- sr-users - kamailio.org

[Serusers] problem with auth_db module
by harry gaillac 02 Jun '05

02 Jun '05

Hello, I have a problem with auth_db module when ser start. However ser start (ser.cfg should be ok) but ser_fifo is missing. Need Help Harry serveur1:/etc/ser# /etc/init.d/ser restart Restarting ser: serListening on udp: 80.119.7.156 [80.119.7.156]:5060 tcp: 80.119.7.156 [80.119.7.156]:5060 Aliases: tcp: 156.7.119-80.rev.gaoland.net:5060 udp: 156.7.119-80.rev.gaoland.net:5060 *: nxs.yi.org:* . serveur1:/etc/ser# tail /var/log/syslog Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: Maxfwd module- initializing Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: permissions - initializing Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: Default allow file (/etc/ser/permissions.allow) not found => empty rule set Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: Default deny file (/etc/ser/permissions.deny) not found => empty rule set Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: auth_db:str_fixup: Invalid table version (use ser_mysql.sh reinstall) Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: ERROR: fix_expr : fix_actions error _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

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[Serusers] interconnect with ipcb
by Iqbal 02 Jun '05

02 Jun '05

Hi A few people here were looking to terminate traffic using SER and looking for pstn end gateways, well if you sign up with ipcb.net (and no I am not connected to them..but hoping if they read this, i will get a pizza out of this :-)) you can connect SER to any of the gateways you wish worldwide. How they do it is simple, they have written a module which you compile into ser, load as normal, it has a user/pass in it, and u assign there IP address as trusted in ur config. Then when any requests are sent, the auth you using that user/pass, and let traffic flow. Works fine. Now I was wondering if some of the programmers on this list could take a look at the module (they will email it to you) surely it would be easy to do the same for any gateway out there....I guess Anyhow now all SER people can get their own gateways. Iqbal

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[Serusers] SER mhomed=1 & SEMS (call accounting)
by Óscar Patrício 02 Jun '05

02 Jun '05

Hi! I have a SER server running as a "bridge" (mhomed=1) between a private and a public network. When someone from outside my domain calls a user of my domain, and that call is answered by the voicemail, when I hang up the call (Bye request) after leaving the message, I get an error in the ser log. When SER receives that Bye request, it shows an error saying it can't find the transaction. In addition to that, the Bye request isn't logged to the database, so i don't know when the call actually finished. Can anyone give me a clue here? Thanx a lot, Óscar Patrício

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[serusers] ping and latency
by michael p 02 Jun '05

02 Jun '05

hi was someone already do a programm to know the latency of each users in the location table of ser? i want to execute it and that is take all registered users in location table and send a OPTIOS message to ser. like serctl with the command : serctl fifo t_uac_dlg OPTIONS "sip:michael@195.XXX.XXX.XXX:XXXXX" "." "From: sip:daemon@dev.annatel.net" "To: <sip:michael@195.XXX.XXX.XXX:XXXXX>" "Contact: <sip:daemon@>" "." "." but with timestamps and put all of them in a new table thanks M.

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[Serusers] SIp Reinvites and billing
by Shidan 02 Jun '05

02 Jun '05

Is there any way to do postpaid billing without ser sitting in the media path, or do we need a B2BUA all the time.

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Re: [Serusers] Call Hunting
by Dave 02 Jun '05

02 Jun '05

Kapil, 1) In failure_route[2] , you cannot have t_on_failure("2"). I believe a "recursive" call does not work. You can write a failure_route[3] and call t_on_failure("3" from failure_route[2] 2) To add something to the usr_preferences table, you can use the SQL INSERT command. You would launch the mysql client, and then enter mysql>insert into usr_preferences values (..,...,..) --- Kapil Dhawan <sersavvy(a)hotmail.com> wrote: > Hi Dave > > Thx for the answer. I am trying avpops module for > failure_route with serial > forking directive with usr_preferences table. But i > am getting several > issues with it. Here are the logs and part of my > ser.cfg > > fork=yes > > > modparam("avpops","avp_url","mysql://root:root@localhost/ser") > modparam("avpops","avp_table","usr_preferences") > modparam("avpops","avp_aliases","serial_fork1=i:665") > modparam("avpops","uuid_column","uuid") > modparam("avpops","username_column","username") > modparam("avpops","domain_column","domain") > modparam("avpops","attribute_column","attribute") > modparam("avpops","value_column","value") > modparam("avpops","type_column","type") > > route { > > ........... > > append_hf("P-hint: usrloc applied\r\n"); > > avp_write("sip:101@192.168.1.1", > "$serial_fork1"); > avp_write("sip:102@192.168.1.1", > "$serial_fork1"); > avp_write("sip:103@192.168.1.1", > "$serial_fork1"); > > route(1); > } > > route[1] > { > # send it out now; use stateful forwarding > as it works reliably > # even for UDP2TCP > > # intercept failure replies > t_on_failure("2"); > > if (!t_relay()) { > sl_reply_error(); > }; > } > > failure_route[2] > { > # use the first element of the list (if any) > and delete it from list > if (avp_pushto("$ruri", "$serial_fork1")) > { > append_branch(); > avp_delete("$serial_fork1"); > t_on_failure("2"); > t_relay(); > } > } > > > > But i am getting several issues. I have three UA's > set 101,102,103. I am > calling from some other domain on 100 which is > aliased to 101. I expect that > if 101 is free call shud come to it. Thats not > happening. Then if 101 and > 102 are busy or some other issue. call should reach > last UA. I didn't get > any proper example to add AVP's to usr_preferences > table. Can you give me > some example for it. > > > > > >From: Dave <ddx66(a)yahoo.com> > >To: serusers(a)lists.iptel.org > >Subject: Re: [Serusers] Call Hunting > >Date: Tue, 31 May 2005 08:30:54 -0700 (PDT) > > > >Yes it is. You need to use the AVPOPS module and > the > >failure_route directive to make it happen. Look at > the > >avpops docs and in there you will see an example > >Dave > >--- Kapil Dhawan <sersavvy(a)hotmail.com> wrote: > > > Hi List > > > > > > Is it possible to do Call Hunting with SER. > > > > > > > >_________________________________________________________________ > > > Adore Arshad Warsi? > > > http://server1.msn.co.in/sp05/iifa/ Think he is > the > > > funniest? > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > >__________________________________ > >Yahoo! Mail > >Stay connected, organized, and protected. Take the > tour: > >http://tour.mail.yahoo.com/mailtour.html > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > _________________________________________________________________ > Think Rani is the best? > http://server1.msn.co.in/sp05/iifa/ Make sure she > wins the award. > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] RE: Re: radiusclient
by Kamran Ahmad 02 Jun '05

02 Jun '05

now i am getting error when starting ser but this </usr/local/lib/ser/modules/auth_radius.so> is already there Jun 2 13:10:31 SER ser[1864]: INFO: signal 15 received Jun 2 13:11:13 SER ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/auth_radius.so>: libradiusclient.so.2: cannot open shared object file: No such file or directory Jun 2 13:11:13 SER ser: parse error (26,13-14): failed to load module Jun 2 13:11:13 SER ser: set_mod_param_regex: No module matching auth_radius found | Jun 2 13:11:13 SER ser: parse error (49,18-19): Can't set module parameter Jun 2 13:11:13 SER ser: set_mod_param_regex: No module matching auth_radius found | Jun 2 13:11:13 SER ser: parse error (50,18-19): Can't set module parameter Jun 2 13:11:13 SER ser: parse error (192,37-38): unknown command, missing loadmodule? Jun 2 13:11:13 SER ser: parse error (223,37-38): unknown command, missing loadmodule? __________________________________ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

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[Serusers] SER not reply
by Sandra Donaire 02 Jun '05

02 Jun '05

Hi!! I've been installed a server yet in a red hat 7 that runs ok. Now I try install a new SER in a fedora core 3 with the same config. I've installed mysql..php...httpd.... Ok, the problem is that I run the server, and apparently runs ok, it load all modules ok, an it's listening in the port correctly, etc. But it doesn't know when arrive request, I see that those arrive correctly with ethereal, but the SER doesn't do any. What can be the problem? Thanks at all. -- Sandra Donaire Arroyo Telefónica I+D División de "Tecnologías de acceso a redes IP"

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[Serusers] configuration files
by Llanos Serna García-Conde 02 Jun '05

02 Jun '05

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[Serusers] configuration files
by Llanos Serna García-Conde 02 Jun '05

02 Jun '05

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RE: [Serusers] Re: radiusclient
by Kamran Ahmad 02 Jun '05

02 Jun '05

i want to install freeradius on 2nd mechine. i have two mechines one for ser and 2nd mechine for freeradius any comments Kamran Ahmad --- Shaikat Mahmud <shaikat(a)stitel.com> wrote: > Hi, > > For radclient exe, you have to install freeradius. > It is not a part of radiusclient. > > shaikat > > > -----Original Message----- > From: Kamran Ahmad [mailto:p_kami@yahoo.com] > Sent: Thursday, June 02, 2005 1:13 PM > To: Shaikat Mahmud; serusers(a)lists.iptel.org > Subject: RE: [Serusers] Re: radiusclient > > > thanks for reply but my problem is that i dont have > radiusclient-0.3.2.tar.gz as recommended in this > document. i tried radiusclient-0.3.3.tar, > radiusclient-0.4.8.tar.gz. but after compiliation > > there is execuatable "radiusclient" but > this document is saying > > #radclient -f digest localhost auth <shared_secret> > > thats the problem > > thanks > Kamran Ahmad > > --- Shaikat Mahmud <shaikat(a)stitel.com> wrote: > > > Hi, > > > > Please read the radius how to. > > > http://www.iptel.org/ser/doc/ser_radius/ser_radius.html > > It works fine. > > > > Shaikat > > > > -----Original Message----- > > From: serusers-bounces(a)lists.iptel.org > > [mailto:serusers-bounces@lists.iptel.org] On > > Behalf Of Kamran Ahmad > > Sent: Thursday, June 02, 2005 12:51 PM > > To: serusers(a)lists.iptel.org > > Subject: [Serusers] Re: radiusclient > > > > hello > > > > can any one tell me how to install radclient > > > > i tried radiusclient-0.3.3.tar , > > radiusclient-0.4.8.tar.gz but nothing working > > > > > > Kamran > > > > > > > > __________________________________ > > Discover Yahoo! > > Use Yahoo! to plan a weekend, have fun online and > > more. Check it out! > > http://discover.yahoo.com/ > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > __________________________________ > Discover Yahoo! > Find restaurants, movies, travel and more fun for > the weekend. Check it out! > > http://discover.yahoo.com/weekend.html > > > __________________________________ Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out! http://discover.yahoo.com/mobile.html

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RE: [Serusers] Re: radiusclient
by Kamran Ahmad 02 Jun '05

02 Jun '05

thanks for reply but my problem is that i dont have radiusclient-0.3.2.tar.gz as recommended in this document. i tried radiusclient-0.3.3.tar, radiusclient-0.4.8.tar.gz. but after compiliation there is execuatable "radiusclient" but this document is saying #radclient -f digest localhost auth <shared_secret> thats the problem thanks Kamran Ahmad --- Shaikat Mahmud <shaikat(a)stitel.com> wrote: > Hi, > > Please read the radius how to. > http://www.iptel.org/ser/doc/ser_radius/ser_radius.html > It works fine. > > Shaikat > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On > Behalf Of Kamran Ahmad > Sent: Thursday, June 02, 2005 12:51 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] Re: radiusclient > > hello > > can any one tell me how to install radclient > > i tried radiusclient-0.3.3.tar , > radiusclient-0.4.8.tar.gz but nothing working > > > Kamran > > > > __________________________________ > Discover Yahoo! > Use Yahoo! to plan a weekend, have fun online and > more. Check it out! > http://discover.yahoo.com/ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > __________________________________ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html

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[Serusers] Re: radiusclient
by Kamran Ahmad 02 Jun '05

02 Jun '05

hello can any one tell me how to install radclient i tried radiusclient-0.3.3.tar , radiusclient-0.4.8.tar.gz but nothing working Kamran __________________________________ Discover Yahoo! Use Yahoo! to plan a weekend, have fun online and more. Check it out! http://discover.yahoo.com/

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[Serusers] Processing REFER Messages
by Java Rockx 02 Jun '05

02 Jun '05

Hi All. I've got a question about handling REFER messages. Below is a REFER message that my Grandstream GXP2000 phone generated when I transferred a call to another SIP phone on the same SER proxy. The problem is that SER did a lookup('location') and couldn't find the phone number in the subscriber table, so it routed the message to the PSTN gateway. SER should have looked at the Refer-To header to get the actual destination of the message. How can I get SER to handle this message correctly? Should REFER messages hit the lookup('location') function in my ser.cfg script? NOTE: my SER proxy sits behind a Cisco 3600 so the ALG functionality has altered the IP. The message shows my SER server as IP address 10.3.0.221<http://10.3.0.221> Regards, Paul U 2005/06/01 14:22:24.073518 66.90.46.29:5060 <http://66.90.46.29:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> REFER sip:4075552279@66.243.109.99:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.130.139 <http://172.31.130.139> ;branch=z9hG4bK09ff837994d9e898. Route: <sip:4075552279@10.3.0.221:5060;nat=yes;ftag=204ed7cd9666ebba;lr>. Route: <sip:216.229.127.60:5060 <http://216.229.127.60:5060>;lr>. From: 3212514238 <sip:3212514238@66.243.109.99:5060>;tag=204ed7cd9666ebba. To: sip:4075552279@66.236.245.100;tag=EB25EC-B85. Contact: <sip:3212514238@172.31.130.139>. Supported: replaces. Refer-To: <sip:3212814820@sip.mycompany.com>. Referred-By: <sip:3212514238@172.31.130.139>. Call-ID: 13195134-3326638548-685571(a)66.243.109.99. CSeq: 51906 REFER. User-Agent: Grandstream GXP2000 1.0.1.8 <http://1.0.1.8>. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Length: 0.

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[Serusers] SER Billing Tutorial
by Felipe Martins 01 Jun '05

01 Jun '05

Hi everybody, Do anybody know where can I find a good SERVoIP Billing tutorial ? I need to generate all the bills in SER. Do anybody know if there is a tutorial for that , I mean, how to generate bills in SER ? Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

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[Serusers] Video Conference tool?
by Stefano 01 Jun '05

01 Jun '05

Hi all, does someone know if there is a free video conferencing tool that works with ser, like sems does with audio call conference, or in a similar way? Thank you vary much Stefano __________________________________________________________________ TISCALI ADSL 1.25 MEGA a soli 19.95 euro/mese Solo con Tiscali Adsl navighi senza limiti di tempo a meno di 20 euro al mese e in piu' telefoni senza pagare il canone Telecom. Scopri come http://abbonati.tiscali.it/adsl/sa/1e25flat_tc/

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Re: [Serusers] sems IP
by Llanos Serna García-Conde 01 Jun '05

01 Jun '05

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[Serusers] Timeout for provisional replies
by Andreas Granig 01 Jun '05

01 Jun '05

Hello, Currently we've the folling tm-timeouts set: modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 180) In my understanding the first timer hits if no final reply for a request (except INVITE) or an ACK for a neg. INVITE-reply arrives within 30 seconds; the second timer hits if no final reply arrives for an INVITE after a provisional reply (the first one?) is received within 180 seconds. But is there a timer for the first provisional reply for an INVITE? I ask because I want to detect if a contact is unreachable. I'd assume that a device answers with "100 trying" (or immediately with 180/183) within some seconds, but which timer do I have to set? Thanks, Andy

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[Serusers] searching for package ser-mysql
by l9ayd amadius 01 Jun '05

01 Jun '05

Hi everyone, Please can someone send me by email the package ser-mysql if he has it or indicate me another site to dowload from it.That's because i can't access the ftp server of :www.iptel.org/ser.I need it urgently.In fact,I need the sql package of ser-mysql for SER version 8.9 and linux redhat 9. Thanks very much in advance!! Best regards!! _________________________________________________________________ Testez le nouveau MSN Messenger ! http://search.msn.fr/

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[Serusers] WG: rpid again
by Dirk Willbrandt 01 Jun '05

01 Jun '05

Hi list ! I have a Problem with the appending of an rpid with the functions avp_load_radius(), append_rpid_hf(). In my Radius DB i have an SIP-AVP field which is filled with the Value rpid:[NUMBER] When i run manuall the Radius Reply-Query i get the correct value for SIP-AVP. Now, i reconfigured my ser.cfg so that an avp_load_radius() is executed after an incoming INVITE. My Radius is answering on the command and sends the correct SIP-AVP Value for the User. Now my Problem: I can't append the RPID or anything else to the SIP-Header after the answer from the Radius. So the call is going out without the rpid in the Header. Can anyone help me ? Here is the Configure-Snip: if (method=="INVITE") { # here comes the invite if (!radius_proxy_authorize("")) { # for the digest avp_load_radius it seems that i must use an proxy auth proxy_challenge("","0"); # and an proxy challenge if(!avp_load_radius("digest")) { # here i get the SIP-AVP from the radius log(1,"Failure with the AVP-Value\n"); # if it fails i'll get a message break; # go out } avp_print(); # print all avp's (this command is never executed :( ) append_hf ("Remote-Party-ID: 111222333\r\n"); # append this silly row - it is never executed append_rpid_hf("", ";party=calling;id-type=subscriber;screen=yes"); # append the rpid from SIP-AVP - and this command is also never executed break; # go out }; route(3); # go to route block 3 break; # end } I tried also this snippet without the breaks and proxy-commands but it doesn't work. I don't know what is wrong - please help me :s Thanks ! Dirk

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[Serusers] Check if user is already authenticated or registered
by Bastian Schern 01 Jun '05

01 Jun '05

Hello everybody, is it possible to check if a user is already registered or authenticated? I want to allow only registered users call to PSTN destinations. Regards Bastian

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Re: [Serusers] sems IP
by Samuel Osorio Calvo 01 Jun '05

01 Jun '05

SEMS is not directly listening to any IP, it relies on the co-located instance of SER to interact with the "IP world". The topology is the following: [SER A]<--IP interface-->[SER B]<=fifo interface=>[SEMS] SER A is the standard SIP proxy, and you can configure it to forward messages to SER B as it would be another gateway (t_relay, rewrite_uri|hos, |etc..) depending on some rules (such as a predefined prefix in the r-uri as is done in the examples). SER B is the front end for SEMS and has a different config file from SER A. It receives the messages from SER A and forward them to SEMS using t_write_unix() so SEMS can receive the messages. As depicted, the interface between SER B and SEMS is not an IP interface but a FIFO or UNIX socket interface (depending on your config files). You can have SER A, SER B, and SEMS running in the same machine or place SER A in one and SER B+SEMS in another. I hope it makes things clearer, Samuel. Unclassified. >>> Llanos Serna García-Conde <llanosserna(a)hotmail.com> 06/01/05 01:04PM >>> Hi all, I have two instance of SER running, listening to the same port (5060), but to different IP addresses. The SER instance with ser.cfg runs in 172.25.97.19 The SER instance with voicemail.cfg runs in 172.25.97.18 SEMS is running in 172.25.97.19 and I want to change it to run in 172.25.97.18. How can I change that? Thanks, Llanos

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[Serusers] Re: [Serdev] Bug? Inconsistent return of loose_route function
by Cesc Santasusana 01 Jun '05

01 Jun '05

Hi, It is the same fix I applied to my working copy when i started this thread. So, it has been running for a week now, no problems so far. Regards, Cesc Unclassified >>> Jan Janak <jan(a)iptel.org> 05/31/05 03:53PM >>> It should be fixed now, please update your sources and re-try. thanks for the report, Jan. On 24-05-2005 10:42, Cesc Santasusana wrote: > Hi, > > Ok ... so it is a bug ... now how to fix it? > > > static inline int after_loose(struct sip_msg* _m, int preloaded) > > ..... > > if (enable_double_rr && is_2rr(&puri.params)) { > > .... > > if (res > 0) { /* No next route found */ > > DBG("after_loose: No next URI found\n"); > > return (preloaded ? NOT_RR_DRIVEN : RR_DRIVEN); > > or simply > > return (preloaded RR_DRIVEN); > > Cesc > > > > Unclassified > > >>> Jan Janak <jan(a)iptel.org> 05/23/05 07:25PM >>> > Hello, comments inline. > > On 23-05-2005 18:57, Cesc Santasusana wrote: > > Hi, > > > > I found what i think is a bug in the return value of loose_route() ..... > > bear with me because it is a rather long email. > > > > First, the scenario: > > > > phone1 ----- SER1 ----- SER2 ----- phone2 > > > > p1 to s1 and p2 to s2 use UDP > > s1 to s2 uses tcp or tls > > > > The bug comes into place when there is a change of protocol in ser, > > from UDP into TCP or TLS. > > We do not use a preloaded route set from the phones. > > What ser does is it adds 2 route headers, using the r2 parameter. > > Route: <route1>, <route2> > > > > CASE A: -------------------------------------------- > > In this scenario, say p1 places a call to p2, going through s1 and s2, > > creating the route set with the record-route. > > Now, let's say p2 generates the BYE message. > > When the BYE gets to s1, the last hop, in the config we do: > > if( loose_route () ) { > > t_relay(); > > } > > But, and here is the inconsistency, loose_route (remember we are using > > TCP/TLS, so there are 2 routes in one header field) returns FALSE. > > > > CASE B: -------------------------------------------- > > If this experiment is repeated using ALL UDP, the route headers when they > > get to s1 contain only one route, and the call to loose_route() returns ..... TRUE! > > > > > > Unless the config file does not allow blind sip message relaying, both cases end > > up working ... the BYE gets to the final destination. But if the config file does not > > accept relaying ... the request-uri in the BYE is not taken as a local domain ... and > > thus the message is rejected ... (To Alex Mack: could this be the solution to your problem? > > the BYEs not reaching the phone? add some log output and check it ... ) > > > > ================================================================ > > So, this is not consistent. Now, what is the solution? > > I think that both case A and B should return the same, but what, true or false? > > Yes, they should return the same value in this case and it should be > true (see below). It looks like you discovered a bug. > > > As I see it, the solution causing less trouble is that both return true. So, the last proxy > > considers this BYE as being loose_routed. > > For this ... change the code in: > > rr::loose_route.c:: static inline int after_loose(struct sip_msg* _m, int preloaded) > > ..... > > if (enable_double_rr && is_2rr(&puri.params)) { > > .... > > if (res > 0) { /* No next route found */ > > DBG("after_loose: No next URI found\n"); > > return (preloaded ? NOT_RR_DRIVEN : RR_DRIVEN); > > or simply > > return (preloaded RR_DRIVEN); > > > > > > And now ... the other option is that both return false ... but this would make big changes. > > Correct me if i am wrong, but wouldn't this be the compliant solution? > > Considering that the r-uri is not part of the route set, when the last proxy processes > > all route headers and there are no more left, there is no loose routing done. The > > destination is taken from the r-uri, not the route set ... thus loose_route should return > > false. > > Or I am getting it all wrong? rfc and goal of loose_route() ? > > The meaning of the return value of loose_route is as follows (or > should be): > > 1 - Record routing was used to route the message > 0 - Record routing was not (usually requests establishing a dialog or > transactional requests). > > So the function should return true in the mentioned case, because > record routing was used although the message does not contain any > Route header fields anymore. > > In other words the purpose of loose_route return value is to distinguish > mid-dialog requests (such as ACK or BYE) from dialog-establishing > requests (such as INVITE). The reason why we have to do this is that we > typically do not apply all the logic in the script to mid-dialog > requests (because you should not change the request-uri of the request > anymore once the dialog has been established -- changes caused by > loose_route do not count). > > You may also wonder why don't we use the contents of the Request-URI > for this -- there should be the Contact value of phone1 in the > Request-URI in mid-dialog requests and the value should not match > uri==myself so all we need to do is to enclose all the processing > logic within if (uri == myself) condition. > > This would work in a perfect world without the need to be backwards > compatible with strict routers. In our world we have to be backwards > compatible with strict routers and we have to deal with broken > implementations too (that strip ;lr parameters). Due to backwards > compatibility with strict routers it can happen that the Request-URI > will contain the URI of the proxy that is currently processing the > request and subsequent if (uri==myself) would be true. > > Jan.

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[Serusers] sems IP
by Llanos Serna García-Conde 01 Jun '05

01 Jun '05

1 0

[Serusers] ser_fifo permissions
by Llanos Serna García-Conde 01 Jun '05

01 Jun '05

1 0

[Serusers] Re: Fwd: Re: error in SQL query
by Christoph Hehl 01 Jun '05

01 Jun '05

Now I get this errormessages: Parse error: parse error, unexpected T_STRING in /srv/www/htdocs/serweb_2004-07-27/html/config.php on line 217 Warning: Cannot modify header information - headers already sent by (output started at /srv/www/htdocs/serweb_2004-07-27/html/config.php:217) in /srv/www/htdocs/serweb_2004-07-27/html/page.php on line 7 Warning: Cannot modify header information - headers already sent by (output started at /srv/www/htdocs/serweb_2004-07-27/html/config.php:217) in /srv/www/htdocs/serweb_2004-07-27/html/page.php on line 8 Warning: Cannot modify header information - headers already sent by (output started at /srv/www/htdocs/serweb_2004-07-27/html/config.php:217) in /srv/www/htdocs/serweb_2004-07-27/html/page.php on line 9 Warning: Cannot modify header information - headers already sent by (output started at /srv/www/htdocs/serweb_2004-07-27/html/config.php:217) in /srv/www/htdocs/serweb_2004-07-27/phplib/session.inc on line 462 Warning: Cannot modify header information - headers already sent by (output started at /srv/www/htdocs/serweb_2004-07-27/html/config.php:217) in /srv/www/htdocs/serweb_2004-07-27/phplib/session.inc on line 463 Warning: Cannot modify header information - headers already sent by (output started at /srv/www/htdocs/serweb_2004-07-27/html/config.php:217) in /srv/www/htdocs/serweb_2004-07-27/phplib/session.inc on line 464 Warning: Cannot modify header information - headers already sent by (output started at /srv/www/htdocs/serweb_2004-07-27/html/config.php:217) in /srv/www/htdocs/serweb_2004-07-27/phplib/session.inc on line 465 Warning: Cannot modify header information - headers already sent by (output started at /srv/www/htdocs/serweb_2004-07-27/html/config.php:217) in /srv/www/htdocs/serweb_2004-07-27/phplib/session.inc on line 466 Database error: cannot use database MySQL Error: 1046 (No Database Selected) Session halted.

1 0

[Serusers] voicemail
by Llanos Serna García-Conde 01 Jun '05

01 Jun '05

1 0

[Serusers] SER Question
by diarmuid.oneill24＠mail.dcu.ie 01 Jun '05

01 Jun '05

Hello, I am a student in DCU University in Dublin, Ireland, and i was down loading the SIP Express Router onto a Sun Blade and i cant seem to permanently set the environment variable "SIP_DOMAIN" to my localhost. I use the followng commands in the terminal to set the variable SIP_DOMAIN to the localhost and it works but when i reboot the Sun Blade and type in "#echo ${SIP_DOMAIN}" i dont get back localhost. If you have any solutions to this problem, i would be delighted to hear of them. #SIP_DOMAIN=localhost #export SIP_DOMAIN #echo ${SIP_DOMAIN} Regards, Diarmuid.

