sr-users November 2009

sr-users@lists.kamailio.org

80 participants
108 discussions

[Kamailio-Users] Social Networking Event
by Daniel-Constantin Mierla 04 Nov '09

04 Nov '09

Hello, several folks working with Kamailio, SIP Router, SER, OpenIMSCore, SEMS and Asterisk are in Berlin next week, so we think of having a dinner (or beer) meeting Thursday, 19:00, Nov 12, 2009. If happens that you are around and want to join, please send me an email to make sure you get a seat. As usual for this kind of event, free participation with everyone paying for himself/herself. Just to have an idea about how it could be, see some photos from previous similar events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ From private discussions so far, we should be already about 10 people, confirmed including myself, Olle E. Johansson (Asterisk) and Elena-Ramona Modroiu. There is a lot to celebrate, from integration work within SIP Router Project to upcoming Kamailio 3.0 (now RC1). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] [Fwd: Re: uac_replace_from again]
by marius zbihlei 04 Nov '09

04 Nov '09

Forgot to forward to list

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[Kamailio-Users] syntax error
by GEMINORIUM crux 04 Nov '09

04 Nov '09

hi, i am new to kamailio, getting syntax error: $var(test) = "udp:"+$rd+":"+$rp; t_relay_to_udp($var(test)); t_relay_to_udp($rd,$rp); working with string. is there any reason ? how to resolve it ? i have to use database so i need to use variable. any suggestion is appreciated. thanks

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[Kamailio-Users] Kamailio 1.5.0 and NAT
by Путято Денис 04 Nov '09

04 Nov '09

Hello everybody I have such problem with kamailio and client which behind NAT This is REGISTER message which came from client. This message came from public IP (for example, 1.1.1.1) and source port 44197. (This information I got from sip_trace) REGISTER sip:sip.ptl.ru:5060;transport=UDP SIP/2.0 From: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=a82eda3-13c4-47798948-1a1296-5259a46 To: <sip:xxxxxxx@sip.ptl.ru:5060> Call-ID: a82eda3-13c4-47798948-1a1296-9641c88(a)sip.ptl.ru CSeq: 446229198 REGISTER Via: SIP/2.0/UDP 10.130.237.163:5060;branch=z9hG4bK-47799e27-6b9841-84448db Max-Forwards: 70 Supported: timer Contact: <sip:xxxxxxx@10.130.237.163:5060;transport=UDP> Expires: 60 Content-Length: 0 This is Unauthorized message was sent by kamailio to 1.1.1.1 and: destination port not 44197 (as I expected) but 5060!!! SIP/2.0 401 Unauthorized From: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=a82eda3-13c4-47798948-1a1296-5259a46 To: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=fdc454a3cdd8f9e5718382b74f33aeed.3b81 Call-ID: a82eda3-13c4-47798948-1a1296-9641c88(a)sip.ptl.ru CSeq: 446229198 REGISTER Via: SIP/2.0/UDP 10.130.237.163:5060;branch=z9hG4bK-47799e27-6b9841-84448db;received=1.1.1.1 WWW-Authenticate: Digest realm="domain name", nonce="4af0129b000000047c1020258f9f0efc4de1b76663dd2ab4" Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 And as you understand 401 reply, which was sent to 1.1.1.1:5060 can't reach client's equipment. A piece of kamailio.cfg relating to REGISTER processing: route(0) { :. if (!is_method("INVITE")) route(8); :. if (is_method("REGISTER")) { route(11); exit; } :.. } route(8) { if (nat_uac_test("5")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } route(11) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if (isflagset(5)) setbflag(6); if (!save("location")) { sl_reply_error(); exit; } } Hearty thanks for any help.

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[Kamailio-Users] 404 sparks next gw and 503
by Anders 03 Nov '09

03 Nov '09

Hi, This list becomes a mix of high-level tech and low-level beginner questions... - and I'm in the latter group. Making two lists might be over-kill, though. Here's my issue: My call termination provider is rightfully returning a 404 for a call to a number that doesn't exist. This currently seems to spark Kamailio to look for the next gw, but in this case there is no next gw set up in the lcr/gw - so it returns a 503 to the caller. If I'm confident that my provider is right that there is no such number, and 404 is the correct answer, wouldn't it be better to return a 404 to the caller rather than a 503? With 503 stats also get screwed up, because my client can just see a bunch of 503s, which is not a very detailed picture of the real situation. If I should return a 404, how would I go about this? What would I do in cfg to ensure that a 404 return does not spark it to look for a next gw, but sends 404 back to the caller? Alternatively, I could try to figure out how to get the return from my provider saved to the acc table on the INVITE record, and I'd have something more detailed to run stats on. Does that make sense? Thanks!

