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37462 discussions

[Serusers] accounting doesn't seem to work?
by Jean-Michel Hiver 19 Aug '06

19 Aug '06

Hi List, I am trying to get SER to run on FreeBSD 6 For some reason, I can't get the acc module to do anything. I have downloaded SER from CVS, changed ./modules/acc/Makefile to enable SQL accounting, and did a gmake bin include_modules="mysql", then untared the resulting .tgz in / SER starts fine, processes SIP message fine, and yields no errors in /var/log/messages. However I can't see any accounting related information in ser's "acc" table (which remains always empty). Any ideas? PS: Here is my ser.cfg (I have replaced the IP addresses with dummy addresses because I don't want them to be archived...). If somebody would be kind enough to take a look at it... debug=3 fork=yes log_stderror=no listen=123.123.123.123 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- 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/textops.so" ## ACC loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic 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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (method=="REGISTER") { sl_send_reply("501", "Not Implemented"); break; }; if (method=="OPTIONS") { sl_send_reply("501", "Not Implemented"); break; }; record_route(); if( (method=="INVITE") || (method=="ACK") || (method=="BYE")) { setflag(1); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; route(3); if (method=="INVITE") { route(2); break; }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; } route[2] { if (uri=~"^sip:001#") { rewritehostport ("123.123.123.201:5060"); }; if (uri=~"^sip:002#") { rewritehostport ("123.123.123.202:5060"); }; if (uri=~"^sip:003#") { sl_send_reply ("401", "Unauthorized"); break(); }; if (uri=~"^sip:004#") { rewritehostport ("123.123.123.204:5060"); }; route(1); } route[3] { if ( !( (src_ip == 123.123.123.201) || (src_ip == 123.123.123.202) || (src_ip == 123.123.123.204) || 1 ) ) { sl_send_reply("401", "Unauthorized"); break(); }; }

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[Serusers] Different behaviour from different clients, same firewall.
by sip 19 Aug '06

19 Aug '06

What are some things that might create one-way audio issues from different clients, all using STUN, all NATted behind the same firewall? Some UAs, I have to add port-forwarding for to ensure two-way audio (they're behaving as though they're behind a symmetric NAT). Others, I don't have any issues with at all. For instance, X-Lite? Always works. Snom? Requires port-forwarding in order to function correctly 100% of the time (often works, doesn't always). Grandstream (phones and ATAs)? Always works. UTStarCom F3000? Requires port-forwarding to function correctly. SJPhone? Always works. Etc., etc. All these devices are behind the same firewall, so I don't think it's necessarily an issue with the method of NAT that the firewall does. It's something that varies inside the UA. Any ideas what, inside a UA, could cause such disparity? I thought it might be the SDP headers, but I can't honestly tell MUCH difference between them. I'm at a bit of a loss for all the bizarre variables involved. N.

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[Serusers] iptel.org/ser
by Gaurav Kulshreshtha 19 Aug '06

19 Aug '06

Anybody noticed that the site iptel.org/ser has a new look and the ser admin and ser programmer guides (among many other links) are gone. Are these doc still accessible ?? Thanks Gaurav

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Re: [Serusers] SER on Ubuntu Breezy
by Andrey Kouprianov 19 Aug '06

19 Aug '06

Hi, Oh my. My appologies! I've mixed up with Asterisk. What ser version do you want to install, btw? If 0.9.x then plz take a look at INSTALL file in the tar ball. There should not be any problem with compiling like in any other linux system, if you've installed all the needed packages. When you make, 1. make #builds only ser core, equivalent to make ser 2. make all #builds everything 3. make bin include_modules="mysql jabber cpl-c auth_radius group_radius uri_radius postgres pa" #builds selective modules (you'll definitely need mysql and the rest is up to you) 4. Refer to INSTALL file for more options. Btw, since Ubuntu is Debian based, then also take a look at the "requirements" section for Debian. 5. Make sureto install MySQL first. I used 4.1.x for many installations so far and it seems to be working really fine. Then make install. This should install ser into default directory ( /usr/local ) with configuration file inside the /usr/local/etc/ser/ and shell scripts with binaries in /usr/local/sbin/. Andrey. On 8/19/06, Eddie Johnson Jr <ilo(a)ilo-infosystems.us> wrote: > Are you saying in this e-mail that a package exist in the repository because > I have checked by searching SER and Sip Express Router and nothing. Any > suggestions will be greatly appreciated. > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kouprianov > Sent: Thursday, August 17, 2006 10:57 PM > To: serusers(a)iptel.org > Subject: Re: [Serusers] SER on Ubuntu Breezy > > Hi, > > A friend of mine has installed SER on Ubuntu 5.10 before from Ubuntu's > package list. I dont know the details, but it should be fairly easy. > Flex and bison MUST be installed on Ubuntu by default. All u need to > install is MySQL server first, and then SER. Ubuntu should take care > of the rest or at least tell u which package is missing. > > On 8/17/06, Eddie Johnson Jr <ilo(a)ilo-infosystems.us> wrote: > > > > > > > > Has anyone done an install of SER on Ubuntu 5.10. I need some initial > setup > > points and I can take it from there. I installed, flex, bison, > > build-essential, ran make clean nothing, ran make install and I get > makefile > > 80 no directory or file. Where do I go from here? > > > > > > > > Regards, > > > > > > > > iloadmin > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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[Users] avp_write in 1.1.0
by Stuart Clark 18 Aug '06