2 1

[Serusers] error in SQL query
by Christoph Hehl 01 Jun '05

01 Jun '05

hello. I have a Problem by login in Servweb. If i klick on the login button i get this errormessages: error in SQL query, line: 30 What's wrong? thanks

4 5

[Serusers] SEMs and SER.cfg
by Mr Greg Plater 01 Jun '05

01 Jun '05

Hi does anyone know were I can find a SEMS how to for the ser.cfg section? I am using SER with mysql and mediaproxy. I have SEMS installed and running but I am a little lost with the ser.cfg file. Does anyone have an example of there working SEMS in the ser.cfg file I can look at for ideas? Thank you

2 1

[Serusers] problem when installing mysql
by l9ayd amadius 01 Jun '05

01 Jun '05

Hi everyone, I have installed ser version 0.8.11 and I want to install the mysql package.But the problem is that the installation fails and I have the message: error: Failed dependencies: libmysqlclient.so.10 is needed by ser-mysql-0.8.11-0 ser = 0.8.11 is needed by ser-mysql-0.8.11-0 In fact I use the command : rpm -i ser-mysql-0.8.11-0.i386.rpm Can anyone explain me the source of the problem and how can I resolve it!!!??? Thanks in advance!!!! _________________________________________________________________ Ne cherchez plus, trouvez ! Avec le nouveau MSN Search. http://search.msn.fr/

2 1

[Serusers] sip proxy and realm are set differently, PSTN can't work properly.
by xwang＠cascotec.com 01 Jun '05

01 Jun '05

Hello we did an experiment with SER, rtpproxy and Asterisk on 1 server (voip1). UA has sip proxy set with 'voip1', realm with 'voip2'. We also have 'domain' module installed. Asterisk runs at port 5061. Most of things works fine. We can do 'echo test' with Asterisk by rewriting host:port to 'voip1:5061'. However, when UA calls PSTN, we did the same thing to rewrite 'host, port' to 'voip1:5061'; then t_relay. It turns out Asterisk calls out correctly (the callee phone rings), but can't send response back to UA. The UA goes dead, and has to be reset. Asterisk shows an active 'sip channel' to this UA. It appears that Asterisk can't find the way back to SER. We can't figure out why 'echo test' works fine, but not real PSTN call. Can someone shed some lights on this issue? thanks! steven

2 1

[Serusers] Fwd: Re: error in SQL query
by Christoph Hehl 01 Jun '05

01 Jun '05

Subject: Re: error in SQL query Date: Mittwoch, 1. Juni 2005 00:58 From: Christoph Hehl <mr_schulz(a)hotmail.com> To: Christoph Hehl <mr_schulz(a)hotmail.com> I use SER version: 0.8.14; serweb_2004-07-27 I couldn find the lines. Can you tell me how can I find it correctly..? -------------------------------------------------------

1 0

[Serusers] How to get the via when callforward?
by Charles Wang 01 Jun '05

01 Jun '05

Hi, ALL: I have three UACs(A, B and C) and register to the same SER. A make a call to INVITE B, but B have a callforward setting to forward this call to C. C forwards this call to a PSTN number. Now, I can find two records for INVITE and BYE from acc table(mysql). But how can I find the last forwarder(C) that forwards this call to PSTN? Can I use uac module to rewrite the From: field? But how can I know the uri of C? -- Best Regards Charles

1 0

[Serusers] SER Reports "out of memory"
by Java Rockx 31 May '05

31 May '05

Hi All. I'm using ser-0.9.2 and testing how it works when the location table has many thousands of rows. ser will not start when the location table has many rows. SER loads the first 1344 records and then pukes on the 1345th record with an out of memory error. I've started ser with "-m 512" but that doesn't seem to make any difference. Am I doing something wrong? Also, for my "test" data I simply created a bunch of bogus records in the location table. I did not do anything with the subscriber table. Does this matter? If so, what needs to be done in order to create a large number of "registered" users? Regards, Paul

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[Serusers] req: rtpproxy video public ip patch
by Giovanni Balasso 31 May '05

31 May '05

Hi serusers, I am trying to set up rtpproxy for video sessions, and I found some problems in my particular situation: I am running ser on a machine with private IP behind a router with public IP which forwards SIP and RTP traffic (ports 5060 and 35000-35xxx). In such a situation I have to modify rtpproxy source with the patch posted some times ago by Greger which makes rtpproxy advertise his public IP. In latest cvs versions of rtpproxy video support has been introduced, so I tried to modify the patch to work with latest cvs rtpproxy. Rtpproxy compiles and runs fine as long as I use audio only sessions, but video can't be seen. I suspect the problem is in the patch because when I test with non-patched rtpproxy, video works. Is there anyone with a deeper understandig (greger? maxim?) of rtpproxy internals who dares to provide a working patch for latest rtpproxy? (will this patch be included in cvs head?) thanks a lot ciao PS I am not registered in serdev list and I can't post there, is there anyone willing to forward my message there? -- Giovanni Balasso giaso(a)yahoo.it

2 1

Re: [Serusers] flatstore problems??
by Jan Janak 31 May '05

31 May '05

Thanks for the report, it should be fixed now in CVS. Jan. On 31-05-2005 01:58, Bayan William Towfiq wrote: > I'm running CVS Head. I'm now able to recreate the situation in > approximately 5 minutes by running test calls through the system and > moving + doing flat_rotate every few seconds. > > These are the first messages that come up: > > WARNING: no fork mode > stateless - initializing > acc - initializing > 0(0) Maxfwd module- initializing > textops - initializing > 0(0) Default allow file (/etc/ser/permissions.allow) not found => empty > rule set > 0(0) Default deny file (/etc/ser/permissions.deny) not found => empty > rule set > > 0(30644) ERROR: get_name: memory allocation failure > 0(30644) flat_reopen_connection: Invalid parameter value > 0(30644) BUG: flat_db_insert: Uninitialized connection > 0(30644) ERROR: acc_request: Error while inserting to database > > I added some debugging statements as well and PATHMAX seems to be 4096 > so the input seems to be okay. I think this is a memory leak in > flatstore where the previously allocated names are not freed. > > int flat_rotate_logs(void) calls flat_reopen_connection(ptr) in a loop, > but flat_reopen_connection(ptr) just calls get_name without freeing the > last buffer. > > William > > > Jan Janak wrote: > > >That looks like a bug in flatstore module. Could you tell me what SER > >version are you using ? > > > > Jan. > > > >On 23-05-2005 11:54, William Towfiq wrote: > > > > > >>I couldn't get much debugging info from all of this because it happened > >>somewhat randomly. I just want to see if anyone else has ever > >>encountered similar problems with flatstore. > >> > >>Thanks, > >>William > >> > >>0(12417) entering route[1] > >>0(12417) ERROR: build_req_buf_from_sip_req: out of memory > >>0(12417) ERROR: print_uac_request: no pkg_mem > >>0(12417) ERROR: t_forward_nonack: failure to add branches > >>0(12417) BUG: flat_db_insert: Uninitialized connection > >>0(12417) ERROR: acc_request: Error while inserting to database > >>0(12417) ------------------------------------------- > >> > >>_______________________________________________ > >>Serusers mailing list > >>serusers(a)lists.iptel.org > >>http://lists.iptel.org/mailman/listinfo/serusers > >> > >> >

2 1

[Serusers] Usrloc entries not expiring in memory
by Andres 31 May '05

31 May '05

Hi, Any idea why usrloc entries start having negative values in the Expires Parameter and never seem to go away? We have a ton of users like this: ...Record(0xb57f93f0)... domain: 'location' aor : '305421XXXX(a)maui.telesip.net' ~~~Contact(0xb57ecf50)~~~ domain : 'location' aor : '305421XXXX(a)maui.telesip.net' Contact : 'sip:305421XXXX@205.X.X.X:58061' Expires : -645 <=============CHECK THIS OUT??? q : Call-ID : 'bd58393-5bfdccb9(a)192.168.1.100' CSeq : 1477 replic : 0 User-Agent: 'Linksys/PAP2-2.0.12(LS)' received : '' State : CS_NEW Flags : 0 next : 0xb5bcf2ac prev : (nil) ~~~/Contact~~~~ ~~~Contact(0xb5bcf2ac)~~~ domain : 'location' aor : '305421XXXX(a)maui.telesip.net' Contact : 'sip:305421XXXX@205.X.X.X:57915' Expires : 602 <= SAME USER...HERE HE IS FINE q : Call-ID : 'de2f27c3-22fcf655(a)192.168.1.100' CSeq : 6 replic : 0 User-Agent: 'Linksys/PAP2-2.0.12(LS)' received : '' State : CS_NEW Flags : 0 next : (nil) prev : 0xb57ecf50 ~~~/Contact~~~~ -- Andres Network Admin http://www.telesip.net

1 0

Re: [Serusers] Call Hunting
by Dave 31 May '05

31 May '05

Yes it is. You need to use the AVPOPS module and the failure_route directive to make it happen. Look at the avpops docs and in there you will see an example Dave --- Kapil Dhawan <sersavvy(a)hotmail.com> wrote: > Hi List > > Is it possible to do Call Hunting with SER. > > _________________________________________________________________ > Adore Arshad Warsi? > http://server1.msn.co.in/sp05/iifa/ Think he is the > funniest? > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

2 1

Re: [Serusers] SER Question
by Samuel Osorio Calvo 31 May '05

31 May '05

Environtment variables are set for the current session (terminal). If you want them to alwaya have a preconfigured value, edit .bash_profile (or the appropriate file for your host) and add there the export SIP_DOMAIN=localhost line. Samuel. Unclassified. >>> <diarmuid.oneill24(a)mail.dcu.ie> 05/30/05 12:03PM >>> Hello, I am a student in DCU University in Dublin, Ireland, and i was down loading the SIP Express Router onto a Sun Blade and i cant seem to permanently set the environment variable "SIP_DOMAIN" to my localhost. I use the followng commands in the terminal to set the variable SIP_DOMAIN to the localhost and it works but when i reboot the Sun Blade and type in "#echo ${SIP_DOMAIN}" i dont get back localhost. If you have any solutions to this problem, i would be delighted to hear of them. #SIP_DOMAIN=localhost #export SIP_DOMAIN #echo ${SIP_DOMAIN} Regards, Diarmuid. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Germany-proper
by mohammad 31 May '05

31 May '05

Hi; Germany-proper White route over Sip & H323 is available.Plz contact me off-list for further info. MSN:hezare3@hotmail.com Regards Mohammad

1 0

[Serusers] ROUTE to ANALOG FXO does not work
by Ozan Blotter 31 May '05

31 May '05

What may be wrong with this ? interface: eth0 (192.168.1.0/255.255.255.0) filter: ip and ( port 5060 ) match: . # U 192.168.1.35:5060 -> 192.168.1.9:5060 REGISTER sip:192.168.1.9:5060 SIP/2.0..From: <sip:201@192.168.1.9:5060> ;ta g=3a4fc965-c1a5a2e..To: <sip:201@192.168.1.9>..Call-ID: c0a80123-13c4-3a4fc 962-1f6-5973..CSeq: 127 REGISTER..Via: SIP/2.0/UDP 192.168.1.35:5060 ;branc h=z9hG4bK-3a4fe27e-621634-49b0..Max-Forwards: 70..Contact: <sip:201@192.168 .1.35:5060>..Expires: 60..Authorization: Digest username="201", realm="192. 168.1.9", nonce="429c75d542b41a31c953e92a094347354d6ad422", uri="sip:192.16 8.1.9:5060", response="2aa4ee1d66adff835b9de526ec1511a8", algorithm=MD5..Co ntent-Length:0.... # U 192.168.1.9:5060 -> 192.168.1.35:5060 SIP/2.0 100 Trying..From: <sip:201@192.168.1.9:5060> ;tag=3a4fc965-c1a5a2e. .To: <sip:201@192.168.1.9>..Call-ID: c0a80123-13c4-3a4fc962-1f6-5973..CSeq: 127 REGISTER..Via: SIP/2.0/UDP 192.168.1.35:5060 ;branch=z9hG4bK-3a4fe27e- 621634-49b0..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Lengt h: 0..Warning: 392 192.168.1.9:5060 "Noisy feedback tells: pid=4355 req_sr c_ip=192.168.1.35 req_src_port=5060 in_uri=sip:192.168.1.9:5060 out_uri=sip :192.168.1.9:5060 via_cnt==1".... # U 192.168.1.9:5060 -> 192.168.1.35:5060 SIP/2.0 401 Unauthorized..From: <sip:201@192.168.1.9:5060> ;tag=3a4fc965-c1 a5a2e..To: <sip:201@192.168.1.9>;tag=e35af7ff49e736750e6bc8e19f478153.5fa6. .Call-ID: c0a80123-13c4-3a4fc962-1f6-5973..CSeq: 127 REGISTER..Via: SIP/2.0 /UDP 192.168.1.35:5060 ;branch=z9hG4bK-3a4fe27e-621634-49b0..WWW-Authentica te: Digest realm="192.168.1.9", nonce="429c7707dd0067d0c1c7197a673469b18a56 38ba", stale=true..Server: Sip EXpress router (0.9.2 (i386/linux))..Content -Length: 0..Warning: 392 192.168.1.9:5060 "Noisy feedback tells: pid=4355 req_src_ip=192.168.1.35 req_src_port=5060 in_uri=sip:192.168.1.9:5060 out_u ri=sip:192.168.1.9:5060 via_cnt==1".... # U 192.168.1.35:5060 -> 192.168.1.9:5060 REGISTER sip:192.168.1.9:5060 SIP/2.0..From: <sip:201@192.168.1.9:5060> ;ta g=3a4fc965-c1a5a2e..To: <sip:201@192.168.1.9>..Call-ID: c0a80123-13c4-3a4fc 962-1f6-5973..CSeq: 128 REGISTER..Via: SIP/2.0/UDP 192.168.1.35:5060 ;branc h=z9hG4bK-3a4fe27f-621a24-3029..Max-Forwards: 70..Contact: <sip:201@192.168 .1.35:5060>..Expires: 60..Authorization: Digest username="201", realm="192. 168.1.9", nonce="429c7707dd0067d0c1c7197a673469b18a5638ba", uri="sip:192.16 8.1.9:5060", response="e297c2b9f9a72418140486b9a5d0225e", algorithm=MD5..Co ntent-Length:0.... # U 192.168.1.9:5060 -> 192.168.1.35:5060 SIP/2.0 100 Trying..From: <sip:201@192.168.1.9:5060> ;tag=3a4fc965-c1a5a2e. .To: <sip:201@192.168.1.9>..Call-ID: c0a80123-13c4-3a4fc962-1f6-5973..CSeq: 128 REGISTER..Via: SIP/2.0/UDP 192.168.1.35:5060 ;branch=z9hG4bK-3a4fe27f- 621a24-3029..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Lengt h: 0..Warning: 392 192.168.1.9:5060 "Noisy feedback tells: pid=4354 req_sr c_ip=192.168.1.35 req_src_port=5060 in_uri=sip:192.168.1.9:5060 out_uri=sip :192.168.1.9:5060 via_cnt==1".... # U 192.168.1.9:5060 -> 192.168.1.35:5060 SIP/2.0 200 OK..From: <sip:201@192.168.1.9:5060> ;tag=3a4fc965-c1a5a2e..To: <sip:201@192.168.1.9>;tag=e35af7ff49e736750e6bc8e19f478153.95cd..Call-ID: c0a80123-13c4-3a4fc962-1f6-5973..CSeq: 128 REGISTER..Via: SIP/2.0/UDP 192.1 68.1.35:5060 ;branch=z9hG4bK-3a4fe27f-621a24-3029..Contact: <sip:201@192.16 8.1.35:5060>;expires=60..Server: Sip EXpress router (0.9.2 (i386/linux))..C ontent-Length: 0..Warning: 392 192.168.1.9:5060 "Noisy feedback tells: pid =4354 req_src_ip=192.168.1.35 req_src_port=5060 in_uri=sip:192.168.1.9:5060 out_uri=sip:192.168.1.9:5060 via_cnt==1".... [4]+ Stopped ngrep . port 5060 ]0;root@localhost:/usr/local/etc/ser[root@localhost ser]# serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'location' size : 512 table: 0xf50b60f8 d_ll { n : 3 first: 0xf50b8248 last : 0xf50b8108 } ...Record(0xf50b8248)... domain: 'location' aor : '54321' ~~~Contact(0xf50b82a8)~~~ domain : 'location' aor : '54321' Contact : 'sip:54321@192.168.1.201:5060' Expires : 58 q : Call-ID : '1386D1B912DA1212(a)192.168.1.201' CSeq : 967 User-Agent: 'Unknown' received : '' State : CS_DIRTY Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50b8388)... domain: 'location' aor : '12345' ~~~Contact(0xf50b83e8)~~~ domain : 'location' aor : '12345' Contact : 'sip:12345@192.168.1.200:5060' Expires : 35 q : Call-ID : '1442D1B912D91212(a)192.168.1.200' CSeq : 965 User-Agent: 'Unknown' received : '' State : CS_DIRTY Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50b8108)... domain: 'location' aor : '201' ~~~Contact(0xf50bbaa8)~~~ domain : 'location' aor : '201' Contact : 'sip:201@192.168.1.35:5060' Expires : 45 q : Call-ID : 'c0a80123-13c4-3a4fc962-1f6-5973' CSeq : 128 User-Agent: 'Unknown' received : '' State : CS_DIRTY Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ---Domain--- name : 'aliases' size : 512 table: 0xf50b4068 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ===/Domain list=== ]0;root@localhost:/usr/local/etc/ser[root@localhost ser]# cat.[K ser.cfg debug=3 fork=yes log_stderror=no listen=192.168.1.9 port=5060 children=4 dns=no rev_dns=no 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) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if ((method=="BYE" || method=="CANCEL") && !(uri=~"^sip:833*@*")) { setflag(1); } if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); break; } if (uri=~"^sip:053[0-9]*@") { # prefix("201"); rewritehostport("192.168.1.35:5060"); t_relay_to_udp("192.168.1.35","5060"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }]0;root@localhost:/usr/local/etc/ser[root@localhost ser]#

1 0

[Serusers] radiusclient
by Kamran Ahmad 31 May '05

31 May '05

i am trying to follow this http://www.iptel.org/ser/doc/ser_radius/ser_radius.html and using radiusclient-0.3.3.tar.gz on one mechine and freeradius on another but i am unable to fix it radclient is not installed there is radiusclient so i cannot use this root@/usr/local/src# radclient -f digest localhost auth <shared_secret> this thing is working properly on my other mechine but this time radclient is not available any link thanks kamran __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/

1 0

[Serusers] ser not sending ACK back to MGW
by Colin Dalton 31 May '05

31 May '05

Hello, Apologies if this has come up before, but I'm having some problems with SER not forwarding an ACK back to a Cisco AS5300 media gateway. This means that incoming calls break down because the 5300 hasn't received an ACK from the SIP phone. Outgoing calls are fine, because SER is forwarding the 5300's ACK back to the SIP phone correctly. Would anyone know why this is only working in one direction? With incoming calls, the sequence of events is: 1. I dial the PSTN number on my SIP phone. 2. The SIP phone sends the INVITE to SER. 3. SER rewrites the host and forwards it to the Media Gateway (MGW) 4. The MGW brings up an ISDN channel and connects the call 5. When the call is answered, the MGW sends back the 200/OK to SER 6. SER forwards the 200/OK on to the SIP phone and I can converse on the call 7. The SIP phone sends the ACK back to SER, but SER doesn't send it on to the MGW. 8. The MGW waits 19 secs before disconnecting the call because it hasn't received an ACK. In the reverse direction everything works fine, because SER sends on the ACK it receives from the AS5300. My configuration is appended below. Running IOS 12.3(9) on the 5300, the SIP phone is a Siemens Optipoint 400. SER version is 0.8.10 on Fedora Core 3. Thanks, Colin --- config --- # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; if (uri=~"^sip:9[0-9]*@*") { # pstn escape = 9 # strip(9); rewritehostport("159.134.250.247:5060"); log(1,"Forwarding to PSTN/n"); forward(159.134.250.247, 5060); } else { # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; forward(uri:host,uri:port); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; }; }

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[Serusers] Re: [Serdev] Bug? Inconsistent return of loose_route function
by Cesc Santasusana 31 May '05

31 May '05

Hi, Ok ... so it is a bug ... now how to fix it? > static inline int after_loose(struct sip_msg* _m, int preloaded) > ..... > if (enable_double_rr && is_2rr(&puri.params)) { > .... > if (res > 0) { /* No next route found */ > DBG("after_loose: No next URI found\n"); > return (preloaded ? NOT_RR_DRIVEN : RR_DRIVEN); > or simply > return (preloaded RR_DRIVEN); Cesc Unclassified >>> Jan Janak <jan(a)iptel.org> 05/23/05 07:25PM >>> Hello, comments inline. On 23-05-2005 18:57, Cesc Santasusana wrote: > Hi, > > I found what i think is a bug in the return value of loose_route() .... > bear with me because it is a rather long email. > > First, the scenario: > > phone1 ----- SER1 ----- SER2 ----- phone2 > > p1 to s1 and p2 to s2 use UDP > s1 to s2 uses tcp or tls > > The bug comes into place when there is a change of protocol in ser, > from UDP into TCP or TLS. > We do not use a preloaded route set from the phones. > What ser does is it adds 2 route headers, using the r2 parameter. > Route: <route1>, <route2> > > CASE A: -------------------------------------------- > In this scenario, say p1 places a call to p2, going through s1 and s2, > creating the route set with the record-route. > Now, let's say p2 generates the BYE message. > When the BYE gets to s1, the last hop, in the config we do: > if( loose_route () ) { > t_relay(); > } > But, and here is the inconsistency, loose_route (remember we are using > TCP/TLS, so there are 2 routes in one header field) returns FALSE. > > CASE B: -------------------------------------------- > If this experiment is repeated using ALL UDP, the route headers when they > get to s1 contain only one route, and the call to loose_route() returns ..... TRUE! > > > Unless the config file does not allow blind sip message relaying, both cases end > up working ... the BYE gets to the final destination. But if the config file does not > accept relaying ... the request-uri in the BYE is not taken as a local domain ... and > thus the message is rejected ... (To Alex Mack: could this be the solution to your problem? > the BYEs not reaching the phone? add some log output and check it ... ) > > ================================================================ > So, this is not consistent. Now, what is the solution? > I think that both case A and B should return the same, but what, true or false? Yes, they should return the same value in this case and it should be true (see below). It looks like you discovered a bug. > As I see it, the solution causing less trouble is that both return true. So, the last proxy > considers this BYE as being loose_routed. > For this ... change the code in: > rr::loose_route.c:: static inline int after_loose(struct sip_msg* _m, int preloaded) > ..... > if (enable_double_rr && is_2rr(&puri.params)) { > .... > if (res > 0) { /* No next route found */ > DBG("after_loose: No next URI found\n"); > return (preloaded ? NOT_RR_DRIVEN : RR_DRIVEN); > or simply > return (preloaded RR_DRIVEN); > > > And now ... the other option is that both return false ... but this would make big changes. > Correct me if i am wrong, but wouldn't this be the compliant solution? > Considering that the r-uri is not part of the route set, when the last proxy processes > all route headers and there are no more left, there is no loose routing done. The > destination is taken from the r-uri, not the route set ... thus loose_route should return > false. > Or I am getting it all wrong? rfc and goal of loose_route() ? The meaning of the return value of loose_route is as follows (or should be): 1 - Record routing was used to route the message 0 - Record routing was not (usually requests establishing a dialog or transactional requests). So the function should return true in the mentioned case, because record routing was used although the message does not contain any Route header fields anymore. In other words the purpose of loose_route return value is to distinguish mid-dialog requests (such as ACK or BYE) from dialog-establishing requests (such as INVITE). The reason why we have to do this is that we typically do not apply all the logic in the script to mid-dialog requests (because you should not change the request-uri of the request anymore once the dialog has been established -- changes caused by loose_route do not count). You may also wonder why don't we use the contents of the Request-URI for this -- there should be the Contact value of phone1 in the Request-URI in mid-dialog requests and the value should not match uri==myself so all we need to do is to enclose all the processing logic within if (uri == myself) condition. This would work in a perfect world without the need to be backwards compatible with strict routers. In our world we have to be backwards compatible with strict routers and we have to deal with broken implementations too (that strip ;lr parameters). Due to backwards compatibility with strict routers it can happen that the Request-URI will contain the URI of the proxy that is currently processing the request and subsequent if (uri==myself) would be true. Jan.

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[Serusers] Sources to download STUN-server missing
by Franz Edler 31 May '05

31 May '05

Hello, I am missing the files to download myStun from: http://developer.berlios.de/projects/mystun/ Any suggestions? Has the project been removed? Franz

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[Serusers] Logging to mysql?
by Lol Zimmerli 31 May '05

31 May '05

Hello! I would like to output log() and xlog() to mysql instead of syslog; is there a way to achieve it this way? Thx -- # Lol Zimmerli // S y s C o ® // http://www.sysco.ch/ Make sure special cases are truly special. - The Elements of Programming Style (Kernighan & Plaugher)

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[Serusers] invite + options
by João Salgueiro 31 May '05

31 May '05

Hi! i need access to the callee sdp before the connection is established, so i guess i need to do the following: UAa -> sends INVITE SER -> holds the INVITE and sends an OPTIONS to UAb UAb -> responds to OPTIONS SER sends INVITE to UAb is this possible? i need access to the callee sdp before the connection is established. thanks in advance, Joao Salgueiro

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[Serusers] Call Hunting
by Kapil Dhawan 31 May '05

31 May '05

Hi List Is it possible to do Call Hunting with SER. _________________________________________________________________ Adore Arshad Warsi? http://server1.msn.co.in/sp05/iifa/ Think he is the funniest?

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[Serusers] Bad username or password
by Christoph Hehl 30 May '05

30 May '05

Hello. If I will login as Admin I get follow errormessages: Bad username or password Username: admin Password: heslo SER version: 0.8.14; serweb_2004-07-27 In the php.ini i have Register_globals=on Whats wrong...?

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[Serusers] SER Newbie
by Daniel Eboa 30 May '05

30 May '05

Hello list, I'm new in SER, and i have problem setting up the application. I have asterisk pbx using SIP and a billing application for voip using radius authentication. The billing use MSSQL database and the Radius server is under Linux. I installed radiusclient on the same machine with ser-0.8.11-0.rpm. The problem is that when I do: ser start, I have this: 127.0.0.1 [127.0.0.1]:5060 XXX.XXX.XXX.XX [XXX.XXX.XXX.XX]:5060 Aliases: host-XXX-XXX-XXX-XX.interpacket.net:5060 localhost:5060 localhost.localdomain:5060 SER:5060 XX is the IP of the SER machine. But when I do "serctl monitor", I got this: [cycle #: 1; if constant make sure server lives and fifo is on] And nothing happen. This tells me that ser is not running. Can somebody help me with my setup. Thanks. Daniel

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Re: [Serusers] No sound with ser 0.9.0 and mediaproxy 1.2.1
by Greger V. Teigre 30 May '05

30 May '05

If you had it working with mediaproxy 1.3.1, try upgrading it again. Also, there is a debug option for mediaproxy so you can see what happens, but I don't recall what it is now. g-) Roger Lewau wrote: > Hello Greger, > > Thanks for your ideas, however all SIP messages are ok, including SDP > line as you mentioned. There is no firewall blocking UDP>1000 infront > of mediaproxy. The situation is actually that mediaproxy dont relay > RTP packets. Both clients send their packets to mediaproxy to the > ports specified in the RDP line, I can detect them with ngrep. Also > there is no error from the dispatcher since it is not used. > Mediaproxy and SER sits on the same machine... > > I tried using the dispatcher but got the same result as without it. > It must be a configuration issue with mediaproxy, because when I used > the mediaproxy sample ser.cfg it worked quite nicely. However that > was on a previous verison of SER and mediaproxy 1.3.1. Also SIP > conversation is good between SER and clients. It seems there is a > problem in the communication between ser and mediaproxy. Mediaproxy > just listen but do not send... There are sadly enough no entries in > the log that relates to this issue... > > Any more ideas? Or anybody else? > > -----Ursprungligt meddelande----- > Från: Greger V. Teigre [mailto:greger@teigre.com] > Skickat: den 29 maj 2005 09:51 > Till: Roger Lewau; serusers(a)lists.iptel.org > Ämne: Re: [Serusers] No sound with ser 0.9.0 and mediaproxy 1.2.1 > > Firewall in front of mediaproxy with closed ports UDP > 1000? DoS > protection that kicks in (lots of small UDP packets). I know Zyxel > will do that sometimes. > Could also be a problem in the setup of your clients. Do they find a > common codec? > Look in /var/log/messages for errors from proxydispatcher or > mediaproxy. You could have a problem with the communication. > Check the INVITE from ser to callee and the OK from callee to caller. > Both should have an SDP line with m= and then the IP address of your > mediaproxy. g-) > > Roger Lewau wrote: >> Hello list... >> >> I've installed SER 0.9.0 and mediaproxy 1.2.1 and dowloaded the >> getting started scripts (issue 04a) from onsip.org. >> Setup everything as instructed, added a couple of users to the >> database and tried to setup a call between one public and one NATed >> UA. >> Both phones register properly and signalling seems to be working >> fine, I ring each device from the other, problem is there is no >> sound. >> >> So I checked the mediaproxy sessions... and it looks like this >> >> >> Caller Via Called >> Status Duration Codec Type Traffic >> ---------------------------------------------------------------------- >> ------ >> ---------------------------------- >> 82.182.194.x:5004 - 212.247.91.x:35014 - 82.182.194.x:49160 inactive >> 0'07" Unknown Audio 0/0/0 >> >> Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 >> >> I dont really understand the output but I think "inactive" as well as >> "Unknown Audio" is not a good sign. >> >> How ever, I then used Ngrep to see if there are some rtp traffic >> going on and sure enough, there is traffic comming in from both UAs >> but no traffic is leaving the proxy. Strange! >> >> Anyone with some bright idea on why mediaproxy is not relaying the >> sound? >> >> My ser.cfg >> =================================== >> >> debug=3 >> fork=yes >> log_stderror=no >> >> listen=212.247.91.<hidden> # INSERT YOUR IP ADDRESS HERE >> port=5060 >> children=4 >> alias=sip.<somedomain>.com >> dns=no >> rev_dns=no >> >> 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.so" >> loadmodule "/usr/local/lib/ser/modules/uri_db.so" >> loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" >> loadmodule "/usr/local/lib/ser/modules/nathelper.so" >> loadmodule "/usr/local/lib/ser/modules/textops.so" >> loadmodule "/usr/local/lib/ser/modules/avpops.so" >> loadmodule "/usr/local/lib/ser/modules/domain.so" >> loadmodule "/usr/local/lib/ser/modules/permissions.so" >> >> modparam("auth_db|permissions|uri_db|usrloc", "db_url", >> "mysql://ser:heslo@localhost/ser") >> modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", >> "password_column", "password") >> >> #modparam("nathelper", "rtpproxy_disable", 1) #modparam("nathelper", >> "natping_interval", 0) >> >> modparam("mediaproxy","natping_interval", 30) >> modparam("mediaproxy","mediaproxy_socket", >> "/var/run/mediaproxy.sock") >> modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-client >> s") >> modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-client >> s") >> >> modparam("usrloc", "db_mode", 2) >> >> modparam("registrar", "nat_flag", 6) >> >> modparam("rr", "enable_full_lr", 1) >> >> modparam("tm", "fr_inv_timer", 27) >> modparam("tm", "fr_inv_timer_avp", "inv_timeout") >> >> modparam("permissions", "db_mode", 1) >> modparam("permissions", "trusted_table", "trusted") >> >> route { >> # ----------------------------------------------------------------- >> # Sanity Check Section >> # ----------------------------------------------------------------- >> if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too >> Many Hops"); break; }; >> >> if (msg:len > max_len) { >> sl_send_reply("513", "Message Overflow"); break; }; >> >> # ----------------------------------------------------------------- >> # Record Route Section >> # ----------------------------------------------------------------- >> if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP >> ADDRESS HERE >> record_route_preset("212.247.91.<hidden>:5060;nat=yes"); } else if >> (method!="REGISTER") { record_route(); >> }; >> >> # ----------------------------------------------------------------- >> # Call Tear Down Section >> # ----------------------------------------------------------------- >> if (method=="BYE" || method=="CANCEL") { end_media_session(); }; >> >> # ----------------------------------------------------------------- >> # 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 (uri!=myself) { >> route(1); >> break; >> }; >> >> if (uri==myself) { >> if (method=="ACK") { >> route(6); >> break; >> } else if (method=="CANCEL") { >> route(3); >> break; >> } else if (method=="INVITE") { >> route(3); >> break; >> } else if (method=="REGISTER") { >> route(2); >> break; >> }; >> >> lookup("aliases"); >> if (uri!=myself) { >> route(1); >> break; >> }; >> >> if (!lookup("location")) { >> sl_send_reply("404", "User Not Found"); >> break; >> }; >> }; >> log("NO MATCH"); >> route(1); >> } >> >> route[1] { >> >> # ----------------------------------------------------------------- >> # Default Message Handler >> # ----------------------------------------------------------------- >> >> t_on_reply("1"); >> >> if (!t_relay()) { >> >> if (method=="INVITE" || method=="ACK") { >> end_media_session(); >> }; >> >> sl_reply_error(); >> }; >> } >> >> route[2] { >> >> # ----------------------------------------------------------------- >> # REGISTER Message Handler >> # ----------------------------------------------------------------- >> >> sl_send_reply("100", "Trying"); >> >> if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); >> fix_nated_register(); force_rport(); }; >> >> if (!www_authorize("","subscriber")) { www_challenge("","0"); >> break; }; >> >> if (!check_to()) { >> sl_send_reply("401", "Unauthorized"); break; }; >> >> consume_credentials(); >> >> if (!save("location")) { >> sl_reply_error(); >> }; >> } >> >> route[3] { >> >> # ----------------------------------------------------------------- >> # CANCEL and INVITE Message Handler >> # ----------------------------------------------------------------- >> >> if (client_nat_test("3")) { >> setflag(7); >> force_rport(); >> fix_nated_contact(); >> }; >> >> if (method=="INVITE" && !allow_trusted()) { >> >> if (!proxy_authorize("","subscriber")) { >> proxy_challenge("","0"); >> break; >> } else if (!check_from()) { >> sl_send_reply("403", "Use From=ID"); >> break; >> }; >> >> consume_credentials(); >> }; >> >> lookup("aliases"); >> if (uri!=myself) { >> route(1); >> break; >> }; >> >> if (uri=~"^sip:[+|00][0-9]*@") { # International PSTN inleds antingen >> med + eller 00 route(4); break; }; >> >> if (!lookup("location")) { >> >> if (uri=~"^sip:0[1-9]*@") { # Domestic PSTN startar med 0 >> route(4); >> break; >> }; >> >> sl_send_reply("404", "User Not Found"); break; }; >> >> if (method=="CANCEL") { >> route(1); >> break; >> }; >> >> route(5); >> route(1); >> } >> >> route[4] { >> >> # ----------------------------------------------------------------- >> # PSTN Handler >> # ----------------------------------------------------------------- >> >> rewritehost("192.0.2.245"); # INSERT YOUR PSTN GATEWAY IP ADDRESS >> >> avp_write("i:45", "inv_timeout"); >> >> route(5); >> route(1); >> } >> >> route[5] { >> >> # ----------------------------------------------------------------- >> # RTP Proxy Enabler >> # ----------------------------------------------------------------- >> >> if (isflagset(6) || isflagset(7)) { >> use_media_proxy(); >> }; >> } >> >> route[6] { >> >> # >> ---------------------------------------------------------------------- >> -- >> # ACK Handler # >> ---------------------------------------------------------------------- >> -- >> >> # >> ---------------------------------------------------------------------- >> -- >> # Aliases Section # >> ---------------------------------------------------------------------- >> -- >> lookup("aliases"); >> if (uri!=myself) { >> route(1); >> break; >> }; >> >> lookup("location"); >> >> route(1); >> } >> >> >> onreply_route[1] { >> >> if ((isflagset(6) || isflagset(7)) && >> (status=~"(180)|(183)|2[0-9][0-9]")) { >> >> if (!search("^Content-Length:\ +0")) { >> use_media_proxy(); >> }; >> }; >> >> if (client_nat_test("1")) { >> fix_nated_contact(); >> }; >> } >> >> ================================== >> >> My mediaproxy.ini >> ================================== >> ; >> ; Configuration file for MediaProxy >> ; >> >> [Dispatcher] >> >> [MediaProxy] >> start = yes >> socket = /var/run/mediaproxy.sock >> group = ser >> ;listen = None >> allow = Any >> proxyIP = 212.247.91.<hidden> >> portRange = 35000:65000 >> TOS = 0xb8 >> idleTimeout = 60 >> holdTimeout = 3600 >> ;forceClose = 0 >> accounting = off >> >> [Accounting] >> user = dbuser >> password = dbpass >> host = dbhost >> database = radius >> table = radacct >> >> ================================== >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] H.350 directory services
by Tarun Abhichandani 30 May '05

30 May '05

Hi Marc: We have H.350 authentication structure on our openLDAP server. We use it to authenticate users from our home-grown SIP software. Let me know if you need any specific information. Will try my best to provide you, Tarun ________________________________ From: serusers-bounces(a)lists.iptel.org on behalf of Marc Sutter Sent: Thu 5/26/2005 8:45 AM To: serusers(a)lists.iptel.org Subject: [Serusers] H.350 directory services Hi all, I'm looking for a solution to use H.350 directory services to authentificate UA's through ser. Is ther any ldap module for authentification out there ? Thanks for your help Marc _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] serctl moni question.
by Ricardo Martinez 30 May '05

30 May '05

Hello. In the serctl moni appears a "failure" indicator. What represent this indicator? Could be something related to exec_dst() function? Thanks! Ricardo Martinez-.