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[Kamailio-Users] OpenSER/Kamailio behind NAT
by systeo 02 Nov '09

02 Nov '09

Hi, I successfully installed Kamailio on the Amazon computing cloud (EWS), and it runs fine. However, I can't get to it (register) from any UACs from the public internet. In amazon, all the instances (i.e. "virtual machines) are in a "private" IP, kindda like the private IP you get in a LAN. However, Amazoin gives you a "public IP" that can be mapped to any machine instance in your account, i.e. the machine that kamailio is running. However, when i use this public IP in the configuration of the UACs, it cannot get to it. How can I get kamailio to listen on this public IP? Secondly, and related to the first, when I run kamailio in my LAN (behind a router), say on 192.168.1.113, and all my UACs are in the same LAN, say 192.168.1.112, 192.168.1.114, etc, I can register ok and make calls just fine. However, when I try to come in from the public internet, no UACs can register to my Kamailio server (running on 192.168.1.113), even though I am forwarding port 5060 from my router to 192.168.1.113 (where kamailio is running). I think this problem is related to the first one, in that I cannot get Openser (kamailio) to listen on a public IP, when it is behing a "LAN". Can anyone help? Lastly, the same issue with rtpproxy also behind the LAN, but it needs a pulic IP, right? Please help...anyonw. Hassen BEN SLIMA Analyst developer TEL : 00216 74 863 507 FAX : 00216 74 863 506 E-MAIL : <mailto:tarak@systeo.biz> hassen(a)systeo.biz Web : <http://www.systeo.biz> www.systeo.biz Adresse : Route de Tunis km 10, Technopôle de Sfax - 3021 - Sakiet EZZIT - Sfax - Tunisie <http://www.systeo.biz/> Sans titre-4

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[Kamailio-Users] force_rtp_proxy() usage with IP that is not natively homed
by Alex Balashov 02 Nov '09

02 Nov '09

Greetings, Please excuse the apparent silliness of this request, but I am curious if there is a combination of flags passed to force_rtp_proxy() and/or rtpproxy invocation options that can allow force_rtp_proxy() to substitute an SDP endpoint IP that is not a physical device/interface on the host on which rtpproxy is running? Let me explain further: 1. I have a host with a single network interface that has private IP 192.168.1.254. 2. I am receiving an inbound call from outside the network, coming in on a public IP address 4.4.4.4 that is a 1-to-1 DNAT/SNAT to 192.168.1.254. 3. The call is forwarded to a PBX on the same LAN as the native interface, e.g. 192.168.1.250. 4. I would like to set up SDP in such a way that 192.168.1.250 sendsmedia to the 192.168.1.254 interface while the outside host initiating the call sends its media to 4.4.4.4. Under normal circumstances in which both the public and private interface were physically present on the machine, this would call for the use of "bridging" mode (rtpproxy invoked with -l 4.4.4.4/192.168.1.254 and use of "ie" and "ei" flags with force_rtp_proxy() - we have previously established that rtpproxy_offer/answer do not actually seem to work as advertised). But is it possible to "bridge" to an IP that is not routed to the box? I expected that force_rtp_proxy()'s second parameter - ip_address - would do this. However, this does not appear to let me override the SDP endpoint with any IP address I want; it seems that I am still limited to choosing from those IPs physically available on the rtpproxy host. I would guess that this is because ip_address is passed as a parameter through rtpproxy's control API socket and is validated by rtpproxy itself, and if it is not a local existing address, is overridden with one that is. Is there a clever combination of flags that can be used to accomplish this, or is it a lost cause? Thanks! -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Serusers] error: SER/FIFO not accessible: 2
by Brian Artigas 02 Nov '09

02 Nov '09

Hello, I'm trying to add a user to the database and getting the following error back: error: SER/FIFO not accessible: 2 Here is an excerpt from my ser.cfg file: debug=3 fork=no log_stderror=yes listen=192.168.1.30 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo_mode=0777 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) Anyone have any ideas? Thanks, Brian

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sr-users November 2009