------------- ; Configuration file for MediaProxy ; [Relay] dispatchers = 192.168.X.Y relay_ip = <public ip> port_range = 8000:10000 log_level = DEBUG stream_timeout = 90 [Dispatcher] socket = /var/run/mediaproxy/dispatcher.sock listen = 192.168.X.Y listen_management = 192.168.X.Y log_level = DEBUG [TLS] certs_path = tls verify_interval = 300 [OpenSIPS] max_connections = 100 =========================================================== Start media-dispatcher / media-relay # media-dispatcher --no-fork # media-relay --no-fork When I see log it show dispatcher to relay and relay to dispatcher connected.. But when I call 101 to 102 it gives me error, as show below *Media Dispatcher error* error: No suitable relay found *Media Relay error* warning: Cannot use port pair 8684/8685 warning: Cannot use port pair 8686/8687 -- Regards, Chandrakant Solanki

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Re: [Kamailio-Users] Caller and Callee behind the same NAT
by juergen＠glowka.de 29 Dec '09

29 Dec '09

On Tuesday 29 December 2009 01:55:27 juergen(a)glowka.de wrote: > Hello, > > I'm new here. So I say "hello" to everybody. > > Maybe, somebody can help me or push me in the right direction. > > If there are Caller and Callee behind the same NAT, so it should > be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), > instead of using the rtp-proxy (this causes a jitter noise) . > > > I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. > > Has anybody a code snip or a sample kamailio.cfg for me? > I would be thankful for it. Better than giving you the solution, try to guest it ... Think a little ... what have in common a REQUEST from UAC A to UAC B if they are behing THE SAME NAT ROUTER ? -- Raúl Alexis Betancor Santana Dimensión Virtual ____________________________________________ Dear Mr. Santana Thanks for trying to help. oh, I guess, they've got the same IP-Address ;-) I found this code-snip in : dokuwiki/doku.php/examples:caller-callee-behind-same-nat ______________________________________ else if ( isflagset(2) and isflagset(3) ) { log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("i:450", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Clients Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(2); resetflag(3); } } _______________________________________ But this Code-snip contains many mistakes so I changed it to : _______________________________________ if ( isflagset(5) and isbflagset(6) ) { lookup("location"); log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("$avp(i:450)", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(5); resetbflag(6); } } _______________________________________ This seems to work, but I still have got this looping, jittering feeping noise witch is getting louder and louder ... Two or three words I can talk, then it sounds, as if a group of Navajo-Indians gone attack me .... :-) Could it be the client? I use 2 PDAs, Windows Mobile, Portsip Ver. 2.6 Best Wishes, also for me Juergen Glowka

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[Serusers] using ser to involve nat
by WangZhenzhong 29 Dec '09

29 Dec '09

Hi every body I am using ser to solve the nat problem, In my environment: invite A-->SER-->B 1. SER receive the invite message REQA from A(invite B) 2. SER judge the use B, if the B is in public network, will fix nat in the REQA message if judge the B is in private network , will not fix anything and I want to know, how to judge the B's network, when the SER receive the INVITE from A? Is there any examle code used in ser.cfg thank you very much 2009-12-29 姓名：王振中公司：艾诺通信系统(苏州)有限公司地址：苏州工业园区苏虹西路81号苏虹工业坊A8单元二楼

1 0

[Kamailio-Users] Caller and Callee behind the same NAT
by juergen＠glowka.de 29 Dec '09

29 Dec '09

Hello, I'm new here. So I say "hello" to everybody. Maybe, somebody can help me or push me in the right direction. If there are Caller and Callee behind the same NAT, so it should be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), instead of using the rtp-proxy (this causes a jitter noise) . I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. Has anybody a code snip or a sample kamailio.cfg for me? I would be thankful for it. Best wishes Juergen Glowka _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

2 1

Re: [Kamailio-Users] adding inbound and outbound routes seperately
by Klaus Darilion 28 Dec '09

28 Dec '09

please always Cc: the list.... On 23.12.2009 16:24, Rodney wrote: > Well could you explain how to actually get peering functioning > between Kamailio on inbound and outbound? > There is no "peering functioning" in kamailio. You self have to define how you want to peer. For example if you want to peer with somebody sending you inbound requests, you have to define at least: - number format - authentication (IP-based, TLS, digest) One you have defined those parameters, you can implement it, e.g. digest authentication or IP based authentication. The same for outbound route - you have to configure Kamailio so that outbound requests are sent in the proper format. And Kamailio can not register to any other SIP service. If you need such a feature, you better use Asterisk. regards klaus > > -----Original Message----- From: Klaus Darilion > [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 23, > 2009 8:49 AM To: Rodney Cc: users(a)lists.kamailio.org Subject: Re: > [Kamailio-Users] adding inbound and outbound routes seperately > > > On 22.12.2009 22:35, Rodney wrote: >> Ok, I have kamailio and siremis, and I need to know how to set up >> a separate in bound and outbound route. >> >> Peering with inbound and registering with outbound >> > > Kamailio is a sip proxy, thus it can not register to other SIP > proxy. > > regards klaus > >> Any ideas guys? >> >> R >> >> >> >> _______________________________________________ Kamailio (OpenSER) >> - Users mailing list Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] Newbie questions
by Antonio Goméz Soto 28 Dec '09

28 Dec '09

Hi, I have used asterisk before, but now am looking at kamailio for scaling up. I've read 'Building Telephony Systems with OpenSER', but I still have lots of questions.... so, here it goes: - I could not find a way to reboot a phone remotely, can it send SIP NOTIFY to all phones? - does it have something like the Asterisk Manager Interface, so I can connect/create an operator panel for it? Does that already exist somewhere? Something like FOP where operators can transfer calls, and check availability? - according to the docs it is not a BTBUA. What does this mean? I know that asterisk is a BTBUA, but what can a BTBUA do that a SIP proxy cannot? Is it only the moh and conferences? - lastly, is there a good tutorial on load-balancing somewhere? Thanks, Antonio.

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[Serusers] About iptel 400 missing cookie
by Dennis.Cheng 28 Dec '09

28 Dec '09

Hi folks, I used a Draytek Vigor 2820PBX to register iptel, and I found 400 missing cookie issue. When my pbx send the reinvite to hold another side, iptel always responded pbx 400 missing cookie. But I checked invite Via field, its branch included magic cookie "z9hG4bK". I attached wireshark packets. BR Dennis

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