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Re: [Serusers] Ser+Nat +Polycom phones
by Greger V. Teigre 30 May '05

30 May '05

http://lists.iptel.org/pipermail/serusers/2005-January/014688.html harry gaillac wrote: > Where can I get your patch ? > > Harry > --- "Greger V. Teigre" <greger(a)teigre.com> a écrit : > >> Running rtpproxy on a NATed box is a problem (you >> need a patch I posted a >> while back to announce the correct public IP). >> Running all daemons on the >> same server with a public IP is NOT a problem. >> g-) >> harry gaillac wrote: >>> Hello, >>> >>> SER+nathelper+rtpproxy run on the same box that >> nat. >>> Is it a problem ? >>> >>> Harry >>> >>> >>> --- "Greger V. Teigre" <greger(a)teigre.com> a écrit >>> >>> >>>> That's strange. We are using some Polycom 300 and >>>> they work ok. No STUN >>>> support and sometimes a very erratic registration >>>> pattern, but except from >>>> that, they seem to perform ok. They interoperate >>>> with sipuras, >>>> grandstreams, sjphone, cisco gw, xlite, etc. >>>> g-) >>>> >>>> Juha Heinanen wrote: >>>>> harry, >>>>> >>>>> we tried polycom video phones a month or so ago >>>> and their sip >>>>> implementation was worst i had seen so far. the >>>> phones didn't support >>>>> digest authentication and we were also hit the >>>> problem you mentioned >>>>> with domain in request-uri. i suggest to junk >>>> those phones before >>>>> their sip implementation is closer to rfc3261. >>>>> >>>>> -- juha >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>> >>> >>> >>> >>> >>> >>> >>> >> > _____________________________________________________________________________ >>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace >> de stockage pour vos >>> mails, photos et vidéos ! Créez votre Yahoo! Mail >> sur >>> http://fr.mail.yahoo.com >> >> > > > > > > > > _____________________________________________________________________________ > Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos > mails, photos et vidéos ! Créez votre Yahoo! Mail sur > http://fr.mail.yahoo.com

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[Serusers] Ser+Nat +Polycom phones
by harry gaillac 30 May '05

30 May '05

Hello, I try to configure Ser and polycom phones behind nat. phone 1 ------- | ser | ----- internet phone 2 ------- Ser listen on private network and internet. Polycom phones send public ip of Ser in SIP header. According to sip_call file (ethereal) ser know that phones are behind nat (lines 2,3,6,8) however when phone1 invite phone2 "404 Not found" message it sent by Ser . If Ser Know that phones are behind nat How can I fix that problem in ser.cfg . Thanks for your help. Harry _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

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[Serusers] e164 exchanges, lookups and what happens when they are down
by Iqbal 30 May '05

30 May '05

Hi Had a strange problem this morning, freenum seemed to be down, hence the lookups in ser just sort of hung, and the looped round again, but no calls were being made, it seems as if no response, is received then it "hangs" no in a literal sense. Anyonw experienced this, or knows of a way around, it works fine if it recieves a lookup failure , i.e if the data is not in there. Iqbal

3 2

[Serusers] Re: error in compile ser-0.9.0
by Kamran Ahmad 30 May '05

30 May '05

hello i already have fifo="/tmp/ser_fifo" in my ser.cfg there is some problem in compile ------------------------------------------------------ make[1]: Entering directory `/ser-0.9.0/modules/lcr' ../../Makefile.rules:77: fifo.d: No such file or directory ../../Makefile.rules:77: lcr_mod.d: No such file or directory make[1]: Leaving directory `/ser-0.9.0/modules/lcr' make[1]: Entering directory `/ser-0.9.0/modules/lcr' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -I/opt/axxany/siprouter/radiusclient/include -DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c fifo.c -o fifo.o gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -I/opt/axxany/siprouter/radiusclient/include -DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c lcr_mod.c -o lcr_mod.o gcc -shared -L/opt/axxany/siprouter/radiusclient/lib -Wl,-O2 -Wl,-E fifo.o lcr_mod.o -o lcr.so ----------------------------------------------------- >Do you have the following line in your ser.cfg: > >fifo=/tmp/ser_fifo > > >Try to configure it, and see what happens > > >On Tue, 24 May 2005 06:43:04 -0700 (PDT) >Kamran Ahmad <p_kami(a)yahoo.com> wrote: > >> fifo=/tmp/ser_fifo > >-- >Felipe Martins >Mundivox Communications >Tecnologia e Projetos >fmartins(a)mundivox.com > >Tel.: +55 +21 +3820 8839 >Cel.: +55 +21 +9823 8602 >Fax.: +55 +21 +3820 8844 >www.mundivox.com __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/

2 1

[Serusers] SER on multihomed host
by Ravi 30 May '05

30 May '05

Hi, I am running SER on a host with 2 network interfaces. The behavior I observe is that when SER receives a message say REGISTER on interface 1 it sends the 200OK on interface 2. Due to the way the network has been configured, this 200OK never makes it back to the UA which sent the REGISTER. So I want to make sure that SER sends the response back on the same interface it got the message on. How do I do this? I cannot disable the second network interface. In ser.cfg I set "listen" to the IP address of the interface I want to use; I also set "mhomed" to 1, still no luck. Any suggestions are highly welcome. thanks, -Ravi.

2 1

[Serusers] How to generate Bills - Trabas VoIP Billing software
by Felipe Martins 30 May '05

30 May '05

Hi everybody, I'm thinking of implementing Trabas VoIP Billing Software (www.trabas.com) in order to manage my VoIP billing. I've been to many voip billing software, nearly all I found very good but also very expensive ... Because of that, I would like to make some questions. 1. Has anybody have implemented such software ? 2. How to generate bills (CDR Data) in SER ? Is there a howto or tutorial for that at the internet ? 3. I use SER with MySQL, does Trabas work with SER and MySQL ? Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

1 0

[Serusers] Questions about domain field in Proxy-Authenticate
by Paul Belanger 30 May '05

30 May '05

>From RFC3261: 20.27 Proxy-Authenticate A Proxy-Authenticate header field value contains an authentication challenge. The use of this header field is defined in [H14.33]. See Section 22.3 for further details on its usage. Example: Proxy-Authenticate: Digest realm="atlanta.com", domain="sip:ss1.carrier.com", qop="auth", nonce="f84f1cec41e6cbe5aea9c8e88d359", opaque="", stale=FALSE, algorithm=MD5 My question revolves around the domain="sip:ss1.carrier.com" field. I notice that SER does not use the option, however I have another proxy that does. I have searched for information about this field, but not able to get the information I need. Why would you use a realm and domain field at the same time? Is 1 preferred to another? Should both be used? Any help would be great. PB

2 1

[Serusers] flatstore problems??
by William Towfiq 30 May '05

30 May '05

I couldn't get much debugging info from all of this because it happened somewhat randomly. I just want to see if anyone else has ever encountered similar problems with flatstore. Thanks, William 0(12417) entering route[1] 0(12417) ERROR: build_req_buf_from_sip_req: out of memory 0(12417) ERROR: print_uac_request: no pkg_mem 0(12417) ERROR: t_forward_nonack: failure to add branches 0(12417) BUG: flat_db_insert: Uninitialized connection 0(12417) ERROR: acc_request: Error while inserting to database 0(12417) -------------------------------------------

2 1

[Serusers] Please help - ser won't start
by mma 30 May '05

30 May '05

hi, i have the same probleme with jabber module : 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) XJAB:mod_init: error - database module not found 0(0) init_mod(): Error while initializing module jabber ERROR: error while initializing modules did you find a solution to this problem please i need help regards

1 0

[Serusers] rtpproxy & video
by Giovanni Balasso 30 May '05

30 May '05

Hi all, I would like to know if anyone has some experience (maybe some dev?) with rtpproxy and video. I have done some testing but with no result, audio works like a charm but it looks like video is unsupported. I hope someone can tell me I'm wrong and point me to the right way to set up video. I set up ser.cfg routing for video the same way it is set up for audio, is this correct? (it should as SIP is media independent...) I read in some past message that video support was added to cvs rtpproxy so i downloaded it and made some test with no luck, so I was wondering wether the problem is in my cfg or in rtpproxy. My ser.cfg is from onsip.org with some customization, I was testing it with eyebeam which sends another INVITE when starting video, this INVITE enters loose_route section and then force_rtp_proxy("l") command, rtpproxy responds but I can't get video. Any comments? thanks a lot ciao -- Giovanni Balasso giaso(a)yahoo.it

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[Serusers] migration 0.8.14 -> 0.9.0 and SQL broken
by mma 30 May '05

30 May '05

hi, i have the same probleme with jabber module : 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) XJAB:mod_init: error - database module not found 0(0) init_mod(): Error while initializing module jabber ERROR: error while initializing modules did you find a solution to your probleme; please i need help Regards

1 0

[Serusers] Re: routing and accounting in SER
by Giovanni Balasso 30 May '05

30 May '05

Alle 13:03, lunedì 30 maggio 2005, Ozan Blotter ha scritto: > Dear Giovanni Balasso, > > Thanks for you reply, but some parts i'm confused... Because i have two > setflag(1) in my ser.cfg, which one i need to remove or add, if i delete > the one on top of route, BYE messages are not recorded into database. Can > you please tell me how can i ? > You should use one setflag when the call starts (on INVITE) and one when the call ends (BYE or CANCEL), so I suggest - remove setflag on top of route, then add if ((method=="BYE" || method=="CANCEL") && !(uri=~"^sip:833*@WHATEVER")) { setflag(1); } between record route and loose route sections (as you record_route your calls, BYEs or CANCELs are relayed in loose_route section, so you need to set acc flag before relaying BYE in loose_route) - you can safely remove || method=="BYE" || method=="CANCEL" and keep only method=="INVITE" in uri==myself section (remember BYE and CANCEL hit loose_route!), and add uri check as above to the following setflag So your routing logic now should work like this INVITE --> call for 833* ? --> if not: flag 1 (call started) --> call relayed --> BYE --> call for 833*? --> if not: flag 1 (call ended) --> SIP session closed ciao -- Giovanni Balasso giaso(a)yahoo.it

1 0

[Serusers] Accounting & Dial Plan
by Ozan Blotter 30 May '05

30 May '05

Dear Friends, A friend of me is writing Basic Billing for SER :) in PHP. I'm gonna place it somewhere so anyone can add remove features to it and repost it. Will let all of you know and download it. But for now i need to create a dial plan. I do not want to make accounting for numbers beginning with 833 prefix, is there a way for it ? Another thing is , i've tried with so many things but always give errors in config. I have 2 x 4 Port Analog GSM Gateways, first IP is 192.168.1.10 and second IP is 192.168.1.20, SER Server has two ethernets, one is for Public IP 212.154.XXX.YYY for general use and the second ethernet is 192.168.1.5 connected to HUB where Gateways's LAN is. All ports of FXO's are registered with SER like 1111,2222,3333,4444 (Gateway A) and 5555,6666,7777,8888 (Gateway B). I want SER to send some calls to that numbers depending on Prefix : Before i worked with Ondo SIP Proxy a little and was successful with dial plan. Ex: a user dials 05353490056, proxys adds 1111 prefix, route to registered SIP client 1111 and send the call as 111105353490056(a)IP.of.Proxy to Gateway, prefix 1111 is stripped from SIP and the gateway connects to GSM like 05353490056(a)IP.of.Gateway Ex for Gateways: * for 1111,2222,3333 and 4444 : i want to use 053 prefix, so when a user dial a number like 05353490056, SER has to route it to 1111, if it's busy then route to 2222 so on till 4444. * for 5555 and 6666 : i want to use 050 and 055 prefix, again same as above. * for 7777 and 8888 : i want to use 054 prefix. * for prefixes apart from above and 833, i want to route all calls to 85.96.XXX.YYY , important thing is the clients are registered on SER but i want to route calls over from SER and by keeping their username/password combination since the clients are created on 85.96.XXX.YYY and that Gateway does auth for users, but SER also has to do accounting for this calls. Do i need so many things ? Thanks, Ozan Blotter *ps : I'm using SER 0.9.2 from CVS :) current SER.CFG as follows : [root@localhost ser]# cat ser.cfg debug=3 fork=yes log_stderror=no listen=192.168.1.9 # put your server IP address here port=5060 children=4 dns=no rev_dns=no 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/acc.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", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "log_fmt", "fimos") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 2) modparam("acc", "early_media", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "log_flag", 1) route { setflag(1); # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="INVITE" || method=="BYE" || method=="CANCEL") { setflag(1); route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:053[0-9]*@") { prefix("1111"); rewritehost ("192.168.1.10"); break; }; sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } [root@localhost ser]#

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Re: [Serusers] serctl moni question.
by Samuel Osorio Calvo 30 May '05

30 May '05

I think it indicates the number of transactions whic had returned failure messages. As happens with Waiting, these are internal counters that may have not been properly adapted through the SER versions and therefore its value could be not exact. Samuel. Unclassified. >>> Ricardo Martinez <rmartinez(a)redvoiss.net> 05/28/05 12:03AM >>> Hello. In the serctl moni appears a "failure" indicator. What represent this indicator? Could be something related to exec_dst() function? Thanks! Ricardo Martinez-. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] SER 0.9 | How to check URI against uri table ?
by lamasse＠free.fr 30 May '05

30 May '05

Hi Ser Users ! I'd like to use the uri table in SER database to check if the URI submitted on registration in 'TO' header is allowed for the user that tries to register (username for credentials and SIP URI are not the same). I tried to use the check_to() funtion of the uri_db module : "Check From username against URI table or digest credentials." But this check_to() function always reject the user because the URI is not the username, and it does not even try to check the uri table. How should I use this function (or another) to enable this verification ? Thanks for your help

1 0

[Serusers] QOS of VoIP
by Ritesh Jalan 30 May '05

30 May '05

Hi All >From where we can get the data for 1) ASR on various countries 2) Average Call drop on VoIP 3) Average Call Quality This we require to get an idea of what types of problem normally users use to face on voip and what is the average percentage of those problems. Pls. help me if anybody have the factsheet for various service provider on these paramaters Thanks & Regards Ritesh Jalan

1 0

[Serusers] Cannot compile ser 0.9 on FreeBSD
by Wilson 30 May '05

30 May '05

HI all, I downloaded the ser 0.9 source file from ftp.berlios.de My operation sytem is freebsd 4.11 When i try to compile the ser source code by using: gmake exclude_modules="postgres extcmd cpl auth_radius group_radius uri_radius avp_radius" prefix=/usr/home/gzbaby/ser proper This step runs well.And then i type a command as: gmake exclude_modules="postgres extcmd cpl auth_radius group_radius uri_radius avp_radius" prefix=/usr/home/gzbaby/ser all It returned me the following error: --------------------------------------------------------------------------------------------------------------------------- Makefile.rules:81: action.d: No such file or directory Makefile.rules:81: crc.d: No such file or directory Makefile.rules:81: daemonize.d: No such file or directory Makefile.rules:81: data_lump.d: No such file or directory Makefile.rules:81: data_lump_rpl.d: No such file or directory Makefile.rules:81: dprint.d: No such file or directory Makefile.rules:81: dset.d: No such file or directory Makefile.rules:81: error.d: No such file or directory Makefile.rules:81: fifo_server.d: No such file or directory Makefile.rules:81: flags.d: No such file or directory Makefile.rules:81: forward.d: No such file or directory Makefile.rules:81: hash_func.d: No such file or directory Makefile.rules:81: ip_addr.d: No such file or directory Makefile.rules:81: main.d: No such file or directory Makefile.rules:81: md5.d: No such file or directory Makefile.rules:81: md5utils.d: No such file or directory Makefile.rules:81: modparam.d: No such file or directory Makefile.rules:81: msg_translator.d: No such file or directory Makefile.rules:81: pass_fd.d: No such file or directory Makefile.rules:81: proxy.d: No such file or directory Makefile.rules:81: qvalue.d: No such file or directory Makefile.rules:81: re.d: No such file or directory Makefile.rules:81: receive.d: No such file or directory Makefile.rules:81: resolve.d: No such file or directory Makefile.rules:81: route.d: No such file or directory Makefile.rules:81: route_struct.d: No such file or directory Makefile.rules:81: script_cb.d: No such file or directory Makefile.rules:81: socket_info.d: No such file or directory Makefile.rules:81: sr_module.d: No such file or directory Makefile.rules:81: stats.d: No such file or directory Makefile.rules:81: tcp_main.d: No such file or directory Makefile.rules:81: tcp_read.d: No such file or directory Makefile.rules:81: timer.d: No such file or directory Makefile.rules:81: tsend.d: No such file or directory Makefile.rules:81: udp_server.d: No such file or directory Makefile.rules:81: unixsock_server.d: No such file or directory Makefile.rules:81: usr_avp.d: No such file or directory Makefile.rules:81: mem/f_malloc.d: No such file or directory Makefile.rules:81: mem/mem.d: No such file or directory Makefile.rules:81: mem/memtest.d: No such file or directory Makefile.rules:81: mem/q_malloc.d: No such file or directory Makefile.rules:81: mem/shm_mem.d: No such file or directory Makefile.rules:81: mem/vq_malloc.d: No such file or directory Makefile.rules:81: parser/hf.d: No such file or directory Makefile.rules:81: parser/msg_parser.d: No such file or directory Makefile.rules:81: parser/parse_allow.d: No such file or directory Makefile.rules:81: parser/parse_content.d: No such file or directory Makefile.rules:81: parser/parse_cseq.d: No such file or directory Makefile.rules:81: parser/parse_disposition.d: No such file or directory Makefile.rules:81: parser/parse_diversion.d: No such file or directory Makefile.rules:81: parser/parse_event.d: No such file or directory Makefile.rules:81: parser/parse_expires.d: No such file or directory Makefile.rules:81: parser/parse_fline.d: No such file or directory Makefile.rules:81: parser/parse_from.d: No such file or directory Makefile.rules:81: parser/parse_hname2.d: No such file or directory Makefile.rules:81: parser/parse_hostport.d: No such file or directory Makefile.rules:81: parser/parse_methods.d: No such file or directory Makefile.rules:81: parser/parse_nameaddr.d: No such file or directory Makefile.rules:81: parser/parse_param.d: No such file or directory Makefile.rules:81: parser/parse_rpid.d: No such file or directory Makefile.rules:81: parser/parse_rr.d: No such file or directory Makefile.rules:81: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:81: parser/parse_to.d: No such file or directory Makefile.rules:81: parser/parse_uri.d: No such file or directory Makefile.rules:81: parser/parse_via.d: No such file or directory Makefile.rules:81: parser/parser_f.d: No such file or directory Makefile.rules:81: parser/digest/digest.d: No such file or directory Makefile.rules:81: parser/digest/digest_parser.d: No such file or directory Makefile.rules:81: parser/digest/param_parser.d: No such file or directory Makefile.rules:81: parser/contact/contact.d: No such file or directory Makefile.rules:81: parser/contact/parse_contact.d: No such file or directory Makefile.rules:81: db/db.d: No such file or directory Makefile.rules:81: db/db_fifo.d: No such file or directory Makefile.rules:81: lex.yy.d: No such file or directory Makefile.rules:81: cfg.tab.d: No such file or directory yacc -d -b cfg cfg.y yacc: 1 shift/reduce conflict flex cfg.lex gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c action.c -o action.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c crc.c -o crc.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c daemonize.c -o daemonize.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c data_lump.c -o data_lump.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c data_lump_rpl.c -o data_lump_rpl.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c dprint.c -o dprint.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c dset.c -o dset.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c error.c -o error.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c fifo_server.c -o fifo_server.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c flags.c -o flags.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c forward.c -o forward.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c hash_func.c -o hash_func.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c ip_addr.c -o ip_addr.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c main.c -o main.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c md5.c -o md5.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c md5utils.c -o md5utils.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c modparam.c -o modparam.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c msg_translator.c -o msg_translator.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c pass_fd.c -o pass_fd.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c proxy.c -o proxy.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c qvalue.c -o qvalue.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c re.c -o re.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c receive.c -o receive.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c resolve.c -o resolve.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c route.c -o route.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c route_struct.c -o route_struct.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c script_cb.c -o script_cb.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c socket_info.c -o socket_info.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c sr_module.c -o sr_module.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c stats.c -o stats.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c tcp_main.c -o tcp_main.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c tcp_read.c -o tcp_read.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c timer.c -o timer.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c tsend.c -o tsend.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c udp_server.c -o udp_server.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c unixsock_server.c -o unixsock_server.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/home/wilson/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -c usr_avp.c -o usr_avp.o In file included from usr_avp.c:45: mem/shm_mem.h: In function `_shm_malloc': mem/shm_mem.h:135: too many arguments to function `fm_malloc' mem/shm_mem.h: In function `_shm_realloc': mem/shm_mem.h:146: too many arguments to function `fm_realloc' usr_avp.c: In function `destroy_avp': usr_avp.c:313: too many arguments to function `fm_free' usr_avp.c: In function `destroy_avp_list_unsafe': usr_avp.c:328: too many arguments to function `fm_free' usr_avp.c: In function `destroy_avp_list': usr_avp.c:343: too many arguments to function `fm_free' usr_avp.c: In function `add_avp_galias': usr_avp.c:420: too many arguments to function `fm_malloc' usr_avp.c:426: too many arguments to function `fm_malloc' usr_avp.c:437: too many arguments to function `fm_malloc' usr_avp.c:459: too many arguments to function `fm_free' usr_avp.c:461: too many arguments to function `fm_free' gmake: *** [usr_avp.o] Error 1 ---------------------------------------------------------------------------------------------------------------------------- Can anyone do me a favour? your help will be highly appreciated. Thank you so much! Kind Regards, Wilson augzbaby(a)yahoo.com.au 2005-05-30

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[Serusers] No sound with ser 0.9.0 and mediaproxy 1.2.1
by Roger Lewau 29 May '05

29 May '05

Hello list... I've installed SER 0.9.0 and mediaproxy 1.2.1 and dowloaded the getting started scripts (issue 04a) from onsip.org. Setup everything as instructed, added a couple of users to the database and tried to setup a call between one public and one NATed UA. Both phones register properly and signalling seems to be working fine, I ring each device from the other, problem is there is no sound. So I checked the mediaproxy sessions... and it looks like this Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------- ---------------------------------- 82.182.194.x:5004 - 212.247.91.x:35014 - 82.182.194.x:49160 inactive 0'07" Unknown Audio 0/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 I dont really understand the output but I think "inactive" as well as "Unknown Audio" is not a good sign. How ever, I then used Ngrep to see if there are some rtp traffic going on and sure enough, there is traffic comming in from both UAs but no traffic is leaving the proxy. Strange! Anyone with some bright idea on why mediaproxy is not relaying the sound? My ser.cfg =================================== debug=3 fork=yes log_stderror=no listen=212.247.91.<hidden> # INSERT YOUR IP ADDRESS HERE port=5060 children=4 alias=sip.<somedomain>.com dns=no rev_dns=no 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.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") #modparam("nathelper", "rtpproxy_disable", 1) #modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("212.247.91.<hidden>:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # 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 (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; log("NO MATCH"); route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:[+|00][0-9]*@") { # International PSTN inleds antingen med + eller 00 route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:0[1-9]*@") { # Domestic PSTN startar med 0 route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.0.2.245"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(1); break; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ================================== My mediaproxy.ini ================================== ; ; Configuration file for MediaProxy ; [Dispatcher] [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = ser ;listen = None allow = Any proxyIP = 212.247.91.<hidden> portRange = 35000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 ;forceClose = 0 accounting = off [Accounting] user = dbuser password = dbpass host = dbhost database = radius table = radacct ==================================

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Re: [Serusers] Error loading mysql module - undefined symbol fm_malloc
by a c 29 May '05

29 May '05

and # make prefix=/usr/local install Makefile.defs:460: Old gcc detected (2.9x), use gcc >= 3.1 for better results --- a c <lra101(a)yahoo.com> wrote: > I don't see any errors but could the gcc version > make > a difference? > > %gcc -v > Reading specs from > /usr/lib/gcc-lib/i386-redhat-linux/2.96/specs > gcc version 2.96 20000731 (Red Hat Linux 7.1 > 2.96-81) > > > --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > > > I would check one more time and make sure that > > compilation and linking give > > no errors/warnings... > > g-) > > > > a c wrote: > > > hello Greger, > > > > > > seems like it is already commented out..hmm > > > > > > > > > DEFS+= $(extra_defs) \ > > > -DNAME='"$(MAIN_NAME)"' > > > -DVERSION='"$(RELEASE)"' -DARCH='"$(ARCH)"' \ > > > -DOS='"$(OS)"' -DCOMPILER='"$(CC_VER)"' > > > -D__CPU_$(ARCH) -D__OS_$(OS) \ > > > -DCFG_DIR='"$(cfg-target)"'\ > > > -DPKG_MALLOC \ > > > -DSHM_MEM -DSHM_MMAP \ > > > -DDNS_IP_HACK \ > > > -DUSE_IPV6 \ > > > -DUSE_MCAST \ > > > -DUSE_TCP \ > > > -DDISABLE_NAGLE \ > > > #-DDBG_QM_MALLOC \ > > > #-DF_MALLOC \ > > > #-DDBG_F_MALLOC \ > > > #-DNO_DEBUG \ > > > #-DNO_LOG \ > > > #-DVQ_MALLOC \ > > > #-DCONTACT_BUG \ > > > #-DDBG_LOCK \ > > > #-DNOSMP \ > > > #-DEXTRA_DEBUG \ > > > #-DUSE_SHM_MEM \ > > > #-DSTATS \ > > > #-DNO_LOG > > > > > > > > > --- "Greger V. Teigre" <greger(a)teigre.com> > wrote: > > > > > >> Looks like you have compiled with a memory > alloc > > >> function not available. > > >> Check your Makefile.defs file. There are many > > >> different options. Look at > > >> this section: > > >> DEFS+= $(extra_defs) \ > > >> -DNAME='"$(MAIN_NAME)"' > > >> -DVERSION='"$(RELEASE)"' -DARCH='"$(ARCH)"' > > >> \ > > >> -DOS='"$(OS)"' > -DCOMPILER='"$(CC_VER)"' > > >> -D__CPU_$(ARCH) -D__OS_$(OS) > > >> \ > > >> -DCFG_DIR='"$(cfg-target)"'\ > > >> -DPKG_MALLOC \ > > >> -DSHM_MEM -DSHM_MMAP \ > > >> -DDNS_IP_HACK \ > > >> -DUSE_IPV6 \ > > >> -DUSE_MCAST \ > > >> -DUSE_TCP \ > > >> -DDISABLE_NAGLE \ > > >> -DDBG_QM_MALLOC \ > > >> -DF_MALLOC \ > > >> #-DDBG_F_MALLOC \ > > >> #-DNO_DEBUG \ > > >> #-DNO_LOG \ > > >> #-DVQ_MALLOC \ > > >> #-DCONTACT_BUG \ > > >> #-DDBG_LOCK \ > > >> #-DNOSMP \ > > >> #-DEXTRA_DEBUG \ > > >> #-DUSE_SHM_MEM \ > > >> #-DSTATS \ > > >> #-DNO_LOG > > >> > > >> You can comment out F_MALLOC > > >> g-) > > >> > > >> a c wrote: > > >>> hello, > > >>> > > >>> having some problems loading the mysql module. > I > > am > > >>> using the latest dev release. couldn't find > much > > help > > >>> in the archieves.. > > >>> > > >>> ser -cf ser.mysql.cfg > > >>> 0(1439) ERROR: load_module: could not open > > module > > >>> </usr/local/lib/ser/modules/mysql.so>: > > >>> /usr/local/lib/ser/modules/mysql.so: undefined > > symbol: > > >>> fm_malloc > > >>> 0(1439) parse error (15,13-14): failed to load > > module > > >>> ERROR: bad config file (1 errors) > > >>> > > >>> > > >>> ----------------config------------------ > > >>> debug=3 > > >>> fork=yes > > >>> log_stderror=no > > >>> > > >>> listen=192.168.0.5 > > >>> port=5060 > > >>> children=4 > > >>> > > >>> dns=no > > >>> rev_dns=no > > >>> > > >>> fifo="/tmp/ser_fifo" > > >>> fifo_db_url="mysql://ser:heslo@localhost/ser" > > >>> > > >>> loadmodule > "/usr/local/lib/ser/modules/mysql.so" > > >>> > > >>> > > >>> > > __________________________________________________ > > >>> Do You Yahoo!? > > >>> Tired of spam? Yahoo! Mail has the best spam > > protection around > > >>> http://mail.yahoo.com > > >>> > > >>> > _______________________________________________ > > >>> Serusers mailing list > > >>> serusers(a)lists.iptel.org > > >>> > http://lists.iptel.org/mailman/listinfo/serusers > > >> > > >> > > > > > > > > > > > > > > > __________________________________ > > > Do you Yahoo!? > > > Yahoo! Small Business - Try our new Resources > site > > > http://smallbusiness.yahoo.com/resources/ > > > > > > > > > __________________________________ > Do you Yahoo!? > Yahoo! Small Business - Try our new Resources site > http://smallbusiness.yahoo.com/resources/ > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Serusers] Re: SER CANCEL Processing
by a c 29 May '05