18 Aug '06

Hey all.. First time poster, but long time reader. Thanks to all that contribute to this excellent project, this software is very flexible! I just wanted to mention that while upgrading my config to 1.1.0, I found that.. avp_write("$ruri/username", "$avp(myuser)") or avp_write("$ru/username", "$avp(myuser)") does not work as expected, and does not return just the username. Instead I needed to change it to.. avp_write("$rU", "$avp(myuser)") BTW, am I able to update via SVN to a newer bug fixed version of the 1.1.0 branch, without taking on any new things in the HEAD branch? Is it just as simple as this, and I have just not tried it. I wish to use this code in a production environment. Kind regards, Stuart.

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RV: [Serusers] Via RPORT for CANCEL message.
by Ricardo Martinez 18 Aug '06

18 Aug '06

Hello Greger and thanks for your answer I tested sending the CANCEL to route(3), and it seems to be working ok now. But now i'm facing another problem. The OK message for the BYE seems not to be routed back to the NAT'd port, instead is routed to the default port. Please check the file attached (BYE_debug.TXT). As i pointed in my first post, the calling endpoint is NAT'd. Following the BYE sequence seems to be right, it hits the loose_route section and then the route(1), but since the BYE does not contain the rport parameter in the VIA header the OK is routed back to the port advertised in the VIA header. (5061) and not to the received port. Maybe i'm doing something wrong?. I tested a new solution, but i don't know if is correct or is according to the ONSIP document. I added a client_nat_test("3") in the Call Tear Down Section- .......... # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("200.100.100.246:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { if (client_nat_test("3")) { force_rport(); xlog("L_INFO", "[%rm] - From:%fu ; To:%ru ; Call-ID:%ci ; Desde:%is\n - FORCE RPORT ON"); }; end_media_session(); setflag(1); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; ....... Hope that someone could help me here. Thanks Regards, Ricardo Martinez.- -----Mensaje original----- De: Greger V. Teigre [mailto:greger@teigre.com] Enviado el: jueves, 17 de agosto de 2006 4:05 Para: Ricardo Martinez CC: serusers(a)iptel.org Asunto: Re: [Serusers] Via RPORT for CANCEL message. Having a quick look, you may have found a bug in the ONsip.org scripts. A CANCEL is not loose routed (a dialog has not yet been set up) and must be handled the same way as an INVITE, i.e. the only way to find the route for CANCEL is to route the CANCEL the same way you routed the INVITE. However, it seems that you have somehow has done that, as your domain has changed to @gw. in the CANCEL doing downstream (i.e. forwarded by your SER). So, CANCEL should be sent to route(3), the INVITE handler and not route(1). I'm quite sure this was correct in a previous version, we even had some discussions around how to handle it and how to describe it, but it seems that something has happened along the way. g-) Ricardo Martinez wrote: Hello. My main configuration is based on the ser.cfg from the issue 5 plus some modifications included by me. Beside that i'm using version: ser 0.9.3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 11:03:37 Dec 22 2005 with gcc 3.2 I have a question regarding to sending a CANCEL message for an INVITE from a NAT'd endpoint. As far as i know the force_rport() command adds the received IP port to the top most via header in the SIP message, this enables subsequent SIP message to return to the proper port later on in a SIP transaction. My problem is that for a CANCEL message coming from a NAT'd endpoint this command seems not to be working. This is the scenario. NAT'd endpoint : 200.100.100.248 SER : 200.100.100.246 SER-2 : 200.100.100.36 GW : 200.100.100.69 The NAT'd endpoint send an INVITE to the proxy, but then in the middle of the transaction decide to CANCEL the request. As you can see in the "cancel_debug.txt" file included on this mail, the CANCEL message does not contain the "rport" in the Via header, so it seems to be routed back to the default sip port (5060). Is there a way to force the rport in a CANCEL? for a enpoint unable to put the rport by itself? I want to do this according to the onsip document, in a compatible way.. Anyway i made a little test including a force_rport(); in the CANCEL handler .......... # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("200.100.100.246:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); setflag(1); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="CANCEL") { force_rport(); route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="ACK") { route(1); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); this solve my problem but it seems not to be so accurate. I hope that someone can help me Thanks! Ricardo Martinez.-