29 May '05

thanks guys..t_relay was the problem. I did not have this in the config. issue is resolved. ac --- a c <lra101(a)yahoo.com> wrote: > I believe the 487 does not get matched because the > UA > which does the CANCEL never gets a 200 OK back since > the CANCEL was not send out to the terminating UA by > SER. For CANCEL, are we supposed to have special > config? I don't have any specifically handles CANCEL > {other than 487}. > > ac > > > --- Steve Blair <blairs(a)isc.upenn.edu> wrote: > > > > > I am having a similar problem. I am not yet > > convinced that it is SER > > as much as my configuration but if SER does have > > difficulty processing > > CANCEL messages it would be helpful to understand > > under what > > circumstances. > > > > By the way I have been using t_check_status in > a > > similar way since > > v 0.8 code. When I upgraded to 0.9.0, .0 and now > .2 > > the t_check_status > > fails to be matched. > > > > -Steve > > > > Thanks,Steve > > > > a c wrote: > > > > >any ideas anyone? seems like SER does not respond > > to > > >CANCEL requests and eventually takes the failure > > >route. > > > > > >I have the following in my failure route, but it > > never > > >gets hit, since SER does not send the 200 OK back > > or > > >pass the CANCEL message forward. log file > attached. > > > > > >failure_route[1] > > >{ > > > if(t_check_status("487")) { > > > # don't continue on cancellation > > > xlog("L_WARN", "OH OH OH\n"); > > > break; > > > } > > > > > >ac > > > > > > > > >--- a c <lra101(a)yahoo.com> wrote: > > > > > > > > > > > >>Seems like SER does not respond back to CANCEL > > >>Requests. Is this a issue / some config problem > on > > >>my > > >>side? ngrep traces attached > > >> > > >>interface: eth0 (192.168.0.0/255.255.255.0) > > >>filter: ip and ( port 5060 ) > > >># > > >>U 192.168.0.2:5062 -> 192.168.0.5:5060 > > >> INVITE sip:9999999999@voip.com SIP/2.0..Via: > > >>SIP/2.0/UDP 192.168.0.2:5062;r > > >> > > >> > > >> > > >> > > > >port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > > > > > > > > >>102 <sip:102@voi > > >> p.com:5062>;tag=3282942385..To: > > >><sip:9999999999@voip.com>..Contact: <sip:10 > > >> 2@192.168.0.2:5062>..Call-ID: > > >>29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. > > >> 0.2..CSeq: 940 INVITE..Max-Forwards: > > >>70..Content-Type: application/sdp..Use > > >> r-Agent: X-Lite release 1103m..Content-Length: > > >>258....v=0..o=102 7659613 76 > > >> 59633 IN IP4 192.168.0.2..s=X-Lite..c=IN IP4 > > >>192.168.0.2..t=0 0..m=audio 80 > > >> 00 RTP/AVP 0 8 3 98 96..a=rtpmap:0 > > >>pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpma > > >> p:3 gsm/8000..a=rtpmap:98 > iLBC/8000..a=rtpmap:96 > > >>telephone-event/8000..a=fm > > >> tp:96 0-15.. > > > > > >> > > >> > > >># > > >>U 192.168.0.5:5060 -> 192.168.0.2:5062 > > >> SIP/2.0 100 trying -- your call is important > to > > >>us..Via: SIP/2.0/UDP 192.16 > > >> > > >> > > >> > > >> > > > >8.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > > > > > > > > >> 102 > <sip:102@voip.com:5062>;tag=3282942385..To: > > >><sip:9999999999@voip.com>. > > >> .Call-ID: > > >> > > >> > > >> > > > >29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: > > > > > > > > >>940 INVIT > > >> E..Server: Sip EXpress router (0.9.2 > > >>(i386/linux))..Content-Length: 0..Warn > > >> ing: 392 192.168.0.5:5060 "Noisy feedback > tells: > > > > >>pid=2943 req_src_ip=192.1 > > >> 68.0.2 req_src_port=5062 > > >>in_uri=sip:9999999999@voip.com > out_uri=sip:9999999 > > >> 999(a)voip.com via_cnt==1".... > > > > > >> > > >> > > >># > > >>U 192.168.0.5:5060 -> 192.168.0.3:5060 > > >> INVITE sip:9999999999@voip.com > > >>SIP/2.0..Record-Route: <sip:192.168.0.5;ftag > > >> =3282942385;lr=on>..Via: SIP/2.0/UDP > > >>192.168.0.5;branch=z9hG4bKe034.1018c5d > > >> 1.0..Via: SIP/2.0/UDP > > >> > > >> > > >> > > > >192.168.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8C > > > > > > > > >> F4A479B6D9DDB17C5CFC1..From: 102 > > >><sip:102@voip.com:5062>;tag=3282942385..To > > >> : <sip:9999999999@voip.com>..Contact: > > >><sip:102@192.168.0.2:5062>..Call-ID: > > >> > > >> > > >> > > >> > > > >29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: > > > > > > > > >>940 INVITE..Max-For > > >> wards: 16..Content-Type: > > >>application/sdp..User-Agent: X-Lite release > 1103m. > > >> .Content-Length: 258....v=0..o=102 7659613 > > 7659633 > > >>IN IP4 192.168.0.2..s=X- > > >> Lite..c=IN IP4 192.168.0.2..t=0 0..m=audio > 8000 > > >>RTP/AVP 0 8 3 98 96..a=rtpm > > >> ap:0 pcmu/8000..a=rtpmap:8 > pcma/8000..a=rtpmap:3 > > >>gsm/8000..a=rtpmap:98 iLBC > > >> /8000..a=rtpmap:96 > > telephone-event/8000..a=fmtp:96 > > >>0-15.. > > >># > > >>U 192.168.0.3:5060 -> 192.168.0.5:5060 > > >> SIP/2.0 100 Trying..From: 102 > > >><sip:102@voip.com:5062>;tag=3282942385..To: < > > >> sip:9999999999@voip.com>..Call-Id: > > >>29469D19-CDF1-49ED-8245-A67E1B1DBB88@192 > > >> .168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP > > >>192.168.0.5;branch=z9hG4bKe034 > > >> .1018c5d1.0..Via: SIP/2.0/UDP > > >>192.168.0.2:5062;rport=5062;branch=z9hG4bKE29 > > >> 503D9D8CF4A479B6D9DDB17C5CFC1..Content-Length: > > >>0.... > > >> > > >># > > >>U 192.168.0.3:5060 -> 192.168.0.5:5060 > > >> SIP/2.0 180 Ringing..From: 102 > === message truncated === __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/

3 11

Fw: [Serusers] Ser+Nat +Polycom phones
by Greger V. Teigre 29 May '05

29 May '05

Sorry, but for security reasons I don't read other people's dumps. You just need to check whether the m= line refers to your rtpproxy's public IP and not the private. And of course that rtpproxy and nathelper communicate with eachother... g-) > harry gaillac wrote: >> I've built rtpproxy with -i option : >> >> root 14362 1 0 16:39 ? 00:00:00 >> /usr/local/bin/rtpproxy -l 192.168.0.1 -i >> 80.119.11.112 >> >> However I can't send or receive invite from/to user >> agent . >> look at invitation file (ethereal)! >> Should I configure phone per port ? >> What's wrong ? >> >> Harry >> >> >> --- "Greger V. Teigre" <greger(a)teigre.com> a écrit : >> >>> The patch was current for rtpproxy from repository >>> at that point. I sent the >>> patch to Maxim and asked if it could be included in >>> CVS, but I never heard >>> from him. >>> If you add the patch by hand, you should be able >>> to make rtpproxy >>> announce the public IP in the INVITEs and OKs >>> (instead of rtpproxy's >>> private/interface address). >>> g-) >>> >>> harry gaillac wrote: >>>> I tried to patch main.c without success according >>> to >>>> main.c.rej but i try to add -i by hand. >>>> >>>> I failed to patch main.c . >>>> >>>> If I add -i options to rtpproxy will it solve >>> problem >>>> of signaling behind nat (on same box) ? >>>> >>>> >>>> Harry >>>> >>>> --- "Greger V. Teigre" <greger(a)teigre.com> a écrit >>>> >>>> >>>>> >>>> >>> >> http://lists.iptel.org/pipermail/serusers/2005-January/014688.html >>>>> harry gaillac wrote: >>>>>> Where can I get your patch ? >>>>>> >>>>>> Harry >>>>>> --- "Greger V. Teigre" <greger(a)teigre.com> a >>> écrit >>>>>> >>>>>> >>>>>>> Running rtpproxy on a NATed box is a problem >>> (you >>>>>>> need a patch I posted a >>>>>>> while back to announce the correct public IP). >>>>>>> Running all daemons on the >>>>>>> same server with a public IP is NOT a problem. >>>>>>> g-) >>>>>>> harry gaillac wrote: >>>>>>>> Hello, >>>>>>>> >>>>>>>> SER+nathelper+rtpproxy run on the same box that nat. >>>>>>>> Is it a problem ? >>>>>>>> >>>>>>>> Harry >>>>>>>> >>>>>>>> >>>>>>>> --- "Greger V. Teigre" <greger(a)teigre.com> a >>>>> écrit >>>>>>>> >>>>>>>> >>>>>>>>> That's strange. We are using some Polycom 300 >>>>> and >>>>>>>>> they work ok. No STUN >>>>>>>>> support and sometimes a very erratic >>>>> registration >>>>>>>>> pattern, but except from >>>>>>>>> that, they seem to perform ok. They >>>>> interoperate >>>>>>>>> with sipuras, >>>>>>>>> grandstreams, sjphone, cisco gw, xlite, etc. >>>>>>>>> g-) >>>>>>>>> >>>>>>>>> Juha Heinanen wrote: >>>>>>>>>> harry, >>>>>>>>>> >>>>>>>>>> we tried polycom video phones a month or so >>> ago and their sip >>>>>>>>>> implementation was worst i had seen so far. >>> the phones didn't >>>>>>>>>> support digest authentication and we were also hit the >>>>>>>>>> problem you mentioned with domain in request-uri. i suggest >>>>>>>>>> to junk those phones before their sip implementation is >>>>>>>>>> closer to >>> rfc3261. >>>>>>>>>> >>>>>>>>>> -- juha >>>>>>>>>> >>>>>>>>>> >>>>> _______________________________________________ >>>>>>>>>> Serusers mailing list >>>>>>>>>> serusers(a)lists.iptel.org >>>>>>>>>> >>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>> >>>> >>> >> _____________________________________________________________________________ >>>>>>>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage >>>>>>>> pour vos mails, photos et vidéos ! Créez votre Yahoo! >>> Mail sur >>>>>>>> http://fr.mail.yahoo.com >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>> >>> >> _____________________________________________________________________________ >>>>>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace >>>>> de stockage pour vos >>>>>> mails, photos et vidéos ! Créez votre Yahoo! Mail sur >>>>>> http://fr.mail.yahoo.com >>>>> >>>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> >> _____________________________________________________________________________ >>>> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace >>> de stockage pour vos >>>> mails, photos et vidéos ! Créez votre Yahoo! Mail >>> sur >>>> http://fr.mail.yahoo.com >>> >>> >> >> >> >> >> >> >> _____________________________________________________________________________ >> Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos >> mails, photos et vidéos ! Créez votre Yahoo! Mail sur >> http://fr.mail.yahoo.com

1 0

[Serusers] problems with NAT, ATA not responding to INVITEs
by Brian McCrary 29 May '05

29 May '05

Hello, I've been trying to add support for ATAs behind NAT firewalls for several days, and have had nothing but frustrations with it, so hopefully someone can clue me in. I tried modifying my existing ser config which has been working great, but failing that, I finally got the stock config file from ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-mediaproxy.3.05.cfg, modified my IP address for NAT and used that (after adding the domain module). My first problem is that when an ATA registers behinf the NAT firewall, the NAT flag is never set, so if doesn't get the pings every 30 seconds. I can manually set that flag by just having setflag(6) in the config file and that seems to temporarily fix that problem, but I can't tell that it does anything else. Outbound calls from the ATA work just fine, which they always have even where I had no NAT stuff in SER at all. Inbound calls to the ATA get ignored my it though, this is a Cisco ATA-186. I hooked up an old hub so I could get the messages using ngrep on the side with the NAT router and the main server, and here's what I see. 7771111001 is the ATA, 7778881000 is a local phone. 11.11.11.100 is the SER router, 11.11.11.44 is the PSTN gateway initiating the call, 11.11.11.18 is the NAT router's public IP, and 10.0.2.3 is the ATA. SER router's ngrep: U 11.11.11.100:5060 -> 11.11.11.18:1387 27067@0:1480 ...k....INVITE sip:7771111001@11.11.11.18:1387;user=phone;transport=udp SI P/2.0..Record-Route: <sip:11.11.11.100;ftag=6B826304-1A6A;lr=on>..Via: SIP/2 .0/UDP 11.11.11.100;branch=z9hG4bKeda1.2545.3..Via: SIP/2.0/UDP 11.11.11.44 :5060;x-route-tag="tgrp:lnx";branch=z9hG4bK8C9D..From: <sip:7778881000@11.11 11.44>;tag=6B826304-1A6A..To: <sip:7771111001@11.11.11.100>..Date: Thu, 26 May 2005 14:10:42 GMT..Call-ID: C7AED903-CD2611D9-8130A2A3-EEBB2457(a)11.11. 11.44..Supported: 100rel,timer..Min-SE: 1800..Cisco-Guid: 3350084827-34418 24217-2183397385-1128595304..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 16..Remote-Party-ID : <sip:7778881000@11.11.11.44>;party=calling;screen=no;privacy=off..Timesta mp: 1117116642..Contact: <sip:7778881000@11.11.11.44:5060>..Expires: 180..A llow-Events: telephone-event..Content-Type: application/sdp..Content-Length : 506....v=0..o=CiscoSystemsSIP-GW-UserAgent 2462 1180 IN IP4 11.11.11.44.. s=SIP Call..c=IN IP4 11.11.11.44..t=0 0..m=audio 16536 RTP/AVP 18 2 0 100 1 01..c=IN IP4 11.11.11.44..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=yes..a=rt pmap:2 G726-32/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:100 X-NSE/8000..a=fmtp: 100 192-194,200-202..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a= X-sqn:0..a=X-cap: 1 audio RTP/AVP 100..a=X-cpar: a=rtpmap:100 X-NSE/8000..a =X-cpar: a=fmtp:100 192-194,200-202..a=X-cap: 2 image u NAT (ATA location) ngrep: U 11.11.11.100:5060 -> 10.0.2.3:5060 27070@0:1462 .......&INVITE sip:7771111001@11.11.11.18:1405;user=phone;transport=udp S IP/2.0..Record-Route: <sip:11.11.11.100;ftag=6B826304-1A6A;lr=on>..Via: SIP /2.0/UDP 11.11.11.100;branch=z9hG4bKeda1.2545.6..Via: SIP/2.0/UDP 11.11.11.44:5060;x-route-tag="tgrp:lnx";branch=z9hG4bK8C9D..From: <sip:7778881000@ 11.11.11.44>;tag=6B826304-1A6A..To: <sip:7771111001@11.11.11.100>..Date: Th u, 26 May 2005 14:10:42 GMT..Call-ID: C7AED903-CD2611D9-8130A2A3-EEBB2457@ 11.11.11.44..Supported: 100rel,timer..Min-SE: 1800..Cisco-Guid: 335008482 7-3441824217-2183397385-1128595304..User-Agent: Cisco-SIPGateway/IOS-12.x. .Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 16..Remot e-Party-ID: <sip:7778881000@11.11.11.44>;party=calling;screen=no;privacy=o ff..Timestamp: 1117116642..Contact: <sip:7778881000@11.11.11.44:5060>..Exp ires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..C ontent-Length: 478....v=0..o=CiscoSystemsSIP-GW-UserAgent 2462 1180 IN IP4 11.11.11.44..s=SIP Call..c=IN IP4 11.11.11.44..t=0 0..m=audio 16536 RTP/A VP 18 2 0 100 101..c=IN IP4 11.11.11.44..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=yes..a=rtpmap:2 G726-32/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:100 X- NSE/8000..a=fmtp:100 192-194,200-202..a=rtpmap:101 telephone-event/8000..a =fmtp:101 0-16..a=X-sqn:0..a=X-cap: 1 audio RTP/AVP 100..a=X-cpar: a=rtpma p:100 X-NSE/8000..a=X-cpar: a=fmtp:100 192-194,200-202.. It looks like the ATA is getting the message, but just flat out ignoring it. Anyone else have this problem? Brian

2 1

RE: [Serusers] avp_radius and avpops
by Ricardo Martinez 29 May '05

29 May '05

Yes. I noticed that the variable used by the avp_load_radius("caller") to store the answer from the RADIUS server is in fact the one i defined first (var1), but is has a prefix called "caller_" , so the variable is finally : "caller_var1". Jackpot!. So i used this in my ser.cfg: if( !avp_load_radius("caller")) { xlog("L_INFO", "time [%Tf] FAIL AVP_LOAD_RADIUS!\n"); }; if( !avp_pushto("$ruri", "s:caller_var1/g")) { xlog("L_INFO", "time [%Tf] FAIL avp_pushto()!\n"); }; Thanks Greg for your help. Just one more thing. This could be indicated in the module's README, i think is an important thing that a user should know. Regards!!. Ricardo Martinez.- > -----Mensaje original----- > De: Greger V. Teigre [mailto:greger@teigre.com] > Enviado el: Viernes, 27 de Mayo de 2005 3:57 > Para: Ricardo Martinez; serusers(a)lists.iptel.org > Asunto: Re: [Serusers] avp_radius and avpops > > > Do the debugging dance, so that you can verify what the calls > to avp_* do... > g-) > > Ricardo Martinez wrote: > > Hello Greg > > Thanks for your answer. > > I made the change but it seems not to do anything.... > > Can someone help me here? > > > > Thanks in advance > > > > Regards > > Ricardo.- > > > >> -----Mensaje original----- > >> De: Greger V. Teigre [mailto:greger@teigre.com] > >> Enviado el: Miércoles, 25 de Mayo de 2005 16:24 > >> Para: Ricardo Martinez; serusers(a)lists.iptel.org > >> Asunto: Re: [Serusers] avp_radius and avpops > >> > >> > >>> if( !avp_pushto("$RURI", "s:var1/g")) { > >> replace with: > >> "s:$var1/g" > >> I don't know about the case sensivtivity of destination, I > >> always use lower > >> case... > >> g-) > >> > >> Ricardo Martinez wrote: > >>> Hello List. > >>> I have a question regarding the use of avp_radius and avpops. I'm > >>> using avp_radius to obtain an AVP value from my database > via radius. > >>> What i what to do is replace this value for the RURI. > >>> Here i have a couple of questions. > >>> 1.- The value returned by the avp_radius (the SIP-AVP) where is > >>> stored ? It suppose that the SIP-AVP returned by radius > >> has the form > >>> of "name:value". That "name" refers to the name of what?. > >>> For example i'm returning : > >> "var1:sip:1234567@mydomain.com". What i > >>> see in the debug is : > >>> > >>> avp_load_user: AVP 'var1'='sip:1234567@mydomain.com' has > been added > >>> > >>> This is what i got in my ser.cfg (a snippet). > >>> > >>> if (method=="INVITE" || method=="CANCEL") { > >>> if( !avp_load_radius("caller")) { > >>> log (1, "AVP_RADIUS: Fail on > avp_radius\n"); > >>> }; > >> > >>> > >>> if( !avp_pushto("$RURI", "s:var1/g")) { > >>> log (1, "AVPOPS: Fail on AVPOPS\n"); > >>> }; > >>> }; > >>> > >>> Again the debug = 9 . > >>> > >>> 6(23815) avp_load_user: AVP > >> 'var1'='sip:1234567@mydomain.com' has been > >>> added > >>> 6(23815) qm_free(0x8123400, 0x8166ccc), called from avp_radius.c: > >>> load_avp_user(344) > >>> 6(23815) qm_free: freeing frag. 0x8166cb4 alloc'ed from > >>> avp_radius.c: load_avp_user(330) > >>> 6(23815) DEBUG:avpops:pushto_avp: no avp found > >>> 6(23815) AVPOPS: Fail on AVPOPS > >>> > >>> What i'm doing wrong? > >>> Thanks! > >>> > >>> Regards > >>> Ricardo Martinez > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers >

3 2

[Serusers] mediaproxy register problem
by Sönmez Güneş 29 May '05

29 May '05

Hi everybody, I setup ser 0.8.14 with mysql support on fedora core 2. And two UA could connect over SER successfully. I give a static IP to my SIP PROXY and followed ONSIP instructions about mediaproxy. But I could not register UAC's behind NAT. begining of my route block: if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("PROXY_IP:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # 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; }; my register block: # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_contact(); force_rport(); }; if (!www_authorize("PROXY_IP","subscriber")) { www_challenge("PROXY_IP","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; invite block: # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("PROXY_IP","subscriber")) { proxy_challenge("PROXY_IP","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); Any help would be appreciated...

3 4

[Serusers] rtproxy based on URI
by Adrian A 29 May '05

29 May '05

Is it possible to configure SER to use RTP proxy only for specific clients, ie ones which have a particular username pattern? I would like to proxy the media only for those phones I have which do not support ICE but support STUN.

2 1

[Serusers] dead SIP service providers
by Michael Kendall 29 May '05

29 May '05

Hello All, The following two sites appear to be SIP service providers: www.freeworlddialup.com/ voip.brujula.net/english/messenger-howto.html But when I follow their instruction for setting up Windiows Messenger 4.6, my login attempts tell me, "The service provider is temporarly unavailable, try again later" This makes me suspect that they have shut down their voip servers, and simply failed to delete their web pages. Does anybody know the status of these two companies? They have not responded to my email queries. Freeworlddialup provides a simple tool to evaluate your internet connection before trying their service. The tool consistently claims that my port 5060 is blocked, even though I am directly connected to my cable modem and have all firewalls turned off. It seems like this false result could be caused by an attempt establish a 5060 connection to a non-existant voip server, again supporting my theory that their server has been shut down. Does this theory make sense? Also, both providers say that they work with Windows Messenger 4.6 and 4.7 but not with 5.1. Did Microsoft commit some crime in version 5.1 to make it incompatible with third party SIP service providers? thanks for your help, Michael

2 1

[Serusers] missing Digest-Algorithm = "MD5"
by davor jovanovic 29 May '05

29 May '05

Hi all, I'm using SER 0.8.14, FreeRadius 1.0.2. and XLite 1103. My question is about MD5, why I can't see Digest-Algorithm = "MD5" in FreeRadius and XLite client outputs. Is this a problem in XLite or SER? Everything is configured as in HOW TO and I get authenticated with FreeRadius so I suppose that configuration is OK. Dictionaries are the same in radiusclient on SER machine and in FreeRadius. I can provide more outputs if someone is interested to help. Any help would be appreciated. *****This is output from FreeRadius Server****** (no Digest-Algorithm = "MD5" received) ... rad_recv: Access-Request packet from host 161.53.0.131:33854, id=131, length=207 User-Name = "djovanovic.srce" Digest-Attributes = 0x0a11646a6f76616e6f7669632e73726365 Digest-Attributes = 0x0109737263652e6872 Digest-Attributes = 0x022a3432393439336464646235643930376433653433323765333063366566646566336436 6137643933 Digest-Attributes = 0x040d7369703a737263652e6872 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "263ec6802b382ddf8d58b363fd0629ad" Service-Type = Sip-Session Sip-Uri-User = "djovanovic.srce" NAS-IP-Address = 161.53.0.131 NAS-Port = 5060 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 0 modcall[authorize]: module "preprocess" returns ok for request 0 modcall[authorize]: module "chap" returns noop for request 0 modcall[authorize]: module "mschap" returns noop for request 0 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "djovanovic.srce" Digest-Realm = "srce.hr" Digest-Nonce = "429493dddb5d907d3e4327e30c6efdef3d6a7d93" Digest-URI = "sip:srce.hr" Digest-Method = "REGISTER" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 0 ... *****This is output from XLite****** (no Digest-Algorithm = "MD5" sent) SEND TIME: 203074718 SEND >> 191.153.206.58:5060 INVITE sip:537@srce.hr SIP/2.0 Via: SIP/2.0/UDP 191.153.206.57:5060;rport;branch=z9hG4bK8BB5DE10CC404EEBABE6761213B4AA45 From: davor <sip:djovanovic.srce@srce.hr>;tag=2361756632 To: <sip:537@srce.hr> Contact: <sip:djovanovic.srce@191.153.206.57:5060> Call-ID: 146A8327-BCC6-42E6-A027-83EA1832631B(a)191.153.206.57 CSeq: 40898 INVITE Authorization: Digest username="djovanovic.srce",realm="srce.hr",nonce="429492e156f0af6f7ae5e821d7 3df9be3e04f360",response="17a54d30cc2d5ecf1cc8e714d40210e0",uri="sip:537@src e.hr" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 308

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[Serusers] Call setup delay
by James Peters 29 May '05

29 May '05

I am using SER for VOIP and experiencing issues with call setup when making VOIP to VOIP calls. With 2 end units connected to the same server, there is a 5-7 second delay in the RTP stream when I place a call from one IP phone to another. I also experience a delay in ring back when calling inbound to an IP phone from a standard PSTN line. I am new to configuring SER. I can provide more info if needed. Thanks. James D. Peters Network Engineer NetCarrier, Inc. www.netcarrier.com <http://www.netcarrier.com>

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Re: [Serusers] G723.1 vs G729
by Mohamed Omar 28 May '05

28 May '05

I use dialup connection with my IPPhones which has a buildin modem without any issue on both codec G723 and G729. I sometime get better quality on the some phone if I use dialup then when I was braodband. this happens when I call most aftrican country where the termination is not good. This is because when I use broadband the media has to through some kinds of NAT transval while the dialup normally gets public IP so no NAT transval and the media gets the shorts path. thanks Mohamed Jorge Crichigno <jcrichigno(a)conexion.com.py> wrote: I think the data rates with overhead are as follow: ---------------------------------------------------------------- iLBC - 30 ms (a packet each 30 ms) RTP payload: 50 bytes --> Rate: 13.3 Kbps RTP: 62 bytes UDP: 70 bytes IP: 90 bytes --> total rate: 24 Kbps ------------------------------------------------------------------- iLBC - 20 ms (a packet each 20 ms) RTP payload: 38 bytes --> Rate: 15.2 Kbps RTP: 50 bytes UDP: 58 bytes IP: 78 bytes --> total rate: 31.2 Kbps ------------------------------------------------------------------- G.729 - 2 voice frame per packet RTP payload: 20 bytes --> Rate: 8 Kbps RTP: 32 bytes UDP: 40 bytes IP: 60 bytes --> Total rate: 24 Kbps ------------------------------------------------------------------- G.729 - 4 voice frame per packet RTP payload: 40 bytes --> Rate: 8 Kbps RTP: 52 bytes UDP: 60 bytes IP: 80 bytes --> Total rate: 16 Kbps ------------------------------------------------------------------- G.711 - 20 ms (a packet each 20 ms) RTP payload: 160 bytes --> Rate: 64 Kbps RTP: 172 bytes UDP: 180 bytes IP: 200 bytes --> Total rate: 80 Kbps El vie, 27-05-2005 a las 10:36, Iqbal escribió: > the 5.3, 6.3K are really theoretical, i dont think they include IP > overheads, I used media_sessions.phtml, and looked at the actual calls > per codec, and I dont think u can really get a good call without 50-70K, > also most bandwidth providers (at least here in the UK) are > asymmetrical, so even on a 128K, u could have problems. > > Having said that I have done a nice call on xlite using ilbc on dial up. > > Iqbal > > Kofi Obiri-Yeboah wrote: > > >Just to add a few more details, Greger is right to point out the quality > >inferiority of G.723 compared to those of G711 and G729. In fact, in most VOIP > >deployments, in order to quarantee interoperability, a minimum bandwidth of > >128K is specified. However to reach the wider "lower bandwidth areas" most > >service providers are opting for G.723 which uses either 5.3 or 6.3K. At this > >low bandwidth transmission needs, one could literally reach "dial up modem" > >equipped areas. in fact most VOIP phone hardware and software are begining to > >specify G.723 as their default codec. Note that until the direct media > >connection phase of a VOIP vall setup, wide bandwidth is not required. Also > >note that analogue phones have a maximim bandwidth need of 3K, hence even the > >low quality of G.723/5.3K, compared to the average analogue phone call, is > >superior > > > > > > _______________________________________________ > 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 --------------------------------- Post your free ad now! Yahoo! Canada Personals

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[Serusers] Cant enable accounting with SER
by Tran Tien Duc 28 May '05

28 May '05

Hi All, I really need your help. I have setup the ser-0.8.14, uname �a is: Linux lnx01 2.6.9-1.667smp #1 SMP Tue Nov 14:59:52 EST 2004 i686 i386 GNU/Linux I have also setup Radiusclient-0.4.8 and Freeradius-1.0.2. I've followed all the steps in this tutorial http://www.iptel.org/ser/doc/ser_radius/ser_radius.html The installing don�t have any error. I tested on LAN with 1 server 192.168.10.1 and 2 workstations 192.168.10.2 and 192.168.10.3 However, I have received error when start ser. This is my error: 0(7389) find_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(7389) ERROR: acc: can't get code for the Failed attribute value 0(7389) init_mod(): Error while initializing module acc ERROR: error while initializing modules 0(7389) DEBUG: tm_shutdown : start 0(7389) DEBUG: tm_shutdown : empting DELETE list I should be most grateful to receive your help. Best regards, Tran Tien Duc ser.cfg debug=8 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc","db_url","mysql://ser:heslo@localhost/ser") # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("acc","db_url","mysql://ser:heslo@localhost/ser") modparam("acc","db_flag",1) modparam("acc","log_level",1) modparam("acc","log_flag",1) modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc","radius_flag",1) modparam("acc","radius_missed_flag",2) # ------------------------- 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; setflag(1); if (method=="INVITE") record_route(); # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); break; }; } --------------------------------- Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

1 1

[Serusers] Basic Asterisk Setup
by Samy Antoun 28 May '05

28 May '05

Hi, I have an Asterisk box up and running behind a NAT (Local IP Address 192.168.1.250). I would like to have another Linux box with SER as a proxy/registrar server on the same NAT (Local IP Address 192.168.1.150) to have a remote phone on ANOTHER NAT to register to the Asterisk server. My Questions are: SER Server: - Which ser.cfg shall I use, the default cfg file or any other cfg on the examples folder? - Do I have to have a Domain? - Do I have to Port Forwarding 5060 to the SER box? Asterisk Box: - What are the changes required? Client (Phone): - Do I have to use Outbound Proxy and point it to the WAN IP Address of the NATs Router? - Any other special configurations required? Any help is highly appreciated. Regards. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [serusers] fifo
by michael p 28 May '05

28 May '05

thank you very much but it didn't work too it's crazy ;) if someone else already used what i'm looking fo (serctl fifo t_uac_dlg ...) thanks M. >From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> >To: <ladia6(a)centrum.cz>, <mikep3000(a)hotmail.com>, <serusers(a)lists.iptel.org>, ><greger(a)teigre.com> >Subject: Re: [serusers] fifo >Date: Thu, 26 May 2005 10:49:17 +0200 > >Hi! > >I have never used the command line to use the t_uac_dlg fifo command, but >just a script similar to the example present in the attached file FIFO.txt >(it is from an older version of SER but the FIFO interface has not >changed). >I also attach another document containing the explanation found in the >source code about the FIFO interface. > >I hope with these two documents you will be able to send SIP requests with >t_uac_dlg > >Samuel. > > >Unclassified. > >>> "michael p" <mikep3000(a)hotmail.com> 05/25/05 04:32PM >>> >i'm sorry but it didn't help me > >i think it's different with serctl but thanks > >i don't understand anybody never used this command line in serctl??? > >thanks > > > > > >From: "Greger V. Teigre" <greger(a)teigre.com> > >To: "michael p" <mikep3000(a)hotmail.com>, <ladia6(a)centrum.cz>, > ><serusers(a)lists.iptel.org> > >Subject: Re: [serusers] fifo > >Date: Wed, 25 May 2005 16:26:44 +0200 > > > >Maybe this can help you in the right direction: > >http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 > >g-) > > > >michael p wrote: > >>only in the third link ther is my question but there is no answer > >>Does anyone know what correct format of a packet > >>should be pushed into fifo buffer? For example, > >> > >>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx > >>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx' > >>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . . > >>The above command I've tried,but got errors. > >> > >> > >>and i tried the format into the serctl too > >> > >> > >>thanks for replying > >> > >> > >> > >>>From: Ladislav Andel <ladia6(a)centrum.cz> > >>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> > >>>To: "michael p" <mikep3000(a)hotmail.com> > >>>Subject: Re: [serusers] fifo > >>>Date: Wed, 25 May 2005 15:39:07 +0200 > >>> > >>>I'm sorry, I think I've seen it in the list, but can't find it.. > >>> > >>>try the links below > >>> > >>>http://lists.iptel.org/pipermail/serusers/2004-August/010850.html > >>>http://lists.iptel.org/pipermail/serusers/2004-August/010825.html > >>>http://lists.iptel.org/pipermail/serusers/2005-April/018897.html > >>> > >>>mp> thanks but it didn't help me cause this topic not found in > >>>google in this > >>>mp> name maybe another one > >>> > >>>mp> so if you know which line i have to write it will be more simple > >>> > >>>mp> thanks > >>> > >>> > >>> > >>>>>From: Ladislav Andel <ladia6(a)centrum.cz> > >>>>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> > >>>>>To: "michael p" <mikep3000(a)hotmail.com> > >>>>>CC: serusers(a)lists.iptel.org > >>>>>Subject: Re: [serusers] fifo > >>>>>Date: Wed, 25 May 2005 14:55:45 +0200 > >>>>> > >>>>>This topic was in the list recently. Try to google it out. > >>>>>Lada > >>>>> > >>>>>mp> i already saw the different options but i want to know how to > >>>>>use them to > >>>>>mp> send a message like INVITE or OPTIONS i don't know how i have > >>>>>to write the > >>>>>mp> commande > >>>>> > >>>>>mp> thanks > >>>>> > >>>>>>>From: Llanos Serna García-Conde <llanosserna(a)hotmail.com> > >>>>>>>To: mikep3000(a)hotmail.com > >>>>>>>CC: serusers(a)lists.iptel.org > >>>>>>>Subject: RE: [serusers] fifo > >>>>>>>Date: Wed, 25 May 2005 14:20:27 +0200 > >>>>>>> > >>>>> > >>>>> > >>> > >>> > >>> > >> > >> > >>_______________________________________________ > >>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 > ><< FIFO.txt >> ><< FIFO_source.txt >>

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[Serusers] error in SQL query
by Christoph Hehl 28 May '05

28 May '05

hello. I have a Problem by login in Serveb. If i klick on the login button i get this errormessages: error in SQL query, line: 30 What's wrong? thanks

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[Serusers] serctl moni
by Richard Z 27 May '05

27 May '05

Hi, When i use "serctl moni", it shows, Transaction Statistics Current: 2 (47 waiting) Total: 19299 (12 local) I understand that ser keeps the sessions for 5 seconds before destroying them. But I saw the number of waiting increasing during the last few days. Do it mean that there is a memory leak or something? Thanks, Richard

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[Serusers] G723.1 vs G729
by info＠beeplove.com 27 May '05

27 May '05

Hello List: between G723.1 and G729, what codec do you prefer? I am just trying to understand pros and cons of both codec. Thanks, MOhammad -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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Re: [Serusers] G723.1 vs G729
by info＠beeplove.com 27 May '05

27 May '05

Just curious to know, Does this codec support depends on the actual hardware or the firmware? MOhammad Original Message: ----------------- From: Juha Heinanen jh(a)tutpro.com Date: Fri, 27 May 2005 17:48:36 +0300 To: info(a)beeplove.com, jcrichigno(a)conexion.com.py, serusers(a)iptel.org Subject: Re: [Serusers] G723.1 vs G729 info(a)beeplove.com writes: > Do you guys know about any hardphone that support Speex and iLBC codec? > Most of the hardphone in market support G723, G711 and G729. grandstream and snom, for example, have iLBC support. i have been told that iLBC is also on sipura roadmap, but who knows what will happen to that after the cisco deal. i have also heard rumors that iLBC support is coming to cisco ios, mainly because iLBC was chosen for cable tv terminals. -- juha -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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Re: [Serusers] G723.1 vs G729
by m36828253-1＠imap.1and1.com 27 May '05

27 May '05

I tried G723 on hardphone on dialup over mediaproxy. On Dialup UA end up/down stream was 24k/19k It was nice call but sometimes both party of both end noticed voice loss. Thanks, MOhammad Original Message: ----------------- From: Iqbal iqbal(a)gigo.co.uk Date: Fri, 27 May 2005 15:36:07 +0100 To: kofi(a)radiocomplex.com, serusers(a)iptel.org Subject: Re: [Serusers] G723.1 vs G729 the 5.3, 6.3K are really theoretical, i dont think they include IP overheads, I used media_sessions.phtml, and looked at the actual calls per codec, and I dont think u can really get a good call without 50-70K, also most bandwidth providers (at least here in the UK) are asymmetrical, so even on a 128K, u could have problems. Having said that I have done a nice call on xlite using ilbc on dial up. Iqbal Kofi Obiri-Yeboah wrote: >Just to add a few more details, Greger is right to point out the quality >inferiority of G.723 compared to those of G711 and G729. In fact, in most VOIP >deployments, in order to quarantee interoperability, a minimum bandwidth of >128K is specified. However to reach the wider "lower bandwidth areas" most >service providers are opting for G.723 which uses either 5.3 or 6.3K. At this >low bandwidth transmission needs, one could literally reach "dial up modem" >equipped areas. in fact most VOIP phone hardware and software are begining to >specify G.723 as their default codec. Note that until the direct media >connection phase of a VOIP vall setup, wide bandwidth is not required. Also >note that analogue phones have a maximim bandwidth need of 3K, hence even the >low quality of G.723/5.3K, compared to the average analogue phone call, is >superior > > _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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Re: [Serusers] G723.1 vs G729
by info＠beeplove.com 27 May '05

27 May '05

I tried G723 on hardphone on dialup over mediaproxy. On Dialup UA end up/down stream was 24k/19k It was nice call but sometimes both party of both end noticed voice loss. Thanks, MOhammad Original Message: ----------------- From: Iqbal iqbal(a)gigo.co.uk Date: Fri, 27 May 2005 15:36:07 +0100 To: kofi(a)radiocomplex.com, serusers(a)iptel.org Subject: Re: [Serusers] G723.1 vs G729 the 5.3, 6.3K are really theoretical, i dont think they include IP overheads, I used media_sessions.phtml, and looked at the actual calls per codec, and I dont think u can really get a good call without 50-70K, also most bandwidth providers (at least here in the UK) are asymmetrical, so even on a 128K, u could have problems. Having said that I have done a nice call on xlite using ilbc on dial up. Iqbal Kofi Obiri-Yeboah wrote: >Just to add a few more details, Greger is right to point out the quality >inferiority of G.723 compared to those of G711 and G729. In fact, in most VOIP >deployments, in order to quarantee interoperability, a minimum bandwidth of >128K is specified. However to reach the wider "lower bandwidth areas" most >service providers are opting for G.723 which uses either 5.3 or 6.3K. At this >low bandwidth transmission needs, one could literally reach "dial up modem" >equipped areas. in fact most VOIP phone hardware and software are begining to >specify G.723 as their default codec. Note that until the direct media >connection phase of a VOIP vall setup, wide bandwidth is not required. Also >note that analogue phones have a maximim bandwidth need of 3K, hence even the >low quality of G.723/5.3K, compared to the average analogue phone call, is >superior > > _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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Re: [Serusers] G723.1 vs G729
by info＠beeplove.com 27 May '05

27 May '05

Hi, Do you guys know about any hardphone that support Speex and iLBC codec? Most of the hardphone in market support G723, G711 and G729. Thanks, MOhammad Original Message: ----------------- From: Jorge Crichigno jcrichigno(a)conexion.com.py Date: Fri, 27 May 2005 16:03:23 -0400 To: jh(a)tutpro.com, serusers(a)iptel.org Subject: Re: [Serusers] G723.1 vs G729 Yes, iLBC has the following variants: ------------------------------------------------------------------- iLBC - 30 ms (one packet each 30 ms) RTP payload: 50 bytes --> Transmition rate: 13.3 Kbps RTP: 62 bytes UDP: 70 bytes IP: 90 bytes --> Transmition rate: 24 Kbps ------------------------------------------------------------------- iLBC - 20 ms (one packet each 20 ms) RTP payload: 38 bytes --> Transmition rate: 15.2 Kbps RTP: 50 bytes UDP: 58 bytes IP: 78 bytes --> Transmition rate: 31.2 Kbps ------------------------------------------------------------------- El vie, 27-05-2005 a las 09:45, Juha Heinanen escribiÃ³: > Jorge Crichigno writes: > > > In the following link there is a comparison between some codecs > > > > http://www.speex.org/comparison.html > > iLBC has also 20 ms mode which is not listed in the table. > > -- juha _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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Re: [Serusers] G723.1 vs G729
by info＠beeplove.com 27 May '05

27 May '05

This may be true for open source softphone. What about hardphone? MOhammad Original Message: ----------------- From: Juha Heinanen jh(a)tutpro.com Date: Fri, 27 May 2005 16:51:26 +0300 To: kofi(a)radiocomplex.com, serusers(a)iptel.org Subject: Re: [Serusers] G723.1 vs G729 the problem with BOTH g.723 and g.729 is that they are not free and thus cannot be included in free, open source sip UAs. for that reason, i don't see any reason to favor their use. the free alternatives are iLBC and speex. -- juha _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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Re: [Serusers] VoIp Billing Solution SIP server compatible!!!
by Manoj Ghag 27 May '05

27 May '05

hi, all Can't we go for development of Basic Billing solution in OPEN Source with SER only .. I mean we can make Billing Model ready and can go ahead with the same. Does anyone find any issue in same ... ? I think if we can do this .. it will be very helpful for many people .... Please let me know Regards Manoj Ghag Iqbal <iqbal(a)gigo.co.uk> wrote: Hi The reason why no one is really discussing much about billing here as open source is that most have rolled there own. serweb is a good starting point, but it is not a rating engine, it will just work out how long the call is. 1. Now when you have this length of call you need to work out how much its cost you, and how much it cost the end user... 2. For this you need to create a rates table or database 3. What I have is a prefix table which has 1800+ prefixes each one of these is bound to a gateway, and hence can be routed out differently, when required 4. I then have a cost table, this is what my buying price is 5. I then have a selling price table 6. For resellers we are building a drill down structure where they have there buying price and selling price 7. Now you need a rating structure 8. You may have a different price based on time of day and day of week etc etc 9. So now you have price per prefix, based on time of day and day of week 10. You see when call is made you do the maths and work out the cost to you , and cost to customers, if the former is lesser than the latter ur making money :-), at a gross level, if not, bad business model :-) 12. Different types of cutomers may get charged differently, i.e large corporates should always pay more, hence your price may also vary based on customer_type 13. Now if its postpaid you need to also do a bill run once a month, convert to pdf, and email out, just to make things easy, and then link to your legacy accounting system.. So you see everyone will have something slightly different, but I hope the above steps may help Iqbal On 5/21/2005, "Ollie Gejevski" wrote: >Hey guys, >I notice quite a few people are looking for a Billing solution. Please >contact me as I can help. We already have few companies from this board use >our solution with great success. We are Cisco as well as SIP server ( >SER, SNOM ) compatible. Check out our website HYPERLINK >"http://www.cyneric.com"www.cyneric.com Please contact me HYPERLINK >"mailto:ollie.g@cyneric.com"ollie.g(a)cyneric.com for more information. All >the best > >Ollie > > > >-- >No virus found in this outgoing message. >Checked by AVG Anti-Virus. >Version: 7.0.322 / Virus Database: 266.11.13 - Release Date: 5/19/2005 > > > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers --------------------------------- Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

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[Serusers] Ser 0.9.0 + Mediaproxy 1.3.1 - Problems with NATed Clients Authentication
by Felipe Martins 27 May '05

27 May '05

Hi everybody, I've configured SER to work with mediaproxy, I also configured mediaproxy.ini. My clients are authenticating normally, but only the clients that has a Public IP (ex. 200.201.145.146), all the cliets that are behind NAT can't REGISTER. What may be wrong ? Do anyone uses SER 0.9.0 with mysql authentication and Mediaproxy that could give me a hand ? My ser.cfg and mediaproxy.ini are as follows ================== ser.cfg ====================== debug=3 fork=yes log_stderror=no listen=192.0.2.13 # put your server IP address here port=5060 children=4 dns=no rev_dns=no 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.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.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("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # 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 (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ================== End of ser.cfg ====================== =================== mediaproxy.ini ====================== ; ; Configuration file for MediaProxy ; [Dispatcher] ; ; Section for configuring the proxy dispatcher ; ; The following options are available here: ; ; start Boolean value that specifies if to start the dispatcher. ; Default value: Yes ; ; socket Path to the UNIX socket where the dispatcher receives ; commands from SER. This should match the value for ; mediaproxy_socket in ser.cfg ; Default value: /var/run/proxydispatcher.sock ; ; group Put the socket in this group and make it group writable. ; Default value: ser ; ; defaultProxy Default mediaproxy to use in case the From/To domains ; involved in the call don't define any. ; Valid values for this are: ; ; - None ; don't use any default proxies. domains without ; mediaproxy SRV records won't work ; - /path/to/unix/socket ; use a single MediaProxy server identified by the given ; UNIX socket path ; - IP_or_hostname[:port] ; use a single MediaProxy server identified by its network ; address. The network address consists of an IP address ; or a hostname and an optional port number separated by ; a double colon. If port is missing 25060 will be assumed. ; Examples: ; 10.0.0.1 (connect to 10.0.0.1 on port 25060) ; 10.0.0.1:90 (connect to 10.0.0.1 on port 90) ; mp1.mydomain.com ; mp1.mydomain.com:7000 ; - domain://domain_name ; Use all MediaProxies defined by domain_name, honoring ; their priority and weight to create a cluster of proxies ; with fallback and load balancing capabilities. ; ; Default value: /var/run/mediaproxy.sock ; start = yes socket = /var/run/proxydispatcher.sock group = ser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] ; ; Section for configuring the MediaProxy server ; ; The following options are available here: ; ; start Boolean value that specifies if to start the RTP proxy server. ; Default value: Yes ; ; socket Path to the UNIX socket where MediaProxy receives ; commands from the dispatcher or SER. ; Default value: /var/run/mediaproxy.sock ; ; group Put the socket in this group and make it group writable. ; Default value: ser ; ; listen Network address where MediaProxy receives commands from ; a remote dispatcher. ; Valid values for this are: ; ; - None ; don't listen for network connections at all ; - address[:port] ; listen on the specified address and port ; address can be an IP a hostname or the keyword Any ; (in which case it will listen on 0.0.0.0). If address is ; a hostname, that should map in DNS to an IP address ; present on the machine through an A record. ; If port is missing assume 25060. ; ; Default value: None ; ; allow List of addresses that are allowed to connect to this ; MediaProxy server and send commands. ; They are specified as a comma separated list of entries, with ; each entry being specified in the CIDR network/mask notation ; (ex. 10.0.0.0/8) ; ; In addition simple IP addresses or hostnames are allowed, in ; which case the mask is considered to be 32. ; ; In addition to network ranges/addresses 2 keywords can be used ; for this option: ; None to specify that none is allowed to connect (not very ; useful but this is the default for security reasons) ; Any to specify that anyone is allowed to connect ; (dangerous!) ; ; Example: allow = 10.0.0.0/24, home-pc.mydomain.com, 1.2.3.4 ; ; Default value: None ; ; proxyIP IP address to use to talk to the phones. If not specified, the ; first found will be used. However first found usually means ; first defined in /etc/hosts which may not be what you want. ; If you find that the address that's automatically selected is ; not the one you want, you can specify the right one using this ; option. The address must be one that's present on one of the ; host's interfaces. ; ; portRange The range of ports to use for proxying the rtp streams. ; This option is specified as minport:maxport with minport and ; maxport being even numbers in the range 1024-65536 ; Default value: 35000:65000 ; ; TOS Unless you know what TOS means, leave this option alone. ; The TOS value can be specified either as a decimal number or ; as a hex number in the 0xnn format. ; Default value: 0xb8 ; ; idleTimeout Expire idle sessions after this much time. ; Default 60 seconds ; ; holdTimeout Expire calls on hold after this much time. ; Default value is 3600 seconds ; ; forceClose Forcibly close a RTP session after this many seconds even if ; it's still active. If forceClose is 0, then a session is never ; closed no matter how long it lasts. ; Default value: 0 ; start = yes socket = /var/run/mediaproxy.sock group = ser listen = 200.142.103.114 allow = any ;proxyIP = 10.0.0.1 portRange = 35000:36000 ;TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 accounting = off [Accounting] user = ser password = heslo host = any database = ser table = ser #[Accounting] #user = dbuser #password = dbpass #host = dbhost #database = radius #table = radacct =================== End of mediaproxy.ini ====================== -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

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[Serusers] ser & asterisk: sharing auth info
by Lol Zimmerli 27 May '05

27 May '05

Hello, I must use ser to forward calls to asterisk. It is a bit unclear for me on how to handle auth then. Should the user auth in ser and then ser forward auth info to asterisk? In this case, I must keep passwords in sync and it is not easy since asterisk store them as md5 hash (on the password) and ser can only handle plaintext or md5 hash, but on "user:realm:clearpassword". There's probably something I missed! I would be very happy if someone who did a user --> ser --> asterisk --> user can give us its ser.cfg (or parts of) and also correspondig tips on asterisk's sip.conf Thanks to the one(s) who can bring me back on earth! -- # Lol Zimmerli // S y s C o ® // http://www.sysco.ch/ Use free-form input when possible. - The Elements of Programming Style (Kernighan & Plaugher)

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Re : [Serusers] ser & asterisk: sharing auth info
by Ozan Blotter 27 May '05

27 May '05

Dear List & Friend, If you have any information or get anything like this i will be so glad with exact SER and Asterisk config files, simply tar/zip/rar them :) I'm able to setup a calling card application into Asterisk :) It's nice, GUI rocks , says you remaining credits/units , also CRM , so many things... All i need is install SER and Asterisk on same server... Planning to use SER as a frontend for SIP clients and Asterisk for PSTN Gateway with Voicemail, Invoice Generator, Call History, Calling Card etc etc :) Regards, Ozan Blotter Original Message >>> Message: 4 Date: Fri, 27 May 2005 01:14:41 +0200 From: Lol Zimmerli <lolzim(a)worldcom.ch> Subject: [Serusers] ser & asterisk: sharing auth info To: serusers(a)lists.iptel.org Message-ID: <auto-000125934405(a)smtp1.ipworldcom.ch> Content-Type: text/plain; charset=ISO-8859-15 Hello, I must use ser to forward calls to asterisk. It is a bit unclear for me on how to handle auth then. Should the user auth in ser and then ser forward auth info to asterisk? In this case, I must keep passwords in sync and it is not easy since asterisk store them as md5 hash (on the password) and ser can only handle plaintext or md5 hash, but on "user:realm:clearpassword". There's probably something I missed! I would be very happy if someone who did a user --> ser --> asterisk --> user can give us its ser.cfg (or parts of) and also correspondig tips on asterisk's sip.conf Thanks to the one(s) who can bring me back on earth! -- # Lol Zimmerli // S y s C o ® // http://www.sysco.ch/ Use free-form input when possible. - The Elements of Programming Style (Kernighan & Plaugher)

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[Serusers] /tmp/ser_fifo permissions
by Llanos Serna García-Conde 27 May '05

27 May '05

3 2

[Serusers] Mysql with tcp only
by Elton Machado 27 May '05

27 May '05

Hi, Does anybody knows how can i have ser working with mysql in another box? I have only TCP not socks :( I'm getting this: 0(78861) new_connection(): Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) 0(78861) db_init(): Could not create a connection 0(78861) ERROR:domain_db_init: cannot initialize database connection 0(78861) ERROR: domain:child_init(): Unable to connect to the database When i try to run it :S Thanks in advance.

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[Serusers] RE: I'm interested about SER and LDAP, could you help me to set upthis environment to test it?
by Marc Sutter 27 May '05

27 May '05

Lee, Be carefull with radiusclient. radiusclient-ng will NOT work with ser auth_radius module. I had to use radiusclient 0.4.8 and not the ng 5.0 version, otherwise the ser radius modules will not compile. you can get it from http://developer.berlios.de/project/showfiles.php?group_id=1208 We should continue to communicate through the mailinglist, so other users with same problems could also solve their issues. Best regards Marc On ven, 2005-05-27 at 09:11 +0800, SA5-S0B200-李環祐-新一代IT服務部 wrote: > Hi Marc, > > Thanks for your kindness! I've installed freeradius and radiusclient-nt. But after that, I found my SER did'nt work. It makes me to reinstall SER again. > > It's still on the process of reinstallation. > > Best regards, > Hammer Lee > > -----Original Message----- > From: Marc Sutter [mailto:marc.sutter@xilane.ch] > Sent: Thursday, May 26, 2005 3:55 PM > To: SA5-S0B200-李環祐-新一代IT服務部 > Subject: Re: I'm interested about SER and LDAP, could you help me to set upthis environment to test it? > > Hi Lee, > > I'm working on a little howto to archive my work on this project. > > So, if you can wait some day, I'll send you a copy of this document. > > Cheers > > Marc > > On mer, 2005-05-25 at 10:29 +0800, SA5-S0B200-李環祐-新一代IT服務部 wrote: > > Mr. Sutter, > > > > Sorry for bothering you! I'm interested about SER and LDAP, too. Could > > you explant me how to set up this environment to test it? > > > > Thanks, > > Hammer Lee

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Re: [Serusers] avp_radius and avpops
by Greger V. Teigre 27 May '05

27 May '05

Do the debugging dance, so that you can verify what the calls to avp_* do... g-) Ricardo Martinez wrote: > Hello Greg > Thanks for your answer. > I made the change but it seems not to do anything.... > Can someone help me here? > > Thanks in advance > > Regards > Ricardo.- > >> -----Mensaje original----- >> De: Greger V. Teigre [mailto:greger@teigre.com] >> Enviado el: Miércoles, 25 de Mayo de 2005 16:24 >> Para: Ricardo Martinez; serusers(a)lists.iptel.org >> Asunto: Re: [Serusers] avp_radius and avpops >> >> >>> if( !avp_pushto("$RURI", "s:var1/g")) { >> replace with: >> "s:$var1/g" >> I don't know about the case sensivtivity of destination, I >> always use lower >> case... >> g-) >> >> Ricardo Martinez wrote: >>> Hello List. >>> I have a question regarding the use of avp_radius and avpops. I'm >>> using avp_radius to obtain an AVP value from my database via radius. >>> What i what to do is replace this value for the RURI. >>> Here i have a couple of questions. >>> 1.- The value returned by the avp_radius (the SIP-AVP) where is >>> stored ? It suppose that the SIP-AVP returned by radius >> has the form >>> of "name:value". That "name" refers to the name of what?. >>> For example i'm returning : >> "var1:sip:1234567@mydomain.com". What i >>> see in the debug is : >>> >>> avp_load_user: AVP 'var1'='sip:1234567@mydomain.com' has been added >>> >>> This is what i got in my ser.cfg (a snippet). >>> >>> if (method=="INVITE" || method=="CANCEL") { >>> if( !avp_load_radius("caller")) { >>> log (1, "AVP_RADIUS: Fail on avp_radius\n"); >>> }; >> >>> >>> if( !avp_pushto("$RURI", "s:var1/g")) { >>> log (1, "AVPOPS: Fail on AVPOPS\n"); >>> }; >>> }; >>> >>> Again the debug = 9 . >>> >>> 6(23815) avp_load_user: AVP >> 'var1'='sip:1234567@mydomain.com' has been >>> added >>> 6(23815) qm_free(0x8123400, 0x8166ccc), called from avp_radius.c: >>> load_avp_user(344) >>> 6(23815) qm_free: freeing frag. 0x8166cb4 alloc'ed from >>> avp_radius.c: load_avp_user(330) >>> 6(23815) DEBUG:avpops:pushto_avp: no avp found >>> 6(23815) AVPOPS: Fail on AVPOPS >>> >>> What i'm doing wrong? >>> Thanks! >>> >>> Regards >>> Ricardo Martinez >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] problem with mysql installation
by l9ayd amadius 27 May '05

27 May '05

Hello I have a problem when installing mysql. I have downloaded the mysql package and Wen i type the command: rpm -i ser-mysql-0.8.8-2.i386.rpm, i have the message : Retrieving ftp://ftp.berlios.de/pub/ser/0.8.8/packages/redhat/7.x</title> error: skipping ftp://ftp.berlios.de/pub/ser/0.8.8/packages/redhat/7.x</title> - transfer failed - Unknown or unexpected error I don't understand the cause of the problem. Can someone please explain me the nature of the problem and how can i overcome it??! Thanks in advance!!!! _________________________________________________________________ Vidéoconférence plein écran avec MSN Messenger http://g.msn.fr/FR1001/866

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[Serusers] db_url not working
by kumar gokhare 27 May '05

27 May '05

hello fellows, everytime i get this message while running ser,i am new to ser,so plz help me out n mail me if you can find a way out of this mail to : kumargs84(a)yahoo.co.in 0(15400) set_mod_param_regex: parameter <db_url> not found in module <acc> 0(15400) parse error (76,60-61): Can't set module parameter ERROR: bad config file (1 errors) here's my ser.cfg file, # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) #Uncomment these lines to enter debugging mode debug=3 log_stderror=yes listen=127.0.0.1 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #---usrloc param's--- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #---acc param's --- modparam("acc", "log_level", 2) modparam("acc", "log_flag", 2) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("acc", "db_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ # if (method=="REGISTER") { # setflag(1); # setflag(2); # log(1, "REGISTER received\n"); # } else { # log(1, "non-REGISTER received\n"); # }; # if (uri=~"sip:.*[@:]voip.brujula.net") { # log(1, "request for voip.brujula.net received\n"); # } else { # log(1, "request for other domain received\n"); # }; # 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; # Process record-routing if (loose_route()) { t_relay(); break; }; # labeled all transaction for accounting setflag(1); # record-route INVITES to make sure BYEs will visit our server too if (method=="INVITE") record_route(); # forward the request statefuly now; (we need *stateful* forwarding, # because the stateful mode correlates requests with replies and # drops retranmissions; otherwise, we would have to report on # every single message received) if (!t_relay()) { sl_reply_error(); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication setflag(1); setflag(2); log(1,"REGISTER : Authenticating user\n"); if (!www_authorize("voip.brujula.net","subscriber")) { log(1,"REGISTER : Challenging user\n"); www_challenge("voip.brujula.net", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; }

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[Serusers] Request for test calls
by msg 27 May '05

27 May '05

Greetings: I would appreciate receiving test calls anytime at sip:9200@cybertheque.org; the only codec allowed is G.729. Especially helpful are calls originating in Europe and calls over very low bandwidth (dial-up) connections originating anywhere. I will be looking at latencies, packet loss, jitter, etc. I will answer the call whenever I am near the console. Much thanks, Michael Grigoni Cybertheque Museum

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[Serusers] Re: SER CANCEL Processing
by a c 27 May '05