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[Serusers] RTPPROXY doesn't work: HELP!!!
by chiara.alena＠fastwebnet.it 18 Aug '06

18 Aug '06

Hi everybody! I am running SER + RTPPROXY and I am trying to establish a call between a UA with public IP address and a UA with private IP address connecting to the proxy through a Access Point that has an extern ip address that is private too: UA1 (192.168.0.100) <--> (192.168.0.1) AP (192.168.1.1) <--> (192.168.1.2) SER (public IP) <--> (public IP) UA2 I just used the function force_rtp_proxy of nathelper module to force SER to change the sdp body of INVITE and reply messages, and in this way the fields c= and m= of the sdp get changed correctly. I can establish a call and the UAs send RTP packets to the rtpproxy. The problem is that the rtpproxy doesn't forward the RTP packets coming from the UA with public address (UA2) towards the UA with private address (UA1). Are there special flags I have to set in order to make rtpproxy work properly? Do I have to use other functions of nathelper module besides force_rtp_proxy? I am writing my own module for SER and I programmed it in such a way that the force_rtp_proxy function is called (not from the configuration file but from the module itself) for every INVITE request and reply, so that every RTP stream has to pass through the rtpproxy. If I try to use 2 UAs with public addresses everything works properly... Last question: how can I run rtpproxy in debug mode? And where can I see the log files??? Please someone help me! Thanks a lot! Chiara

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[Users] tcp_blocking_connect() in timeout
by Federico Giannici 18 Aug '06

18 Aug '06

I have just upgraded from 1.0.0 to 1.1.0-snapshot of a couple days ago, and now I get the following errors: Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR: tcpconn_connect: tcp_blocking_connect failed Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR: tcp_send: connect failed Aug 18 14:50:56 eowyn OpenSER[16760]: msg_send: ERROR: tcp_send failed Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR:tm:t_forward_nonack: sending request failed I'm using OpenBSD 3.9-stable amd64. What could be the problem? Thanks. P.S. It could be a good idea to add in the 1.0.x to 1.1.x upgrade web page that the "module_exports" structure has changed. P.S. 2 My system is a 64 bit one, but the standard library directory is the usual "lib" and NOT "lib64". -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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FW: [Serusers] Connect Ser on a Asterisk
by Richard C. Thompson 18 Aug '06

18 Aug '06

Sure In sip.conf create a [user] and a [peer] or you can use [friend] if the para's are the same [SER] type=user disallow=all allow=g729 allow=ulaw allow=alaw context=default canreinvite=no permit=10.0.0.2/255.255.255.255 host=10.0.0.2 trustrpid=no [SER] type=peer insecure=yes host=10.0.0.2 context=default nat=no canreinvite=no promiscredir=yes disallow=all allow=g729 allow=ulaw trustrpid=no Rick _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Daniel Mirrha Santos Sent: Friday, August 18, 2006 8:48 AM To: Ser Subject: [Serusers] Connect Ser on a Asterisk Hi, I'm using SER like a public switch and I need to register it on a Asterisk server. Does anybody know how could I do that? Regards, Daniel

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Re: [Serusers] How to send Calls from Ser to Asterisk Voicemail?
by Ricardo Carvalho 18 Aug '06

18 Aug '06

I think Ser also can fire some kind of interruption that calls some routine when the call rings for some time. This routine being called should send that call to voicemail. Anyone knows how can I do that in ser.cfg? Thanks, Ricardo. (Also thanks Juan for your post!) Juan wrote: > Hi Ricardo. > SER doesn't care if the call is answered or not. > Asterisk yes. From the little I know about * you will have to setup a > mailbox for each customer so if he/she is not online or doesn't answer the > call it will be redirected after certain amount of time to the mailbox. > Hope this helps. > Regards > > Juan > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ricardo Carvalho > Sent: Thursday, August 17, 2006 1:33 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] How to send Calls from Ser to Asterisk Voicemail? > > How should I configure ser.cfg to enable Ser to send calls to Asterisk > Voicemail when after ringing for some while, the phone isn't answered? > > Thanks, > > Ricardo. > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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