27 May '05

any ideas anyone? seems like SER does not respond to CANCEL requests and eventually takes the failure route. I have the following in my failure route, but it never gets hit, since SER does not send the 200 OK back or pass the CANCEL message forward. log file attached. failure_route[1] { if(t_check_status("487")) { # don't continue on cancellation xlog("L_WARN", "OH OH OH\n"); break; } ac --- a c <lra101(a)yahoo.com> wrote: > Seems like SER does not respond back to CANCEL > Requests. Is this a issue / some config problem on > my > side? ngrep traces attached > > interface: eth0 (192.168.0.0/255.255.255.0) > filter: ip and ( port 5060 ) > # > U 192.168.0.2:5062 -> 192.168.0.5:5060 > INVITE sip:9999999999@voip.com SIP/2.0..Via: > SIP/2.0/UDP 192.168.0.2:5062;r > > port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > 102 <sip:102@voi > p.com:5062>;tag=3282942385..To: > <sip:9999999999@voip.com>..Contact: <sip:10 > 2@192.168.0.2:5062>..Call-ID: > 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. > 0.2..CSeq: 940 INVITE..Max-Forwards: > 70..Content-Type: application/sdp..Use > r-Agent: X-Lite release 1103m..Content-Length: > 258....v=0..o=102 7659613 76 > 59633 IN IP4 192.168.0.2..s=X-Lite..c=IN IP4 > 192.168.0.2..t=0 0..m=audio 80 > 00 RTP/AVP 0 8 3 98 96..a=rtpmap:0 > pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpma > p:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:96 > telephone-event/8000..a=fm > tp:96 0-15.. > > > # > U 192.168.0.5:5060 -> 192.168.0.2:5062 > SIP/2.0 100 trying -- your call is important to > us..Via: SIP/2.0/UDP 192.16 > > 8.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > 102 <sip:102@voip.com:5062>;tag=3282942385..To: > <sip:9999999999@voip.com>. > .Call-ID: > 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: > 940 INVIT > E..Server: Sip EXpress router (0.9.2 > (i386/linux))..Content-Length: 0..Warn > ing: 392 192.168.0.5:5060 "Noisy feedback tells: > pid=2943 req_src_ip=192.1 > 68.0.2 req_src_port=5062 > in_uri=sip:9999999999@voip.com out_uri=sip:9999999 > 999(a)voip.com via_cnt==1".... > > > # > U 192.168.0.5:5060 -> 192.168.0.3:5060 > INVITE sip:9999999999@voip.com > SIP/2.0..Record-Route: <sip:192.168.0.5;ftag > =3282942385;lr=on>..Via: SIP/2.0/UDP > 192.168.0.5;branch=z9hG4bKe034.1018c5d > 1.0..Via: SIP/2.0/UDP > 192.168.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8C > F4A479B6D9DDB17C5CFC1..From: 102 > <sip:102@voip.com:5062>;tag=3282942385..To > : <sip:9999999999@voip.com>..Contact: > <sip:102@192.168.0.2:5062>..Call-ID: > > 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: > 940 INVITE..Max-For > wards: 16..Content-Type: > application/sdp..User-Agent: X-Lite release 1103m. > .Content-Length: 258....v=0..o=102 7659613 7659633 > IN IP4 192.168.0.2..s=X- > Lite..c=IN IP4 192.168.0.2..t=0 0..m=audio 8000 > RTP/AVP 0 8 3 98 96..a=rtpm > ap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 > gsm/8000..a=rtpmap:98 iLBC > /8000..a=rtpmap:96 telephone-event/8000..a=fmtp:96 > 0-15.. > # > U 192.168.0.3:5060 -> 192.168.0.5:5060 > SIP/2.0 100 Trying..From: 102 > <sip:102@voip.com:5062>;tag=3282942385..To: < > sip:9999999999@voip.com>..Call-Id: > 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192 > .168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP > 192.168.0.5;branch=z9hG4bKe034 > .1018c5d1.0..Via: SIP/2.0/UDP > 192.168.0.2:5062;rport=5062;branch=z9hG4bKE29 > 503D9D8CF4A479B6D9DDB17C5CFC1..Content-Length: > 0.... > > # > U 192.168.0.3:5060 -> 192.168.0.5:5060 > SIP/2.0 180 Ringing..From: 102 > <sip:102@voip.com:5062>;tag=3282942385..To: > <sip:9999999999@voip.com>;tag=27378..Call-Id: > 29469D19-CDF1-49ED-8245-A67E1 > B1DBB88@192.168.0.2..Cseq: 940 INVITE..Via: > SIP/2.0/UDP 192.168.0.5;branch= > z9hG4bKe034.1018c5d1.0..Via: SIP/2.0/UDP > 192.168.0.2:5062;rport=5062;branch > =z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..Date: > Mon, > 23 May 2005 22:15:14 G > MT..Contact: sip:192.168.0.3..User-Agent: > sipX/2.5.2 > (WinNT)..Accept-Langua > ge: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, > OPTIONS, NOTIFY, REGISTER, > SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, > replaces..Content-Length: 0 > .... > > > # > U 192.168.0.5:5060 -> 192.168.0.2:5062 > SIP/2.0 180 Ringing..From: 102 > <sip:102@voip.com:5062>;tag=3282942385..To: > <sip:9999999999@voip.com>;tag=27378..Call-Id: > 29469D19-CDF1-49ED-8245-A67E1 > B1DBB88@192.168.0.2..Cseq: 940 INVITE..Via: > SIP/2.0/UDP 192.168.0.2:5062;rp > > ort=5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..Date: > Mon, 23 May > 2005 22:15:14 GMT..Contact: > sip:192.168.0.3..User-Agent: sipX/2.5.2 (WinNT) > ..Accept-Language: en..Allow: INVITE, ACK, CANCEL, > BYE, REFER, OPTIONS, NOT > IFY, REGISTER, SUBSCRIBE..Supported: sip-cc, > sip-cc-01, timer, replaces..Co > ntent-Length: 0.... > > > # > U 192.168.0.2:5062 -> 192.168.0.5:5060 > CANCEL sip:9999999999@voip.com SIP/2.0..Via: > SIP/2.0/UDP 192.168.0.2:5062;r > > port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > 102 <sip:102@voi > p.com:5062>;tag=3282942385..To: > <sip:9999999999@voip.com>..Contact: <sip:10 > 2@192.168.0.2:5062>..Call-ID: > 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. > 0.2..CSeq: 940 CANCEL..Max-Forwards: > 70..User-Agent: > X-Lite release 1103m.. > Content-Length: 0.... > > > # > U 192.168.0.2:5062 -> 192.168.0.5:5060 > CANCEL sip:9999999999@voip.com SIP/2.0..Via: > SIP/2.0/UDP 192.168.0.2:5062;r > > port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > 102 <sip:102@voi > p.com:5062>;tag=3282942385..To: > <sip:9999999999@voip.com>..Contact: <sip:10 > 2@192.168.0.2:5062>..Call-ID: > 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. > 0.2..CSeq: 940 CANCEL..Max-Forwards: > 70..User-Agent: > X-Lite release 1103m.. > Content-Length: 0.... > > > # > U 192.168.0.2:5062 -> 192.168.0.5:5060 > CANCEL sip:9999999999@voip.com SIP/2.0..Via: > SIP/2.0/UDP 192.168.0.2:5062;r > > port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > 102 <sip:102@voi > p.com:5062>;tag=3282942385..To: > <sip:9999999999@voip.com>..Contact: <sip:10 > 2@192.168.0.2:5062>..Call-ID: > 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. > 0.2..CSeq: 940 CANCEL..Max-Forwards: > 70..User-Agent: > X-Lite release 1103m.. > Content-Length: 0.... > > > # > U 192.168.0.5:5060 -> 192.168.0.2:5062 > SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP > 192.168.0.2:5062;rport=5062;b > > ranch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: > 102 <sip:102@voip.com: > 5062>;tag=3282942385..To: > <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 > ae5b6e96e979a-6e9f..Call-ID: > 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168.0 > .2..CSeq: 940 INVITE..Server: Sip EXpress router > (0.9.2 (i386/linux))..Cont > ent-Length: 0..Warning: 392 192.168.0.5:5060 > "Noisy > feedback tells: pid=29 > 51 req_src_ip=192.168.0.2 req_src_port=5062 > in_uri=sip:9999999999@voip.com > === message truncated === __________________________________________________ Do You Yahoo!? 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Mail has the best spam protection around http://mail.yahoo.com <<< Log from 192.168.0.5 started May 26, 2005, 10:53:10 >>> # U 192.168.0.2:5062 -> 192.168.0.5:5060 INVITE sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..From: 102 <sip:102@voi p.com:5062>;tag=2354273279..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF(a)192.168. 0.2..CSeq: 55437 INVITE..Max-Forwards: 70..Content-Type: application/sdp..U ser-Agent: X-Lite release 1103m..Content-Length: 262....v=0..o=102 24393782 4 243937884 IN IP4 192.168.0.2..s=X-Lite..c=IN IP4 192.168.0.2..t=0 0..m=au dio 8000 RTP/AVP 0 8 3 98 96..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a =rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:96 telephone-event/8000 ..a=fmtp:96 0-15.. # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.16 8.0.2:5062;rport=5062;branch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..From: 102 <sip:102@voip.com:5062>;tag=2354273279..To: <sip:9999999999@voip.com>. .Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF@192.168.0.2..CSeq: 55437 INV ITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length: 0..Wa rning: 392 192.168.0.5:5060 "Noisy feedback tells: pid=991 req_src_ip=192. 168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:999999 9999(a)voip.com via_cnt==1".... # U 192.168.0.5:5060 -> 192.168.0.2:5060 INVITE sip:9999999999@voip.com SIP/2.0..Record-Route: <sip:192.168.0.5;ftag =2354273279;lr=on>..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK078b.d58ef0f 1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;branch=z9hG4bKC51A010562E F4A59816876A6DEACF5BF..From: 102 <sip:102@voip.com:5062>;tag=2354273279..To : <sip:9999999999@voip.com>..Contact: <sip:102@192.168.0.2:5062>..Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF@192.168.0.2..CSeq: 55437 INVITE..Max-F orwards: 16..Content-Type: application/sdp..User-Agent: X-Lite release 1103 m..Content-Length: 262....v=0..o=102 243937824 243937884 IN IP4 192.168.0.2 ..s=X-Lite..c=IN IP4 192.168.0.2..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 96.. a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:9 8 iLBC/8000..a=rtpmap:96 telephone-event/8000..a=fmtp:96 0-15.. # U 192.168.0.2:5060 -> 192.168.0.5:5060 SIP/2.0 100 Trying..From: 102 <sip:102@voip.com:5062>;tag=2354273279..To: < sip:9999999999@voip.com>..Call-Id: 60D7D848-8486-4E68-8191-05684A2A3EEF@192 .168.0.2..Cseq: 55437 INVITE..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK07 8b.d58ef0f1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;branch=z9hG4bKC 51A010562EF4A59816876A6DEACF5BF..Content-Length: 0.... # U 192.168.0.2:5060 -> 192.168.0.5:5060 SIP/2.0 180 Ringing..From: 102 <sip:102@voip.com:5062>;tag=2354273279..To: <sip:9999999999@voip.com>;tag=19901..Call-Id: 60D7D848-8486-4E68-8191-05684 A2A3EEF@192.168.0.2..Cseq: 55437 INVITE..Via: SIP/2.0/UDP 192.168.0.5;branc h=z9hG4bK078b.d58ef0f1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;bran ch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..Date: Thu, 26 May 2005 15:53:13 GMT..Contact: sip:192.168.0.2..User-Agent: sipX/2.5.2 (WinNT)..Accept-Lang uage: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER , SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces..Content-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 180 Ringing..From: 102 <sip:102@voip.com:5062>;tag=2354273279..To: <sip:9999999999@voip.com>;tag=19901..Call-Id: 60D7D848-8486-4E68-8191-05684 A2A3EEF@192.168.0.2..Cseq: 55437 INVITE..Via: SIP/2.0/UDP 192.168.0.2:5062; rport=5062;branch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..Date: Thu, 26 Ma y 2005 15:53:13 GMT..Contact: sip:192.168.0.2..User-Agent: sipX/2.5.2 (WinN T)..Accept-Language: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, N OTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces.. Content-Length: 0.... # U 192.168.0.2:5062 -> 192.168.0.5:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..From: 102 <sip:102@voi p.com:5062>;tag=2354273279..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF(a)192.168. 0.2..CSeq: 55437 CANCEL..Max-Forwards: 70..User-Agent: X-Lite release 1103m ..Content-Length: 0.... # U 192.168.0.2:5062 -> 192.168.0.5:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..From: 102 <sip:102@voi p.com:5062>;tag=2354273279..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF(a)192.168. 0.2..CSeq: 55437 CANCEL..Max-Forwards: 70..User-Agent: X-Lite release 1103m ..Content-Length: 0.... # U 192.168.0.2:5062 -> 192.168.0.5:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..From: 102 <sip:102@voi p.com:5062>;tag=2354273279..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF(a)192.168. 0.2..CSeq: 55437 CANCEL..Max-Forwards: 70..User-Agent: X-Lite release 1103m ..Content-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;b ranch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..From: 102 <sip:102@voip.com: 5062>;tag=2354273279..To: <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 ae5b6e96e979a-7cb8..Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF(a)192.168.0 .2..CSeq: 55437 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Co ntent-Length: 0..Warning: 392 192.168.0.5:5060 "Noisy feedback tells: pid= 996 req_src_ip=192.168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999999@voip.com via_cnt==0".... # U 192.168.0.5:5060 -> 192.168.0.2:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.5;branch =z9hG4bK078b.d58ef0f1.0..From: 102 <sip:102@voip.com:5062>;tag=2354273279.. Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF@192.168.0.2..To: <sip:9999999 999(a)voip.com>..CSeq: 55437 CANCEL..User-Agent: Sip EXpress router(0.9.2 (i3 86/linux))..Content-Length: 0.... # U 192.168.0.2:5060 -> 192.168.0.5:5060 SIP/2.0 200 OK..From: 102 <sip:102@voip.com:5062>;tag=2354273279..To: <sip: 9999999999(a)voip.com>..Call-Id: 60D7D848-8486-4E68-8191-05684A2A3EEF(a)192.168 .0.2..Cseq: 55437 CANCEL..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK078b.d 58ef0f1.0..Content-Length: 0.... # U 192.168.0.2:5060 -> 192.168.0.5:5060 SIP/2.0 487 Request Terminated..From: 102 <sip:102@voip.com:5062>;tag=23542 73279..To: <sip:9999999999@voip.com>;tag=19901..Call-Id: 60D7D848-8486-4E68 -8191-05684A2A3EEF@192.168.0.2..Cseq: 55437 INVITE..Via: SIP/2.0/UDP 192.16 8.0.5;branch=z9hG4bK078b.d58ef0f1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rpor t=5062;branch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..Date: Thu, 26 May 20 05 15:53:43 GMT..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, R EGISTER, SUBSCRIBE..User-Agent: sipX/2.5.2 (WinNT)..Accept-Language: en..Su pported: sip-cc, sip-cc-01, timer, replaces..Contact: sip:192.168.0.2..Cont ent-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.2:5060 ACK sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.5;branch=z9 hG4bK078b.d58ef0f1.0..From: 102 <sip:102@voip.com:5062>;tag=2354273279..Cal l-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF@192.168.0.2..To: <sip:9999999999 @voip.com>;tag=19901..CSeq: 55437 ACK..User-Agent: Sip EXpress router(0.9.2 (i386/linux))..Content-Length: 0.... # U 192.168.0.2:5060 -> 192.168.0.5:5060 SIP/2.0 200 OK..From: 102 <sip:102@voip.com:5062>;tag=2354273279..To: <sip: 9999999999(a)voip.com>..Call-Id: 60D7D848-8486-4E68-8191-05684A2A3EEF(a)192.168 .0.2..Cseq: 55437 CANCEL..Contact: <sip:9999999999@192.168.0.2:5060>..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK078b.d58ef0f1.0..Date: Thu, 26 May 20 05 15:53:43 GMT..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, R EGISTER, SUBSCRIBE..User-Agent: sipX/2.5.2 (WinNT)..Accept-Language: en..Su pported: sip-cc, sip-cc-01, timer, replaces..Content-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;b ranch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..From: 102 <sip:102@voip.com: 5062>;tag=2354273279..To: <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 ae5b6e96e979a-7cb8..Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF(a)192.168.0 .2..CSeq: 55437 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Co ntent-Length: 0..Warning: 392 192.168.0.5:5060 "Noisy feedback tells: pid= 996 req_src_ip=192.168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999999@voip.com via_cnt==0".... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;b ranch=z9hG4bKC51A010562EF4A59816876A6DEACF5BF..From: 102 <sip:102@voip.com: 5062>;tag=2354273279..To: <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 ae5b6e96e979a-7cb8..Call-ID: 60D7D848-8486-4E68-8191-05684A2A3EEF(a)192.168.0 .2..CSeq: 55437 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Co ntent-Length: 0..Warning: 392 192.168.0.5:5060 "Noisy feedback tells: pid= 996 req_src_ip=192.168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999999@voip.com via_cnt==0".... <<< Log from 192.168.0.5 ended May 26, 2005, 10:54:17 >>>

4 6

[Serusers] Linux Version
by Rosa De Santis 26 May '05

26 May '05

Hello all. does SER works fine on Linux Suse 9.1 ? Thanks in advance Ro

1 0

RE: [Serusers] BYE messages are lost :(
by Glenn Dalgliesh 26 May '05

26 May '05

I think you need to have a set flag that includes loose routes would be my guess _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ozan Blotter Sent: Wednesday, May 25, 2005 5:32 PM To: serusers(a)lists.iptel.org Subject: [Serusers] BYE messages are lost :( Hi SerUsers, I'm using SER 0.9.2 on Fedora Core 3, it works good, i can do MySql Auth. but i couldn't make it work with Accounting Module. I get INVITE and ACK messages only in Database as: ? ? <sip:12345@192.168.1.9;user=phone>;tag=xIzNxkDMxID <sip:54321@192.168.1.9>;tag=xIzM5MDMxID 200 INVITE sip:54321@192.168.1.9 sip:54321@192.168.1.201:5060 sip:12345@192.168.1.9;user=phone sip:54321@192.168.1.9 950D1B912D91212(a)192.168.1.200 12345 192.168.1.9 xIzNxkDMxID xIzM5MDMxID 2005-05-25 21:20:05 20050526002005 ? ? <sip:12345@192.168.1.9;user=phone>;tag=xIzNxkDMxID <sip:54321@192.168.1.9>;tag=xIzM5MDMxID 200 ACK sip:192.168.1.9 sip:54321@192.168.1.201:5060 sip:12345@192.168.1.9;user=phone sip:54321@192.168.1.9 950D1B912D91212(a)192.168.1.200 12345 192.168.1.9 xIzNxkDMxID xIzM5MDMxID 2005-05-25 21:20:05 20050526002005 It never gave me BYE messages :( Below is my ser.cfg >>>>>>>>>>>>>>>>>>> debug=3 fork=yes log_stderror=no listen=192.168.1.9 # put your server IP address here port=5060 children=4 dns=no rev_dns=no 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/acc.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) modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "log_fmt", "fimos") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 2) modparam("acc", "early_media", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "log_flag", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; setflag(1); # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< Thanks, PS : is there anyone for basic billing for SER. CDRtool from agProject seems okay but i have no idea, i need a billing directly from SER's database, not Radius. A PHP Coder can do that within a couple of hours i believe.

2 2

Re: [Serusers] video and SER
by Samuel Osorio Calvo 26 May '05

26 May '05

kphone slo supports video with vic Unclassified. >>> mike jones <mbjngn(a)optonline.net> 05/26/05 05:16PM >>> Hello Does anyone have a recent experience using SER with video endpoints like Econf from france telecom. Does it work? Does any know of any open source Video clients? thanks Michael B. Jones 973 879 2066 cell HYPERLINK "http://promos.hotbar.com/promos/promodll.dll?RunPromo&El=&SG=&RAND=6920 1&partner=hbtools"Upgrade Your Email - Click here! -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 5/25/2005

2 1

[Serusers] auth-int support
by Ravi 26 May '05

26 May '05

Hi All: Does SER (0.8.14) have support for auth-int qop? If not, does anyone know when this is likely to be added? thanks, -Ravi.

1 0

[Serusers] ser+radius accounting
by Tiziano 26 May '05

26 May '05

Hi, i've succesfully setup ser+radius auth. Now i'm trying to do something interesting with accouting on radius for billing purposes but i notice strange way of behaving... i've tried different solutions with combinations of setflag(1) and acc_rad_request("200"); looks like everithing is ok if u call someone and this answers. (just that accounting starts before the party answers) but i sb doesn't answer and u cancel or if u gets 404 i have just an accounting start without stop packet. does sb have a working example of radius accounting? i've already tried a lot of combinations without success... Thanks, Tiziano -- Tiziano <ml(a)asdasd.it>

1 0

[Serusers] H.350 directory services
by Marc Sutter 26 May '05

26 May '05

Hi all, I'm looking for a solution to use H.350 directory services to authentificate UA's through ser. Is ther any ldap module for authentification out there ? Thanks for your help Marc

1 0

[Serusers] video and SER
by mike jones 26 May '05

26 May '05

Hello Does anyone have a recent experience using SER with video endpoints like Econf from france telecom. Does it work? Does any know of any open source Video clients? thanks Michael B. Jones 973 879 2066 cell HYPERLINK "http://promos.hotbar.com/promos/promodll.dll?RunPromo&El=&SG=&RAND=6920 1&partner=hbtools"Upgrade Your Email - Click here! -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 5/25/2005

2 1

[Serusers] where does the acc module get its time from
by Iqbal 26 May '05

26 May '05

Hi I have realised that since I changed to BST on all my servers (using NTP to keep in sync) the time stored for acc requests has not changed, it is still one hour behind. Iqbal

3 2

RE: [Serusers] serctl moni
by Cesar Hernandez 26 May '05

26 May '05

Hello Jiri, For your information I found a way to reproduce the problem by mistake. After reworking my script I was missing a t_reply() in a failure_route[] and on every busy call the transaction waiting counter keep increasing but never decreased after. failure_route[6] { if (t_check_status("486")) { xlog("L_INFO","Call busy: r-uri <%ru> to <%tu> from <%fu>\n"); ##### This t_reply was missing ######## t_reply("486","Busy Here"); break; } else if (t_check_status("404")) { xlog("L_WARN","TRUNK BUSY: r-uri <%ru> to <%tu> from <%fu>\n"); } else { xlog("L_WARN","CALL FAILED: r-uri <%ru> to <%tu> from <%fu>\n"); }; t_reply("480","Temporary Unavailable"); } I looked quickly at the code but was not able to figure out where the problem is hiding, then I had to move on something else. Now with the t_reply in place the counter still increases over time but at slower rate. Maybe this will give you a hint, when you'll have a chance to look at it. -Cesar -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Jiri Kuthan Sent: Thursday, May 26, 2005 2:20 AM To: Richard Z; serusers(a)lists.iptel.org Subject: Re: [Serusers] serctl moni At 11:28 PM 5/25/2005, Richard Z wrote: >Hi, > >When i use "serctl moni", it shows, > >Transaction Statistics >Current: 2 (47 waiting) Total: 19299 (12 local) > >I understand that ser keeps the sessions for 5 seconds before >destroying them. But I saw the number of waiting increasing during the >last few days. > >Do it mean that there is a memory leak or something? I think that there is an error to discrease the counter in some cases. It has been starving on my to-do-list for ages and I am not sure when it will make out of it. -jiri >Thanks, >Richard > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Problem with mysql installation
by l9ayd amadius 26 May '05

26 May '05

Hello I have a problem when installing mysql. I have downloaded the mysql package and Wen i type the command: rpm -i ser-mysql-0.8.8-2.i386.rpm, i have the message : Retrieving ftp://ftp.berlios.de/pub/ser/0.8.8/packages/redhat/7.x</title> error: skipping ftp://ftp.berlios.de/pub/ser/0.8.8/packages/redhat/7.x</title> - transfer failed - Unknown or unexpected error I don't understand the cause of the problem. Can someone please explain me the nature of the problem and how can i overcome it??! Thanks in advance!!!! _________________________________________________________________ Bénéficiez de 250 Mo de stockage avec MSN Hotmail http://g.msn.fr/FR1001/866

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RE: [Serusers] avp_radius and avpops
by Ricardo Martinez 26 May '05

26 May '05

Hello Greg Thanks for your answer. I made the change but it seems not to do anything.... Can someone help me here? Thanks in advance Regards Ricardo.- > -----Mensaje original----- > De: Greger V. Teigre [mailto:greger@teigre.com] > Enviado el: Miércoles, 25 de Mayo de 2005 16:24 > Para: Ricardo Martinez; serusers(a)lists.iptel.org > Asunto: Re: [Serusers] avp_radius and avpops > > > > if( !avp_pushto("$RURI", "s:var1/g")) { > replace with: > "s:$var1/g" > I don't know about the case sensivtivity of destination, I > always use lower > case... > g-) > > Ricardo Martinez wrote: > > Hello List. > > I have a question regarding the use of avp_radius and avpops. I'm > > using avp_radius to obtain an AVP value from my database via radius. > > What i what to do is replace this value for the RURI. > > Here i have a couple of questions. > > 1.- The value returned by the avp_radius (the SIP-AVP) where is > > stored ? It suppose that the SIP-AVP returned by radius > has the form > > of "name:value". That "name" refers to the name of what?. > > For example i'm returning : > "var1:sip:1234567@mydomain.com". What i > > see in the debug is : > > > > avp_load_user: AVP 'var1'='sip:1234567@mydomain.com' has been added > > > > This is what i got in my ser.cfg (a snippet). > > > > if (method=="INVITE" || method=="CANCEL") { > > if( !avp_load_radius("caller")) { > > log (1, "AVP_RADIUS: Fail on avp_radius\n"); > > }; > > > > > if( !avp_pushto("$RURI", "s:var1/g")) { > > log (1, "AVPOPS: Fail on AVPOPS\n"); > > }; > > }; > > > > Again the debug = 9 . > > > > 6(23815) avp_load_user: AVP > 'var1'='sip:1234567@mydomain.com' has been > > added > > 6(23815) qm_free(0x8123400, 0x8166ccc), called from avp_radius.c: > > load_avp_user(344) > > 6(23815) qm_free: freeing frag. 0x8166cb4 alloc'ed from > avp_radius.c: > > load_avp_user(330) > > 6(23815) DEBUG:avpops:pushto_avp: no avp found > > 6(23815) AVPOPS: Fail on AVPOPS > > > > What i'm doing wrong? > > Thanks! > > > > Regards > > Ricardo Martinez > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Cannot compile ser 0.9.0
by Roger Lewau 26 May '05

26 May '05

I downloaded the source for ser 0.9.0 but I am unsuccessfull to make. All I get is this output: "Makefile.sources", line 17: Missing dependency operator "Makefile.sources", line 19: Need an operator "Makefile", line 56: Missing dependency operator Error expanding embedded variable. I am using FreeBSD 5.4 and I have both Bison and gmake installed. Anyone with a bright idea on what can be wrong? /roger

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Re: [serusers] fifo
by Samuel Osorio Calvo 26 May '05

26 May '05

Let's try if I can help you... First of all, why don't you want to try the example script I gave in the first e-mail???? It worked with me and I did not try the command line option.... The HF unparseable error is produced because the callid header must be the string Call-ID (and not as I wrote before). If you don't put Call-ID or Cseq headers, t_uac_dlg will add some default numbers. Depending on what you want to achieve, you must put the right values, for example: *to send an outside dialog request you can ommit Call-ID and Cseq (be carefull with Call-ID in REGISTER). *to send a within-dialog request (e.g. re-INVITE, BYE), you must get the Call-ID and Cseq someway (ethereal, SER logs, ngrep....) and insert it into the command line header. Moreover, be carefull with the From: and To: tags. If you don't specify them, SER will add to the From: header field value the ;tag=xxxxx, which is OK for requests initiating dialogs but not for the rest. These latter must carry the right From: and To: tags. t_uac_dlg does not parse properly the <>, so, do: serctl fifo t_uac_dlg BYE sip:user@IP . To:sip:daemon@ipServer From:sip:user@ip . . ended with <space>.<space>.<space> (otherwise it will wait for the body) This worked for me, Samuel Unclassified. >>> "michael p" <mikep3000(a)hotmail.com> 05/26/05 12:37PM >>> thanks serctl fifo t_uac_dlg BYE sip:1111@ip1 . "To:<sip:222@here> From:<sip:1111@ip1> Callid:xxxx Cseq:xxxx" . . i replaced sip:1111@ip1 by the URI of one user, "To:<sip:222@here> by "To:<sip:daemon@ipserServer> , From:<sip:1111@ip1> by the URI of the user, and after i don't know what exactly i have to put on Callid and Cseq and i'm not sure that's obliged (for invite or options) i tried with bye to put a Callid of a real session and to cut off it but the error message is : 400 fifo_uac: HFs unparseable after i tried: serctl fifo t_uac_dlg OPTIONS sip:michael@195.68.115.168:52786 . "To:<sip:daemon@195.68.105.215> From:<sip:michael@195.68.115.168:52786> . .. and it gave me : > you know like it waits something do you know more to help me? thanks >From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> >To: <ladia6(a)centrum.cz>, <mikep3000(a)hotmail.com>, <serusers(a)lists.iptel.org>, ><greger(a)teigre.com> >Subject: Re: [serusers] fifo >Date: Thu, 26 May 2005 12:01:49 +0200 > >First of all: which error gives you serctl? >Then, try something like: > >serctl fifo t_uac_dlg BYE sip:1111@ip1 . "To:<sip:222@here> >From:<sip:1111@ip1> Callid:xxxx Cseq:xxxx" . . > > >Hope it works! >Samuel > > >Unclassified. > >>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx > > >>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx' > > >>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . . > > > >>> "michael p" <mikep3000(a)hotmail.com> 05/26/05 11:19AM >>> >thank you very much > >but it didn't work too > >it's crazy ;) > >if someone else already used what i'm looking fo (serctl fifo t_uac_dlg >....) > >thanks > >M. > > > >From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> > >To: <ladia6(a)centrum.cz>, <mikep3000(a)hotmail.com>, <serusers(a)lists.iptel.org>, > ><greger(a)teigre.com> > >Subject: Re: [serusers] fifo > >Date: Thu, 26 May 2005 10:49:17 +0200 > > > >Hi! > > > >I have never used the command line to use the t_uac_dlg fifo command, but > >just a script similar to the example present in the attached file >FIFO.txt > >(it is from an older version of SER but the FIFO interface has not > >changed). > >I also attach another document containing the explanation found in the > >source code about the FIFO interface. > > > >I hope with these two documents you will be able to send SIP requests >with > >t_uac_dlg > > > >Samuel. > > > > > >Unclassified. > > >>> "michael p" <mikep3000(a)hotmail.com> 05/25/05 04:32PM >>> > >i'm sorry but it didn't help me > > > >i think it's different with serctl but thanks > > > >i don't understand anybody never used this command line in serctl??? > > > >thanks > > > > > > > > > > >From: "Greger V. Teigre" <greger(a)teigre.com> > > >To: "michael p" <mikep3000(a)hotmail.com>, <ladia6(a)centrum.cz>, > > ><serusers(a)lists.iptel.org> > > >Subject: Re: [serusers] fifo > > >Date: Wed, 25 May 2005 16:26:44 +0200 > > > > > >Maybe this can help you in the right direction: > > >http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 > > >g-) > > > > > >michael p wrote: > > >>only in the third link ther is my question but there is no answer > > >>Does anyone know what correct format of a packet > > >>should be pushed into fifo buffer? For example, > > >> > > >>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx > > >>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx' > > >>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . . > > >>The above command I've tried,but got errors. > > >> > > >> > > >>and i tried the format into the serctl too > > >> > > >> > > >>thanks for replying > > >> > > >> > > >> > > >>>From: Ladislav Andel <ladia6(a)centrum.cz> > > >>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> > > >>>To: "michael p" <mikep3000(a)hotmail.com> > > >>>Subject: Re: [serusers] fifo > > >>>Date: Wed, 25 May 2005 15:39:07 +0200 > > >>> > > >>>I'm sorry, I think I've seen it in the list, but can't find it.. > > >>> > > >>>try the links below > > >>> > > >>>http://lists.iptel.org/pipermail/serusers/2004-August/010850.html > > >>>http://lists.iptel.org/pipermail/serusers/2004-August/010825.html > > >>>http://lists.iptel.org/pipermail/serusers/2005-April/018897.html > > >>> > > >>>mp> thanks but it didn't help me cause this topic not found in > > >>>google in this > > >>>mp> name maybe another one > > >>> > > >>>mp> so if you know which line i have to write it will be more simple > > >>> > > >>>mp> thanks > > >>> > > >>> > > >>> > > >>>>>From: Ladislav Andel <ladia6(a)centrum.cz> > > >>>>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> > > >>>>>To: "michael p" <mikep3000(a)hotmail.com> > > >>>>>CC: serusers(a)lists.iptel.org > > >>>>>Subject: Re: [serusers] fifo > > >>>>>Date: Wed, 25 May 2005 14:55:45 +0200 > > >>>>> > > >>>>>This topic was in the list recently. Try to google it out. > > >>>>>Lada > > >>>>> > > >>>>>mp> i already saw the different options but i want to know how to > > >>>>>use them to > > >>>>>mp> send a message like INVITE or OPTIONS i don't know how i have > > >>>>>to write the > > >>>>>mp> commande > > >>>>> > > >>>>>mp> thanks > > >>>>> > > >>>>>>>From: Llanos Serna García-Conde <llanosserna(a)hotmail.com> > > >>>>>>>To: mikep3000(a)hotmail.com > > >>>>>>>CC: serusers(a)lists.iptel.org > > >>>>>>>Subject: RE: [serusers] fifo > > >>>>>>>Date: Wed, 25 May 2005 14:20:27 +0200 > > >>>>>>> > > >>>>> > > >>>>> > > >>> > > >>> > > >>> > > >> > > >> > > >>_______________________________________________ > > >>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 > > > ><< FIFO.txt >> > ><< FIFO_source.txt >> > >

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Re: [serusers] fifo
by michael p 26 May '05

26 May '05

thanks serctl fifo t_uac_dlg BYE sip:1111@ip1 . "To:<sip:222@here> From:<sip:1111@ip1> Callid:xxxx Cseq:xxxx" . . i replaced sip:1111@ip1 by the URI of one user, "To:<sip:222@here> by "To:<sip:daemon@ipserServer> , From:<sip:1111@ip1> by the URI of the user, and after i don't know what exactly i have to put on Callid and Cseq and i'm not sure that's obliged (for invite or options) i tried with bye to put a Callid of a real session and to cut off it but the error message is : 400 fifo_uac: HFs unparseable after i tried: serctl fifo t_uac_dlg OPTIONS sip:michael@195.68.115.168:52786 . "To:<sip:daemon@195.68.105.215> From:<sip:michael@195.68.115.168:52786> . . and it gave me : > you know like it waits something do you know more to help me? thanks >From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> >To: <ladia6(a)centrum.cz>, <mikep3000(a)hotmail.com>, <serusers(a)lists.iptel.org>, ><greger(a)teigre.com> >Subject: Re: [serusers] fifo >Date: Thu, 26 May 2005 12:01:49 +0200 > >First of all: which error gives you serctl? >Then, try something like: > >serctl fifo t_uac_dlg BYE sip:1111@ip1 . "To:<sip:222@here> >From:<sip:1111@ip1> Callid:xxxx Cseq:xxxx" . . > > >Hope it works! >Samuel > > >Unclassified. > >>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx > > >>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx' > > >>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . . > > > >>> "michael p" <mikep3000(a)hotmail.com> 05/26/05 11:19AM >>> >thank you very much > >but it didn't work too > >it's crazy ;) > >if someone else already used what i'm looking fo (serctl fifo t_uac_dlg >....) > >thanks > >M. > > > >From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> > >To: <ladia6(a)centrum.cz>, <mikep3000(a)hotmail.com>, <serusers(a)lists.iptel.org>, > ><greger(a)teigre.com> > >Subject: Re: [serusers] fifo > >Date: Thu, 26 May 2005 10:49:17 +0200 > > > >Hi! > > > >I have never used the command line to use the t_uac_dlg fifo command, but > >just a script similar to the example present in the attached file >FIFO.txt > >(it is from an older version of SER but the FIFO interface has not > >changed). > >I also attach another document containing the explanation found in the > >source code about the FIFO interface. > > > >I hope with these two documents you will be able to send SIP requests >with > >t_uac_dlg > > > >Samuel. > > > > > >Unclassified. > > >>> "michael p" <mikep3000(a)hotmail.com> 05/25/05 04:32PM >>> > >i'm sorry but it didn't help me > > > >i think it's different with serctl but thanks > > > >i don't understand anybody never used this command line in serctl??? > > > >thanks > > > > > > > > > > >From: "Greger V. Teigre" <greger(a)teigre.com> > > >To: "michael p" <mikep3000(a)hotmail.com>, <ladia6(a)centrum.cz>, > > ><serusers(a)lists.iptel.org> > > >Subject: Re: [serusers] fifo > > >Date: Wed, 25 May 2005 16:26:44 +0200 > > > > > >Maybe this can help you in the right direction: > > >http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 > > >g-) > > > > > >michael p wrote: > > >>only in the third link ther is my question but there is no answer > > >>Does anyone know what correct format of a packet > > >>should be pushed into fifo buffer? For example, > > >> > > >>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx > > >>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx' > > >>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . . > > >>The above command I've tried,but got errors. > > >> > > >> > > >>and i tried the format into the serctl too > > >> > > >> > > >>thanks for replying > > >> > > >> > > >> > > >>>From: Ladislav Andel <ladia6(a)centrum.cz> > > >>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> > > >>>To: "michael p" <mikep3000(a)hotmail.com> > > >>>Subject: Re: [serusers] fifo > > >>>Date: Wed, 25 May 2005 15:39:07 +0200 > > >>> > > >>>I'm sorry, I think I've seen it in the list, but can't find it.. > > >>> > > >>>try the links below > > >>> > > >>>http://lists.iptel.org/pipermail/serusers/2004-August/010850.html > > >>>http://lists.iptel.org/pipermail/serusers/2004-August/010825.html > > >>>http://lists.iptel.org/pipermail/serusers/2005-April/018897.html > > >>> > > >>>mp> thanks but it didn't help me cause this topic not found in > > >>>google in this > > >>>mp> name maybe another one > > >>> > > >>>mp> so if you know which line i have to write it will be more simple > > >>> > > >>>mp> thanks > > >>> > > >>> > > >>> > > >>>>>From: Ladislav Andel <ladia6(a)centrum.cz> > > >>>>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> > > >>>>>To: "michael p" <mikep3000(a)hotmail.com> > > >>>>>CC: serusers(a)lists.iptel.org > > >>>>>Subject: Re: [serusers] fifo > > >>>>>Date: Wed, 25 May 2005 14:55:45 +0200 > > >>>>> > > >>>>>This topic was in the list recently. Try to google it out. > > >>>>>Lada > > >>>>> > > >>>>>mp> i already saw the different options but i want to know how to > > >>>>>use them to > > >>>>>mp> send a message like INVITE or OPTIONS i don't know how i have > > >>>>>to write the > > >>>>>mp> commande > > >>>>> > > >>>>>mp> thanks > > >>>>> > > >>>>>>>From: Llanos Serna García-Conde <llanosserna(a)hotmail.com> > > >>>>>>>To: mikep3000(a)hotmail.com > > >>>>>>>CC: serusers(a)lists.iptel.org > > >>>>>>>Subject: RE: [serusers] fifo > > >>>>>>>Date: Wed, 25 May 2005 14:20:27 +0200 > > >>>>>>> > > >>>>> > > >>>>> > > >>> > > >>> > > >>> > > >> > > >> > > >>_______________________________________________ > > >>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 > > > ><< FIFO.txt >> > ><< FIFO_source.txt >> > >

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[Serusers] SER - change To header field
by root linux 26 May '05

26 May '05

Hi, I have the following config for my college project: - SIP Phones (2.2.2.1) --> SER (2.2.2.2) --> Asterisk (2.2.2.3) --> Clarent C5CM (1.1.1.1) When I make a mobile phone call, I get the following debugging message from Asterisk as below: - Sip read: ACK sip:0193839090@mydomain.com SIP/2.0 Record-Route: <sip:0193839090@2.2.2.2;ftag=i7hY4i2wlrWTpUf0;lr=on> Via: SIP/2.0/UDP 2.2.2.2;branch=0 Via: SIP/2.0/UDP 202.187.27.51:5060;branch=z9hG4bKp7kxPljBf Max-Forwards: 69 User-Agent: PA168S From: "20004" <sip:20004@mydomain.com>;tag=i7hY4i2wlrWTpUf0 To: "0193839090" <sip:0193839090@mydomain.com>;tag=as65d5f808 Call-ID: 8sSYKxJJddC1CAmz(a)2.2.2.1 Contact: <sip:20004@2.2.2.1:5060> CSeq: 1 ACK Content-Length: 0 Is it possible to rewrite the "To Request-URI) to something like sip:0193839090@1.1.1.1 Regards, rootlinux __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/

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Re: [serusers] fifo
by Samuel Osorio Calvo 26 May '05

26 May '05

First of all: which error gives you serctl? Then, try something like: serctl fifo t_uac_dlg BYE sip:1111@ip1 . "To:<sip:222@here> From:<sip:1111@ip1> Callid:xxxx Cseq:xxxx" . . Hope it works! Samuel Unclassified. >>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx > >>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx' > >>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . . >>> "michael p" <mikep3000(a)hotmail.com> 05/26/05 11:19AM >>> thank you very much but it didn't work too it's crazy ;) if someone else already used what i'm looking fo (serctl fifo t_uac_dlg ....) thanks M. >From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> >To: <ladia6(a)centrum.cz>, <mikep3000(a)hotmail.com>, <serusers(a)lists.iptel.org>, ><greger(a)teigre.com> >Subject: Re: [serusers] fifo >Date: Thu, 26 May 2005 10:49:17 +0200 > >Hi! > >I have never used the command line to use the t_uac_dlg fifo command, but >just a script similar to the example present in the attached file FIFO.txt >(it is from an older version of SER but the FIFO interface has not >changed). >I also attach another document containing the explanation found in the >source code about the FIFO interface. > >I hope with these two documents you will be able to send SIP requests with >t_uac_dlg > >Samuel. > > >Unclassified. > >>> "michael p" <mikep3000(a)hotmail.com> 05/25/05 04:32PM >>> >i'm sorry but it didn't help me > >i think it's different with serctl but thanks > >i don't understand anybody never used this command line in serctl??? > >thanks > > > > > >From: "Greger V. Teigre" <greger(a)teigre.com> > >To: "michael p" <mikep3000(a)hotmail.com>, <ladia6(a)centrum.cz>, > ><serusers(a)lists.iptel.org> > >Subject: Re: [serusers] fifo > >Date: Wed, 25 May 2005 16:26:44 +0200 > > > >Maybe this can help you in the right direction: > >http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 > >g-) > > > >michael p wrote: > >>only in the third link ther is my question but there is no answer > >>Does anyone know what correct format of a packet > >>should be pushed into fifo buffer? For example, > >> > >>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx > >>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx' > >>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . . > >>The above command I've tried,but got errors. > >> > >> > >>and i tried the format into the serctl too > >> > >> > >>thanks for replying > >> > >> > >> > >>>From: Ladislav Andel <ladia6(a)centrum.cz> > >>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> > >>>To: "michael p" <mikep3000(a)hotmail.com> > >>>Subject: Re: [serusers] fifo > >>>Date: Wed, 25 May 2005 15:39:07 +0200 > >>> > >>>I'm sorry, I think I've seen it in the list, but can't find it.. > >>> > >>>try the links below > >>> > >>>http://lists.iptel.org/pipermail/serusers/2004-August/010850.html > >>>http://lists.iptel.org/pipermail/serusers/2004-August/010825.html > >>>http://lists.iptel.org/pipermail/serusers/2005-April/018897.html > >>> > >>>mp> thanks but it didn't help me cause this topic not found in > >>>google in this > >>>mp> name maybe another one > >>> > >>>mp> so if you know which line i have to write it will be more simple > >>> > >>>mp> thanks > >>> > >>> > >>> > >>>>>From: Ladislav Andel <ladia6(a)centrum.cz> > >>>>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> > >>>>>To: "michael p" <mikep3000(a)hotmail.com> > >>>>>CC: serusers(a)lists.iptel.org > >>>>>Subject: Re: [serusers] fifo > >>>>>Date: Wed, 25 May 2005 14:55:45 +0200 > >>>>> > >>>>>This topic was in the list recently. Try to google it out. > >>>>>Lada > >>>>> > >>>>>mp> i already saw the different options but i want to know how to > >>>>>use them to > >>>>>mp> send a message like INVITE or OPTIONS i don't know how i have > >>>>>to write the > >>>>>mp> commande > >>>>> > >>>>>mp> thanks > >>>>> > >>>>>>>From: Llanos Serna García-Conde <llanosserna(a)hotmail.com> > >>>>>>>To: mikep3000(a)hotmail.com > >>>>>>>CC: serusers(a)lists.iptel.org > >>>>>>>Subject: RE: [serusers] fifo > >>>>>>>Date: Wed, 25 May 2005 14:20:27 +0200 > >>>>>>> > >>>>> > >>>>> > >>> > >>> > >>> > >> > >> > >>_______________________________________________ > >>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 > ><< FIFO.txt >> ><< FIFO_source.txt >>

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Re: [serusers] fifo
by Samuel Osorio Calvo 26 May '05

26 May '05

Hi! I have never used the command line to use the t_uac_dlg fifo command, but just a script similar to the example present in the attached file FIFO.txt (it is from an older version of SER but the FIFO interface has not changed). I also attach another document containing the explanation found in the source code about the FIFO interface. I hope with these two documents you will be able to send SIP requests with t_uac_dlg Samuel. Unclassified. >>> "michael p" <mikep3000(a)hotmail.com> 05/25/05 04:32PM >>> i'm sorry but it didn't help me i think it's different with serctl but thanks i don't understand anybody never used this command line in serctl??? thanks >From: "Greger V. Teigre" <greger(a)teigre.com> >To: "michael p" <mikep3000(a)hotmail.com>, <ladia6(a)centrum.cz>, ><serusers(a)lists.iptel.org> >Subject: Re: [serusers] fifo >Date: Wed, 25 May 2005 16:26:44 +0200 > >Maybe this can help you in the right direction: >http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 >g-) > >michael p wrote: >>only in the third link ther is my question but there is no answer >>Does anyone know what correct format of a packet >>should be pushed into fifo buffer? For example, >> >>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx >>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx' >>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . . >>The above command I've tried,but got errors. >> >> >>and i tried the format into the serctl too >> >> >>thanks for replying >> >> >> >>>From: Ladislav Andel <ladia6(a)centrum.cz> >>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> >>>To: "michael p" <mikep3000(a)hotmail.com> >>>Subject: Re: [serusers] fifo >>>Date: Wed, 25 May 2005 15:39:07 +0200 >>> >>>I'm sorry, I think I've seen it in the list, but can't find it.. >>> >>>try the links below >>> >>>http://lists.iptel.org/pipermail/serusers/2004-August/010850.html >>>http://lists.iptel.org/pipermail/serusers/2004-August/010825.html >>>http://lists.iptel.org/pipermail/serusers/2005-April/018897.html >>> >>>mp> thanks but it didn't help me cause this topic not found in >>>google in this >>>mp> name maybe another one >>> >>>mp> so if you know which line i have to write it will be more simple >>> >>>mp> thanks >>> >>> >>> >>>>>From: Ladislav Andel <ladia6(a)centrum.cz> >>>>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> >>>>>To: "michael p" <mikep3000(a)hotmail.com> >>>>>CC: serusers(a)lists.iptel.org >>>>>Subject: Re: [serusers] fifo >>>>>Date: Wed, 25 May 2005 14:55:45 +0200 >>>>> >>>>>This topic was in the list recently. Try to google it out. >>>>>Lada >>>>> >>>>>mp> i already saw the different options but i want to know how to >>>>>use them to >>>>>mp> send a message like INVITE or OPTIONS i don't know how i have >>>>>to write the >>>>>mp> commande >>>>> >>>>>mp> thanks >>>>> >>>>>>>From: Llanos Serna García-Conde <llanosserna(a)hotmail.com> >>>>>>>To: mikep3000(a)hotmail.com >>>>>>>CC: serusers(a)lists.iptel.org >>>>>>>Subject: RE: [serusers] fifo >>>>>>>Date: Wed, 25 May 2005 14:20:27 +0200 >>>>>>> >>>>> >>>>> >>> >>> >>> >> >> >>_______________________________________________ >>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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[Serusers] Broken Calls
by Ozan Blotter 26 May '05

26 May '05

Hi List, Now i'm able to log INVITE,ACK,BYE,CANCEL messages into database, the new problem is broken calls: think that a SIP user 12345 dials 54321, the telephone rings, conversation is okay, they talk let say 1 minute and a bit more, suddenly caller or callee unplugs the ATA, so there's no BYE message :( How can i setup it to log that broken calls without CANCEL or BYE messages? Thanks, Ozan Blotter here's my SER.CFG >>>>>>>>>>>>>>>>>>>>>>>> [root@localhost ser]# cat ser.cfg debug=3 fork=yes log_stderror=no listen=192.168.1.9 # put your server IP address here port=5060 children=4 dns=no rev_dns=no 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/acc.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", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "log_fmt", "fimos") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 2) modparam("acc", "early_media", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "log_flag", 1) route { setflag(1); # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="INVITE" || method=="BYE" || method=="CANCEL") { setflag(1); route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } [root@localhost ser]# >>>>>>>>>>>>>>>>>>>>>>>>

1 0

[Serusers] No module matching vm found
by Llanos Serna García-Conde 26 May '05

26 May '05

3 2

[Serusers] BYE messages are lost :(
by Ozan Blotter 25 May '05

25 May '05

Hi SerUsers, I'm using SER 0.9.2 on Fedora Core 3, it works good, i can do MySql Auth. but i couldn't make it work with Accounting Module. I get INVITE and ACK messages only in Database as: ? ? <sip:12345@192.168.1.9;user=phone>;tag=xIzNxkDMxID <sip:54321@192.168.1.9>;tag=xIzM5MDMxID 200 INVITE sip:54321@192.168.1.9 sip:54321@192.168.1.201:5060 sip:12345@192.168.1.9;user=phone sip:54321@192.168.1.9 950D1B912D91212(a)192.168.1.200 12345 192.168.1.9 xIzNxkDMxID xIzM5MDMxID 2005-05-25 21:20:05 20050526002005 ? ? <sip:12345@192.168.1.9;user=phone>;tag=xIzNxkDMxID <sip:54321@192.168.1.9>;tag=xIzM5MDMxID 200 ACK sip:192.168.1.9 sip:54321@192.168.1.201:5060 sip:12345@192.168.1.9;user=phone sip:54321@192.168.1.9 950D1B912D91212(a)192.168.1.200 12345 192.168.1.9 xIzNxkDMxID xIzM5MDMxID 2005-05-25 21:20:05 20050526002005 It never gave me BYE messages :( Below is my ser.cfg >>>>>>>>>>>>>>>>>>> debug=3 fork=yes log_stderror=no listen=192.168.1.9 # put your server IP address here port=5060 children=4 dns=no rev_dns=no 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/acc.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) modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "log_fmt", "fimos") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 2) modparam("acc", "early_media", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "log_flag", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; setflag(1); # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< Thanks, PS : is there anyone for basic billing for SER. CDRtool from agProject seems okay but i have no idea, i need a billing directly from SER's database, not Radius. A PHP Coder can do that within a couple of hours i believe.

2 1

[Serusers] avp_radius and avpops
by Ricardo Martinez 25 May '05

25 May '05

Hello List. I have a question regarding the use of avp_radius and avpops. I'm using avp_radius to obtain an AVP value from my database via radius. What i what to do is replace this value for the RURI. Here i have a couple of questions. 1.- The value returned by the avp_radius (the SIP-AVP) where is stored ? It suppose that the SIP-AVP returned by radius has the form of "name:value". That "name" refers to the name of what?. For example i'm returning : "var1:sip:1234567@mydomain.com". What i see in the debug is : avp_load_user: AVP 'var1'='sip:1234567@mydomain.com' has been added This is what i got in my ser.cfg (a snippet). if (method=="INVITE" || method=="CANCEL") { if( !avp_load_radius("caller")) { log (1, "AVP_RADIUS: Fail on avp_radius\n"); }; if( !avp_pushto("$RURI", "s:var1/g")) { log (1, "AVPOPS: Fail on AVPOPS\n"); }; }; Again the debug = 9 . 6(23815) avp_load_user: AVP 'var1'='sip:1234567@mydomain.com' has been added 6(23815) qm_free(0x8123400, 0x8166ccc), called from avp_radius.c: load_avp_user(344) 6(23815) qm_free: freeing frag. 0x8166cb4 alloc'ed from avp_radius.c: load_avp_user(330) 6(23815) DEBUG:avpops:pushto_avp: no avp found 6(23815) AVPOPS: Fail on AVPOPS What i'm doing wrong? Thanks! Regards Ricardo Martinez

2 1

[Serusers] Issues finding modules for SER
by Anthony Dean 25 May '05

25 May '05

I got the config file someone posted to this list a couple of days back and it fits almost perfectly for what we need ser to do (voicemail, with asterisk or similar AND nat using mediaproxy). However it required a few other modules that dont come with the standard SER package (I am currently using 0.8.14 - which I have working for a simple config). 0(32034) ERROR: load_module: could not open module </usr/local/lib/ser/modules/avpops.so>: 0(32034) ERROR: load_module: could not open module </usr/local/lib/ser/modules/options.so>: 0(32034) ERROR: load_module: could not open module </usr/local/lib/ser/modules/speeddial.so>: 0(32034) ERROR: load_module: could not open module </usr/local/lib/ser/modules/uri_db.so>: 0(32034) ERROR: load_module: could not open module </usr/local/lib/ser/modules/xdz_tools.so>: Can anyone let me know where these modules are to download or better yet, email the tar file with them. I have also tried onsip.org which seems to have some great info on configuring ser with the same modules above as well but I can't download any onf the files (and from the FAQ/Forum I see that its not just me but everyone can't download from them). Would really appreciate it if someone could help me find the above modules. Thanks Tony Dean

2 2

[Serusers] no Digest-Algorithm = "MD5"
by davor jovanovic 25 May '05

25 May '05

Hi all, I'm using SER 0.8.14, FreeRadius 1.0.2. and XLite 1103. My question is about MD5, why I can't see Digest-Algorithm = "MD5" in FreeRadius and XLite client outputs. Is this a problem in XLite or SER? Everything is configured as in HOW TO and I get authenticated with FreeRadius so I suppose that configuration is OK. Dictionaries are the same in radiusclient on SER machine and in FreeRadius. I can provide more outputs if someone is interested to help. Any help would be appreciated. *****This is output from FreeRadius Server****** (no Digest-Algorithm = "MD5" received) ... rad_recv: Access-Request packet from host 161.53.0.131:33854, id=131, length=207 User-Name = "djovanovic.srce" Digest-Attributes = 0x0a11646a6f76616e6f7669632e73726365 Digest-Attributes = 0x0109737263652e6872 Digest-Attributes = 0x022a3432393439336464646235643930376433653433323765333063366566646566336436 6137643933 Digest-Attributes = 0x040d7369703a737263652e6872 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "263ec6802b382ddf8d58b363fd0629ad" Service-Type = Sip-Session Sip-Uri-User = "djovanovic.srce" NAS-IP-Address = 161.53.0.131 NAS-Port = 5060 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 0 modcall[authorize]: module "preprocess" returns ok for request 0 modcall[authorize]: module "chap" returns noop for request 0 modcall[authorize]: module "mschap" returns noop for request 0 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "djovanovic.srce" Digest-Realm = "srce.hr" Digest-Nonce = "429493dddb5d907d3e4327e30c6efdef3d6a7d93" Digest-URI = "sip:srce.hr" Digest-Method = "REGISTER" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 0 ... *****This is output from XLite****** (no Digest-Algorithm = "MD5" sent) SEND TIME: 203074718 SEND >> 191.153.206.58:5060 INVITE sip:537@srce.hr SIP/2.0 Via: SIP/2.0/UDP 191.153.206.57:5060;rport;branch=z9hG4bK8BB5DE10CC404EEBABE6761213B4AA45 From: davor <sip:djovanovic.srce@srce.hr>;tag=2361756632 To: <sip:537@srce.hr> Contact: <sip:djovanovic.srce@191.153.206.57:5060> Call-ID: 146A8327-BCC6-42E6-A027-83EA1832631B(a)191.153.206.57 CSeq: 40898 INVITE Authorization: Digest username="djovanovic.srce",realm="srce.hr",nonce="429492e156f0af6f7ae5e821d7 3df9be3e04f360",response="17a54d30cc2d5ecf1cc8e714d40210e0",uri="sip:537@src e.hr" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 308

1 0

[Serusers] Can't invite
by Rosa De Santis 25 May '05

25 May '05

Hello all. I'm new with this and I'm making test with 3 machines in the same network using SER-SEMS. I receive the REGISTER messages from the users, but I can't make an INVITE from one user to another. I've tried with stateless forwarding, with no result. Please, some help needed. Could anybody tell me what's missing or wrong ? This is the ser.cfg that i'm using. Thanks in advance Ro # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode debug=9 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/vm.so" #loadmodule "/usr/local/lib/ser/modules/dbtext.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") modparam("usrloc", "db_url","sql://ser:heslo@localhost/ser") modparam("auth_db", "db_url","sql://ser:heslo@localhost/ser") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ----------------- setting module-specific parameters --------------- modparam("voicemail", "db_url","sql://ser:heslo@localhost/ser") #modparam("vm", "db_url", "sql://ser:heslo@localhost/ser") modparam("voicemail", "email_column", "email_address") modparam("voicemail", "subscriber_table", "subscriber") modparam("voicemail", "user_column", "username") modparam("voicemail", "domain_column", "domain") # ------------------------- request routing logic ------------------- alias="sip.vivophone.com" alias="201.249.37.26" # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("70")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("sip.vivophone.com", "subscriber")) { www_challenge("sip.vivophone.com", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB #if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # break; #}; # Voicemail specific configuration - begin if( (!lookup("location")) && (method=="ACK" || method=="INVITE" || method=="BYE" || method=="REFER") ) { if(t_newtran()){ t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE" || method=="REFER"){ log("**************** vm start - begin ******************\n"); } if(method=="BYE"){ log("**************** vm end/refer - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; log("**************** vm end/refer - end ********************\n"); break; }; } else { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; }; # Voicemail specific configuration - end }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; }

1 0

Re: [serusers] fifo
by michael p 25 May '05

25 May '05

only in the third link ther is my question but there is no answer Does anyone know what correct format of a packet should be pushed into fifo buffer? For example, serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx 'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx' 'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . . The above command I've tried,but got errors. and i tried the format into the serctl too thanks for replying >From: Ladislav Andel <ladia6(a)centrum.cz> >Reply-To: Ladislav Andel <ladia6(a)centrum.cz> >To: "michael p" <mikep3000(a)hotmail.com> >Subject: Re: [serusers] fifo >Date: Wed, 25 May 2005 15:39:07 +0200 > >I'm sorry, I think I've seen it in the list, but can't find it.. > >try the links below > >http://lists.iptel.org/pipermail/serusers/2004-August/010850.html >http://lists.iptel.org/pipermail/serusers/2004-August/010825.html >http://lists.iptel.org/pipermail/serusers/2005-April/018897.html > >mp> thanks but it didn't help me cause this topic not found in google in >this >mp> name maybe another one > >mp> so if you know which line i have to write it will be more simple > >mp> thanks > > > > >>From: Ladislav Andel <ladia6(a)centrum.cz> > >>Reply-To: Ladislav Andel <ladia6(a)centrum.cz> > >>To: "michael p" <mikep3000(a)hotmail.com> > >>CC: serusers(a)lists.iptel.org > >>Subject: Re: [serusers] fifo > >>Date: Wed, 25 May 2005 14:55:45 +0200 > >> > >>This topic was in the list recently. Try to google it out. > >>Lada > >> > >>mp> i already saw the different options but i want to know how to use >them > >>to > >>mp> send a message like INVITE or OPTIONS i don't know how i have to >write > >>the > >>mp> commande > >> > >>mp> thanks > >> > >> >>From: Llanos Serna García-Conde <llanosserna(a)hotmail.com> > >> >>To: mikep3000(a)hotmail.com > >> >>CC: serusers(a)lists.iptel.org > >> >>Subject: RE: [serusers] fifo > >> >>Date: Wed, 25 May 2005 14:20:27 +0200 > >> >> > >> > >> > > >

2 2

[Serusers] Failure Route only on certain response codes
by Martin Koenig 25 May '05

25 May '05

Hello all, is it possible to limit the scale of t_on_failure / failure_route[x] to certain SIP response codes? In the concrete example, I would like to exclude certain Response Codes (mainly 486) from trying at a second location. With best regards, Martin

2 1

[Serusers] Re: [Serdev] Bug? Inconsistent return of loose_route function
by Alex Mack 25 May '05

25 May '05

Hi! I've tested your changes, Cesc, but it didn't solve the issue with BYE not getting through to the TLS'ed snom phone, my setup: (84040) (84033) Snom190 ------[NAT]------ SER ------[NAT]------ Grandstream |----------TLS--------| |--------UDP-----------| Problem still is: the BYE from GS to Snom never reaches the Snom. I think it's related to connection reuse, or the lack of it. How should this work in real life? Why are there 2 Record-Routes? What does the "r2" parameter do? Anyway, your patch did some changes, because loose_route DID return FALSE for the BYE when it would have to be routed from UDP to TLS. This has been fixed by applying your patch to modules/rr/loose.c. Now it returns TRUE. Here's the debug output for the lost BYE BEFORE the patch: 9(29936) SIP Request: 9(29936) method: <BYE> 9(29936) uri: <sip:84040@*SNOM_PRIVIP*:2069;line=mq1lx8ld;transport=tls> 9(29936) version: <SIP/2.0> 9(29936) parse_headers: flags=1 9(29936) Found param type 232, <branch> = <z9hG4bK2010b4141608130c>; state=16 9(29936) end of header reached, state=5 9(29936) parse_headers: Via found, flags=1 9(29936) parse_headers: this is the first via 9(29936) After parse_msg... 9(29936) preparing to run routing scripts... 9(29936) XLOG: xl_print_log: final buffer length 110 9(29936) time [Wed May 25 14:30:59 2005] method <BYE> r-uri <sip:84040@*SNOM_PRIVIP*:2069;line=mq1lx8ld;transport=tls> 9(29936) parse_headers: flags=128 9(29936) DEBUG: add_param: tag=vff0lxwqz8 9(29936) end of header reached, state=29 9(29936) DEBUG: get_hdr_field: <To> [57]; uri=[sip:84040@mydomain.org] 9(29936) DEBUG: to body ["LCN VoIP Tel 1" <sip:84040@mydomain.org>] 9(29936) get_hdr_field: cseq <CSeq>: <10746> <BYE> 9(29936) DEBUG:maxfwd:is_maxfwd_present: value = 70 9(29936) check_via_address(*NAT_PUBIP*, *GS_PRIVIP*, 0) 9(29936) parse_headers: flags=64 9(29936) DEBUG: get_hdr_body : content_length=0 9(29936) found end of header 9(29936) ------> Request's source is NAT'ed! 9(29936) DEBUG: add_param: tag=cc5e1b5aad9ba0fc 9(29936) end of header reached, state=29 9(29936) parse_headers: flags=256 9(29936) is_preloaded: No 9(29936) grep_sock_info - checking if host==us: 13==14 && [*SNOM_PRIVIP*] == [*SER_PUBIP*] 9(29936) grep_sock_info - checking if port 5060 matches port 2069 9(29936) grep_sock_info - checking if host==us: 13==14 && [*SNOM_PRIVIP*] == [*SER_PUBIP*] 9(29936) grep_sock_info - checking if port 5060 matches port 2069 9(29936) grep_sock_info - checking if host==us: 13==14 && [*SNOM_PRIVIP*] == [*SER_PUBIP*] 9(29936) grep_sock_info - checking if port 5061 matches port 2069 9(29936) check_self: host != me 9(29936) grep_sock_info - checking if host==us: 14==14 && [*SER_PUBIP*] == [*SER_PUBIP*] 9(29936) grep_sock_info - checking if port 5060 matches port 5060 9(29936) after_loose: Topmost route URI: 'sip:*SER_PUBIP*;r2=on;ftag=vff0lxwqz8;lr' is me 9(29936) parse_headers: flags=256 9(29936) found end of header 9(29936) find_next_route: No next Route HF found 9(29936) after_loose: No next URI found 9(29936) XLOG: xl_print_log: final buffer length 43 9(29936) ------- NO Loose Routing for BYE detected! 9(29936) ----> Checking for outbound request... 9(29936) grep_sock_info - checking if host==us: 13==14 && [*SNOM_PRIVIP*] == [*SER_PUBIP*] 9(29936) grep_sock_info - checking if port 5060 matches port 2069 9(29936) grep_sock_info - checking if host==us: 13==14 && [*SNOM_PRIVIP*] == [*SER_PUBIP*] 9(29936) grep_sock_info - checking if port 5060 matches port 2069 9(29936) grep_sock_info - checking if host==us: 13==14 && [*SNOM_PRIVIP*] == [*SER_PUBIP*] 9(29936) grep_sock_info - checking if port 5061 matches port 2069 9(29936) check_self: host != me 9(29936) parse_headers: flags=-1 9(29936) XLOG: xl_print_log: final buffer length 599 9(29936) Message: BYE sip:84040@*SNOM_PRIVIP*:2069;line=mq1lx8ld;transport=tls SIP/2.0 Via: SIP/2.0/UDP *GS_PRIVIP*;branch=z9hG4bK2010b4141608130c Route: <sip:*SER_PUBIP*;r2=on;ftag=vff0lxwqz8;lr> Route: <sip:*SER_PUBIP*:5061;transport=tls;r2=on;ftag=vff0lxwqz8;lr> From: <sip:84033@mydomain.org;user=phone>;tag=cc5e1b5aad9ba0fc To: "LCN VoIP Tel 1" <sip:84040@mydomain.org>;tag=vff0lxwqz8 Call-ID: 3c267025927c-qdq7eg6tg7wp@snom190 CSeq: 10746 BYE User-Agent: Grandstream BT100 1.0.5.18 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 9(29936) XLOG: xl_print_log: final buffer length 18 9(29936) Forcing RTP Relay 9(29936) ERROR: force_rtp_proxy2: support for RTP proxy is disabled 9(29936) DEBUG: t_newtran: msg id=2 , global msg id=0 , T on entrance=0xffffffff 9(29936) parse_headers: flags=-1 9(29936) parse_headers: flags=60 9(29936) t_lookup_request: start searching: hash=65420, isACK=0 9(29936) DEBUG: RFC3261 transaction matching failed 9(29936) DEBUG: t_lookup_request: no transaction found 9(29936) DEBUG: mk_proxy: doing DNS lookup... 9(29936) parse_headers: flags=2048 9(29936) check_via_address(*NAT_PUBIP*, *GS_PRIVIP*, 0) 9(29936) tcp_send: no open tcp connection found, opening new one This is output AFTER changing loose.c: 9(30810) SIP Request: 9(30810) method: <BYE> 9(30810) uri: <sip:84040@*SNOM_PRIVIP*:2072;line=mq1lx8ld;transport=tls> 9(30810) version: <SIP/2.0> 9(30810) parse_headers: flags=1 9(30810) Found param type 232, <branch> = <z9hG4bKa9c44e2971bc9cb0>; state=16 9(30810) end of header reached, state=5 9(30810) parse_headers: Via found, flags=1 9(30810) parse_headers: this is the first via 9(30810) After parse_msg... 9(30810) preparing to run routing scripts... 9(30810) XLOG: xl_print_log: final buffer length 110 9(30810) time [Wed May 25 14:43:47 2005] method <BYE> r-uri <sip:84040@*SNOM_PRIVIP*:2072;line=mq1lx8ld;transport=tls> 9(30810) parse_headers: flags=128 9(30810) DEBUG: add_param: tag=vvtpi0yrd1 9(30810) end of header reached, state=29 9(30810) DEBUG: get_hdr_field: <To> [57]; uri=[sip:84040@mydomain.org] 9(30810) DEBUG: to body ["LCN VoIP Tel 1" <sip:84040@mydomain.org>] 9(30810) get_hdr_field: cseq <CSeq>: <48894> <BYE> 9(30810) DEBUG:maxfwd:is_maxfwd_present: value = 70 9(30810) check_via_address(*NAT_PUBIP*, *GS_PRIVIP*, 0) 9(30810) parse_headers: flags=64 9(30810) DEBUG: get_hdr_body : content_length=0 9(30810) found end of header 9(30810) ------> Request's source is NAT'ed! 9(30810) DEBUG: add_param: tag=265f0cd77557dd43 9(30810) end of header reached, state=29 9(30810) parse_headers: flags=256 9(30810) is_preloaded: No 9(30810) grep_sock_info - checking if host==us: 13==14 && [*SNOM_PRIVIP*] == [*SER_PUBIP*] 9(30810) grep_sock_info - checking if port 5060 matches port 2072 9(30810) grep_sock_info - checking if host==us: 13==14 && [*SNOM_PRIVIP*] == [*SER_PUBIP*] 9(30810) grep_sock_info - checking if port 5060 matches port 2072 9(30810) grep_sock_info - checking if host==us: 13==14 && [*SNOM_PRIVIP*] == [*SER_PUBIP*] 9(30810) grep_sock_info - checking if port 5061 matches port 2072 9(30810) check_self: host != me 9(30810) grep_sock_info - checking if host==us: 14==14 && [*SER_PUBIP*] == [*SER_PUBIP*] 9(30810) grep_sock_info - checking if port 5060 matches port 5060 9(30810) after_loose: Topmost route URI: 'sip:*SER_PUBIP*;r2=on;ftag=vvtpi0yrd1;lr' is me 9(30810) parse_headers: flags=256 9(30810) found end of header 9(30810) find_next_route: No next Route HF found 9(30810) after_loose: No next URI found 9(30810) parse_headers: flags=-1 9(30810) XLOG: xl_print_log: final buffer length 40 9(30810) +++++++ Loose Routing for BYE detected! 9(30810) XLOG: xl_print_log: final buffer length 599 9(30810) Message: BYE sip:84040@*SNOM_PRIVIP*:2072;line=mq1lx8ld;transport=tls SIP/2.0 Via: SIP/2.0/UDP *GS_PRIVIP*;branch=z9hG4bKa9c44e2971bc9cb0 Route: <sip:*SER_PUBIP*;r2=on;ftag=vvtpi0yrd1;lr> Route: <sip:*SER_PUBIP*:5061;transport=tls;r2=on;ftag=vvtpi0yrd1;lr> From: <sip:84033@mydomain.org;user=phone>;tag=265f0cd77557dd43 To: "LCN VoIP Tel 1" <sip:84040@mydomain.org>;tag=vvtpi0yrd1 Call-ID: 3c26701b186a-918969x4677v@snom190 CSeq: 48894 BYE User-Agent: Grandstream BT100 1.0.5.18 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 9(30810) XLOG: xl_print_log: final buffer length 18 9(30810) Forcing RTP Relay 9(30810) ERROR: force_rtp_proxy2: support for RTP proxy is disabled 9(30810) DEBUG: t_newtran: msg id=2 , global msg id=0 , T on entrance=0xffffffff 9(30810) parse_headers: flags=-1 9(30810) parse_headers: flags=60 9(30810) t_lookup_request: start searching: hash=59149, isACK=0 9(30810) DEBUG: RFC3261 transaction matching failed 9(30810) DEBUG: t_lookup_request: no transaction found 9(30810) DEBUG: mk_proxy: doing DNS lookup... 9(30810) parse_headers: flags=2048 9(30810) check_via_address(*NAT_PUBIP*, *GS_PRIVIP*, 0) 9(30810) tcp_send: no open tcp connection found, opening new one I'd like to have that issue resolved, but I don't get where to start. There're some options like force_tcp_alias() but the documentation of those funtcions in the NEWS file is poor at best. Is there more information somewhere? Alex Mack Cesc Santasusana schrieb: >Hi, > >I found what i think is a bug in the return value of loose_route() ... >bear with me because it is a rather long email. > >First, the scenario: > >phone1 ----- SER1 ----- SER2 ----- phone2 > >p1 to s1 and p2 to s2 use UDP >s1 to s2 uses tcp or tls > >The bug comes into place when there is a change of protocol in ser, >from UDP into TCP or TLS. >We do not use a preloaded route set from the phones. >What ser does is it adds 2 route headers, using the r2 parameter. >Route: <route1>, <route2> > >CASE A: -------------------------------------------- >In this scenario, say p1 places a call to p2, going through s1 and s2, >creating the route set with the record-route. >Now, let's say p2 generates the BYE message. >When the BYE gets to s1, the last hop, in the config we do: >if( loose_route () ) { > t_relay(); >} >But, and here is the inconsistency, loose_route (remember we are using >TCP/TLS, so there are 2 routes in one header field) returns FALSE. > >CASE B: -------------------------------------------- >If this experiment is repeated using ALL UDP, the route headers when they >get to s1 contain only one route, and the call to loose_route() returns .... TRUE! > > >Unless the config file does not allow blind sip message relaying, both cases end >up working ... the BYE gets to the final destination. But if the config file does not >accept relaying ... the request-uri in the BYE is not taken as a local domain ... and >thus the message is rejected ... (To Alex Mack: could this be the solution to your problem? >the BYEs not reaching the phone? add some log output and check it ... ) > >================================================================ >So, this is not consistent. Now, what is the solution? >I think that both case A and B should return the same, but what, true or false? > >As I see it, the solution causing less trouble is that both return true. So, the last proxy >considers this BYE as being loose_routed. >For this ... change the code in: >rr::loose_route.c:: static inline int after_loose(struct sip_msg* _m, int preloaded) > ..... > if (enable_double_rr && is_2rr(&puri.params)) { > .... > if (res > 0) { /* No next route found */ > DBG("after_loose: No next URI found\n"); > return (preloaded ? NOT_RR_DRIVEN : RR_DRIVEN); > or simply > return (preloaded RR_DRIVEN); > > >And now ... the other option is that both return false ... but this would make big changes. >Correct me if i am wrong, but wouldn't this be the compliant solution? >Considering that the r-uri is not part of the route set, when the last proxy processes >all route headers and there are no more left, there is no loose routing done. The >destination is taken from the r-uri, not the route set ... thus loose_route should return >false. >Or I am getting it all wrong? rfc and goal of loose_route() ? > > >Regards, > >Cesc > > > >Unclassified > > > > > > > > > > > >_______________________________________________ >Serdev mailing list >serdev(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serdev > > >

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Re: [Serusers] Failure Route only on certain response codes
by Martin Koenig 25 May '05

25 May '05

Hello again, to be a little bit more general. Is it possible to limit failure_route to cases only where the initial t_relay() timed out and no reply of any kind is received? Regards, Martin Martin Koenig wrote: > Hello all, > > is it possible to limit the scale of t_on_failure / failure_route[x] > to certain SIP response codes? > > In the concrete example, I would like to exclude certain Response > Codes (mainly 486) from trying at a second location. > > With best regards, > Martin > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[serusers] fifo
by michael p 25 May '05

25 May '05

hi someone can explain me how use in serctl the fifo command please? serctl fifo ...? for example to send an INVITE or OPTIONS thanks M.

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[Serusers] pstn simple question
by Wojciech Ziniewicz 25 May '05

25 May '05

Hi, Anybody know how to account only the invites and byes that go from the endpoint? For radius accounting i get one BYE and two invites and that is not how it should be. my radius flag is "2" and my accouting code snippet is like that: #we log every invite and corresponding bye to radius accounting if (method=="INVITE" | method=="BYE") { setflag(2); } that gives me two starts and one stop and finally a duplicated radius entry. any help would be appreciated.. please..

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[Serusers] ser dedicated instance for sems
by Llanos Serna García-Conde 25 May '05

25 May '05

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[Serusers] No ringback-tone problem (SER->PSTN)
by Kostas Marneris 25 May '05

25 May '05

Hello, I have the "no ringback tone" problem when I call from a SIP softphone (Linphone) to the PSTN (through a cisco 5300 with PRI connection to telco). Linphone --> SER --> c5300 --> POTS My conf on c5300:- ! dial-peer voice 1001 pots destination-pattern 210....... no digit-strip port 0:D ! I used also the following commands on pots dial-peer with no success. progress_ind setup enable 3 progress_ind connect enable 8 any ideas ? thank you, Kostas

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[Serusers] what next?
by Bharat More 25 May '05

25 May '05

Hi...List... I has done the installation of ser 0.8.14.,Raduus and mysql. Boss said that I can make the calls. two clients are added to redius. Can anyone give me command line example to start ser, RADIUS, and mysql, and also a full command line parameter to make a call... Please Help me I am a new guy to SER ... ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

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[Serusers] CDR Issue
by Bharat More 25 May '05

25 May '05

Hi List... I have to know that how to deal with the issue of CDR in RADIUS can anyone give me details? ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

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[Serusers] ONsip.org downloads are now fixed
by Greger V. Teigre 25 May '05

25 May '05

Many people have had problems with ONsip.org downloads the last couple of days. We had an ftp server down, as well as a domain problem that caused the server to return a "leeching not allowed" error. We apologize for the inconvenience. The downloads should now be available. g-)

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[Serusers] BYE messages from Cisco
by Helge Waastad 25 May '05

25 May '05

Hi, In the following scenario, the Cisco will not generate a BYE message from Cisco VOIP ----> SER ---->CISCO--->PSTN VOIP ---- SER -----CISCO<-- HANG-UP In this scenario, the HANG-UP from PSTN generates a ev_disc_prog on the cisco. I have modified the sesssion tcl script to catch the ev_disc_prog to initiate a call tear-down and then it works the way I want...the cisco generates a BYE message. However, does anyone know if this diverts from any standards? -- mvh/best regards Helge Waastad

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[Serusers] Error with SIP/2.0 404 Not Found.
by Antonio Membrives Nieto 25 May '05

25 May '05

Hi to all, i have installed a Ser 8.12 in Fedora Core 3 and i have a problem when i try call to another user. I received the error 404 but if i use a softphone client in server console, anyone user can talk with me. This is my ser.cfg -> [root@dns2 ser]# more ser.cfg # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=yes # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=yes # (cmd. line: -R) port=5060 children=4 listen=82.159.201.35 alias=telintcable.es fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) #if (uri==myself) { if (uri=~"telintcable.es") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("telintcable.es", "subscriber")) { www_challenge("telintcable.es", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } And trace with ngrep port 5060 in server -> U 82.159.201.35:5060 -> 192.168.110.135:5060 INVITE sip:prueba@192.168.110.135:5060 SIP/2.0..Record-Route: <sip:prueba@82.159.201.35;ftag=4 2799873930355;lr=on>..Content-Length: 339..Contact: <sip:amembrives@192.168.110.5:5060>..Call- ID: 409094EA-AB6D-4329-A43C-8308E14397E6@192.168.110.5..Content-Type: application/sdp..From: " amembrives"<sip:amembrives@telintcable.es>;tag=42799873930355..CSeq: 1 INVITE..Max-Forwards: 6 9..To: <sip:prueba@telintcable.es>..Via: SIP/2.0/UDP 82.159.201.35;branch=z9hG4bK36e9.94679231 .0..Via: SIP/2.0/UDP 192.168.110.5;received=217.125.143.150;rport=5000;branch=z9hG4bKc0a86e050 131c9b1429327c70000188e00000007..User-Agent: SJphone/1.50.271d (SJ Labs)....v=0..o=- 332592903 1 3325929031 IN IP4 192.168.110.5..s=SJphone..c=IN IP4 192.168.110.5..t=0 0..a=direction:activ e..m=audio 49152 RTP/AVP 3 97 98 8 0 101..a=rtpmap:3 GSM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap :98 iLBC/8000..a=fmtp:98 mode=20..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 tel ephone-event/8000..a=fmtp:101 0-11,16.. # U 217.125.143.150:5060 -> 82.159.201.35:5060 OPTIONS sip:dns2.telintcable.es:5060 SIP/2.0..Content-Length: 0..Call-ID: DB40A222-9E72-4412-9 FEA-7A13F0C950C7@192.168.110.135..From: <sip:prueba@telintcable.es>;tag=1944664032666..CSeq: 1 2 OPTIONS..Max-Forwards: 70..To: <sip:dns2.telintcable.es:5060>..Via: SIP/2.0/UDP 192.168.110. 135;rport;branch=z9hG4bKc0a86e870131c9b1429327d40000290400000023.... # U 82.159.201.35:5060 -> 217.125.143.150:5060 SIP/2.0 404 Not Found..Call-ID: DB40A222-9E72-4412-9FEA-7A13F0C950C7@192.168.110.135..From: <s ip:prueba@telintcable.es>;tag=1944664032666..CSeq: 12 OPTIONS..To: <sip:dns2.telintcable.es:50 60>;tag=9d871af07b54b916b825cc89b774548a.916e..Via: SIP/2.0/UDP 192.168.110.135;rport=5060;bra nch=z9hG4bKc0a86e870131c9b1429327d40000290400000023;received=217.125.143.150 ..Server: Sip EXpr ess router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 82.159.201.35:5060 "Noisy fe edback tells: pid=3210 req_src_ip=217.125.143.150 req_src_port=5060 in_uri=sip:dns2.telintcab le.es:5060 out_uri=sip:dns2.telintcable.es:5060 via_cnt==1".... Sorry for my bad english and thank for help.

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[Serusers] (no subject)
by lzf＠alternet-europe.com 25 May '05

25 May '05

Hi, I would like that SER check the uri when it receives a REGISTER message (and not only the username). Does anyone an idea for this? Ludovic.

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[Serusers] Multiple SERs connection to Single DB
by TCCHAN 25 May '05

25 May '05

Hi All: I tried to server farm the SER and have two simple questions. (1)can multiple SERs connect to single copy of MySql database at the same time. I tried modparam("auth_db", "db_url", "sql://ser:heslo@192.228.118.240/ser") modparam("usrloc", "db_url", "sql://ser:heslo@192.228.118.240/ser") modparam("group", "db_url", "sql://ser:heslo@192.228.118.240/ser") on two machines with ser, the results is that the first one is Ok but the second one has a error message to many db connection opened Did I misses anything? (2) what is the recommended way of server farming SER and how to load balance the traffic. Thanks very much for any help available. Regards, TC Chan

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[Serusers] Authentication Based On IP Addresses.
by Bharat More 24 May '05

24 May '05

Hi All... I am using SER 0.8.14 ,RADIUS,and mysql. I want to authenticate all the clients based on IP addresses. How to proceed...? where to make the changes? can any one give me the sample code function?... Thanks for any hint...Bharat ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

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Re: [SPAM] [Serusers] Re: Module programming question
by Daniel Corbe 24 May '05

24 May '05

I get the same error message when I move everything into the .c file. -Daniel On 5/24/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > I did not make a complete review but the first thing I saw is that you should move all the module interface to the .c file. > > Samuel. > > > Unclassified. > >>> Daniel Corbe <daniel.junkmail(a)gmail.com> 05/23/05 05:34PM >>> > Actually those were incredibly useful. Thank you. > > Now, when I compile my module I am getting the following error message: > > 16:18:13 root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc > -c -o radius_cache.o radius_cache.c > In file included from radius_cache.c:46: > radius_cache.h:58: `NAME' undeclared here (not in a function) > radius_cache.h:58: syntax error before string constant > radius_cache.h:62: warning: initialization from incompatible pointer type > > I'm not sure what to do about that. It seems to be something internal > and specific to MODULE_VERSION in the SER header. > > My module: > http://www.rafb.net/paste/results/N0uQyJ91.html > > My module's header file: > http://www.rafb.net/paste/results/vo3vG345.html > > Thanks for the help. > > -Daniel > > > On 5/23/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > There's a simple module, called print that is meant to be mainly an > > example. So I recommend you to take a look at its interface and to start > > from there your module. > > I attacht also several files that a little bit old but still they are > > really useful. > > > > Luck! > > > > Samuel. > > > > > > Unclassified. > > >>> Daniel Corbe <daniel.junkmail(a)gmail.com> 05/22/05 10:41PM >>> > > Here's another question: > > > > I got my module to compile by taking _init out and reworking the > > initalization of the module into something else. > > > > Now when SER loads I get: > > > > May 22 21:34:00 proxy2 ser: ERROR: no version info in module > > </usr/local/lib/ser/modules/radius_cache.so>: Undefined symbol > > "module_version" > > > > What is module_version? How do I define it? > > > > Thanks. > > > > -Daniel > > > > > > On 5/22/05, Daniel Corbe <daniel.junkmail(a)gmail.com> wrote: > > > Hello, > > > > > > I have a SER module which I've written in C and I'm trying to > > compile. > > > When I compile it I get this error: > > > > > > 20:53:24 > > root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc > > > -shared -o radius_cache.so radius_cache.o > > > radius_cache.o: In function `_init': > > > radius_cache.o(.text+0x0): multiple definition of `_init' > > > /usr/lib/crti.o(.init+0x0): first defined here > > > > > > Is there something else in SER that exports _init already? > > > > > > Thanks. > > > > > > -Daniel > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > >

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Re: [Serusers] UAC module (backport to 0.9.0)
by Michael Ulitskiy 24 May '05

24 May '05

It's in this thread in my post from 05/16. Also, yesterday, I posted it to serdev. Michael On Tuesday 24 May 2005 09:00 am, you wrote: > Would u please send me link for that patch? > > Thanks, > Mohammad > > > Michael Ulitskiy wrote: > > >It works with above-posted patch and with limitations stated in the docs > >as Daniel pointed out. > >You won't be able to authenticate to RFC-compliant implementations. > > > >Michael > > > >On Monday 23 May 2005 12:05 pm, info(a)beeplove.com wrote: > > > > > >>Does this module work on ser-0.9.0? > >>Or, was you guys able to make it work on ser-0.9.0 > >> > >>If you were able to make it worked, would you please share the tricks, > >> > >>Thanks, > >>Mohammad > >> > >> > >>Original Message: > >>----------------- > >>From: Michael Ulitskiy mdu113(a)acedsl.com > >>Date: Mon, 23 May 2005 12:01:40 -0400 > >>To: daniel(a)voice-system.ro, serusers(a)lists.iptel.org > >>Subject: Re: [Serusers] UAC module (backport to 0.9.0) > >> > >> > >>On Sunday 22 May 2005 04:51 am, Daniel-Constantin Mierla wrote: > >> > >> > >>>Hello, > >>> > >>>the issue with CSeq is known and it is stated in the documentation (see > >>>the note there): > >>> > >>>http://www.voice-system.ro/docs/uac/ar01s04.html#uac_auth > >>> > >>> > >>Thanks. Actually I read that docs, but somehow overlooked that note. > >>Thanks for pointing it out. > >> > >> > >> > >>>Daniel > >>> > >>> > >>Michael > >> > >> > >> > >>>On 05/16/05 22:05, Michael Ulitskiy wrote: > >>> > >>> > >>> > >>>>Hello, > >>>> > >>>>I succeeded to make UAC module calculate credentials and > >>>>resend message with the following simple patch on t_reply.c > >>>>of tm module: > >>>> > >>>>--- t_reply.c.bak 2005-05-13 14:57:25.000000000 -0400 > >>>>+++ t_reply.c 2005-05-13 15:16:45.000000000 -0400 > >>>>@@ -761,6 +761,10 @@ > >>>> a callback; save branch count to be able to determine > >>>> later if new branches were initiated */ > >>>> branch_cnt=Trans->nr_of_outgoings; > >>>>+ /* also append the current reply to the transaction to > >>>>+ * make it available in failure routes - a kind of > >>>> > >>>> > >>"fake" > >> > >> > >>>>+ * save of the final reply per branch */ > >>>>+ Trans->uac[branch].reply = reply; > >>>> > >>>> /* run ON_FAILURE handlers ( route and callbacks) */ > >>>> if ( has_tran_tmcbs( Trans, > >>>> > >>>> > >>TMCB_ON_FAILURE_RO|TMCB_ON_FAILURE) > >> > >> > >>>>@@ -769,6 +773,11 @@ > >>>> > >>>> > >>>> > >>picked_branch==branch?reply:Trans->uac[picked_branch].reply, > >> > >> > >>>> picked_code); > >>>> } > >>>>+ > >>>>+ /* now reset it; after the failure logic, the reply may > >>>>+ * not be stored any more and we don't want to keep into > >>>>+ * transaction some broken reference */ > >>>>+ Trans->uac[branch].reply = 0; > >>>> > >>>> /* look if the callback perhaps replied transaction; it > >>>> > >>>> > >>also > >> > >> > >>>> covers the case in which a transaction is replied > >>>> > >>>> > >>localy > >> > >> > >>>>It's actually taken from cvs-head. > >>>>Now ser.cfg written below works as expected with just one > >>>>important issue. Authentication doesn't work anyway because > >>>>ser doesn't increase CSeq number when resending message > >>>>(INVITE) with credentials. I'm trying to authenticate > >>>>to asterisk and in case of asterisk it immediately replies > >>>>"503 Service Unavailable" if it receives INVITE with the same CSeq. > >>>>I thought it could be asterisk-specific problem, but RFC3261 in > >>>>section "8.1.3.5 Processing 4xx Responses" says: > >>>>"new request constitutes a new transaction and SHOULD have the same > >>>>value of the Call-ID, To, and From of the previous request, but the CSeq > >>>>should contain a new sequence number that is one higher than the > >>>> > >>>> > >>previous." > >> > >> > >>>>So I guess this is not an asterisk-specific. > >>>>It is my understanding that in order for this to work ser must increment > >>>>CSeq when authenticating to UAS, but decrement CSeq in all subsequent > >>>>in-dialog messages that will be sent to call originator including BYEs. > >>>>This, in turn, requires for ser to be call-statefull which is not the > >>>> > >>>> > >>case. > >> > >> > >>>>Conclusion: uac authentication in ser is not possible. > >>>>Please correct me if I'm wrong (I honestly want to be wrong on this > >>>> > >>>> > >>matter:)) > >> > >> > >>>>Thank you, > >>>> > >>>>Michael > >>>> > >>>>On Sunday 15 May 2005 12:36 am, you wrote: > >>>> > >>>> > >>>> > >>>> > >>>>>Michael Ulitskiy wrote: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>Hello, > >>>>>> > >>>>>>Has anyone succeeded in getting UAC authentication to work? > >>>>>>I'm doing the following: > >>>>>> > >>>>>>route { > >>>>>> resetflag(1); > >>>>>> t_on_failure("1"); > >>>>>> route(1); > >>>>>>} > >>>>>> > >>>>>>route[1] > >>>>>>{ > >>>>>> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") > >>>>>> > >>>>>> > >>{ > >> > >> > >>>>>> sl_send_reply("479", "We don't forward to private IP > >>>>>> > >>>>>> > >>addresses"); > >> > >> > >>>>>> break; > >>>>>> }; > >>>>>> if (!t_relay()) { > >>>>>> sl_reply_error(); > >>>>>> }; > >>>>>>} > >>>>>> > >>>>>>failure_route[1] { > >>>>>> # authentication reply received? > >>>>>> if ( t_check_status("401|407") ) { > >>>>>> if (!isflagset(1) && uac_auth()) { > >>>>>> setflag(1); > >>>>>> t_on_failure("1"); > >>>>>> append_branch(); > >>>>>> route(1); > >>>>>> } else { > >>>>>> t_reply("500","Error occurred"); > >>>>>> } > >>>>>> break; > >>>>>> } > >>>>>> > >>>>>>} > >>>>>> > >>>>>>When uac_auth() is called I get the following in the: > >>>>>>0(28973) DEBUG:uac:uac_auth: picked reply is (nil), code 407 > >>>>>>0(28973) BUG:uac:uac_auth: empty reply on picked branch > >>>>>> > >>>>>>Any suggestions or ideas? > >>>>>>Thank you, > >>>>>> > >>>>>>Michael > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>I am also getting same thing with different msg .. > >>>>>BUG:uac:uac_auth: empty reply on picked branch > >>>>> > >>>>>Anybody know how to overcome this? > >>>>> > >>>>>Mohammad > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>_______________________________________________ > >>>>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 > >> > >>-------------------------------------------------------------------- > >>mail2web - Check your email from the web at > >>http://mail2web.com/ . > >> > >> > >> > >> > >> > > > > > > > >

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[Serusers] Advice needed
by Michael Ulitskiy 24 May '05

24 May '05

Hello, I'd like ask for advice on what is in your opinion the best solution in the following scenario. I have a bunch of sip servers (asterisk boxes as my users need pbx functionality) that can make sip call to each other and my PSTN gateway. Now I want to purchase PSTN terminitaion in several different markets (and probably more in the future). All those terminations will require authentication. I want all my boxes when they see non-local call to send it to a central routing server that would determine where this call should be sent and authenticate to the appropriate provider so that I don't have to configure all credentials on all asterisk boxes. Also I want it not to deal with the media at all. All media streams should go directly from asterisk box to the PSTN termination provider. So basically it should be central SIP router that is able to authenticate calls if neccessary. I thought I could do it with SER and its UAC module, but it appears UAC module doesn't work and probably won't work (see my previous post in this list about UAC backport to 0.9.0). Also I don't want to use asterisk in this place as asterisk always wants to stay in media path and I'd really like to avoid of getting into hassle with re-invites. So the question is what are my options and what you would advice as a solution. Are there any software out there that can do it (preferably open-source, of course) or what else you could suggest to do to get desired results. Thanks a lot, -- See you later, Michael

2 6

[serusers] serctl in C
by michael p 24 May '05

24 May '05

hi is there someone that know how can i write in C this part of program that i wrote in shell in serctl (in the ping part of the prog when i do the command "serctl ping URI") : ping) # error handling is hacked -- filter_fl should not # consume positive status -- that should be done by # calling app if [ "$#" -ne 2 ] ; then usage exit 1 fi TIMESTAMPS=`date "+%s.%N"` print_status $TIMESTAMPS RET=`fifo_cmd t_uac_dlg OPTIONS "$2" "." \ "From: sip:daemon@$SIP_DOMAIN" \ "To: <$2>" "Contact: <sip:daemon@!!>" "." "." \ | head -1 ` TIMESTAMPE=`date "+%s.%N"` print_status $TIMESTAMPE TIME= echo "(${TIMESTAMPE} - ${TIMESTAMPS}) *1000" | bc -l print_status $TIME ;; thanks a lot ps: i need the part of the fifo_cmd etc..., not the part of timestamp (but if you know if there is the same date function in C please tell me ;) )

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Re: [Serusers] error in compile ser-0.9.0
by Kamran Ahmad 24 May '05

24 May '05

hello can you any one tell me what is problem in this i am using ser 0.9.0 when ever i am trying to install ser. this is the problem in this installation Makefile.rules:81: action.d: No such file or directory Makefile.rules:81: crc.d: No such file or directory Makefile.rules:81: daemonize.d: No such file or directory Makefile.rules:81: data_lump.d: No such file or directory Makefile.rules:81: data_lump_rpl.d: No such file or directory Makefile.rules:81: dprint.d: No such file or directory Makefile.rules:81: dset.d: No such file or directory Makefile.rules:81: error.d: No such file or directory Makefile.rules:81: fifo_server.d: No such file or directory Makefile.rules:81: flags.d: No such file or directory Makefile.rules:81: forward.d: No such file or directory Makefile.rules:81: hash_func.d: No such file or directory Makefile.rules:81: ip_addr.d: No such file or directory Makefile.rules:81: main.d: No such file or directory Makefile.rules:81: md5.d: No such file or directory Makefile.rules:81: md5utils.d: No such file or directory Makefile.rules:81: modparam.d: No such file or directory Makefile.rules:81: msg_translator.d: No such file or directory Makefile.rules:81: pass_fd.d: No such file or directory Makefile.rules:81: proxy.d: No such file or directory Makefile.rules:81: qvalue.d: No such file or directory Makefile.rules:81: re.d: No such file or directory Makefile.rules:81: receive.d: No such file or directory Makefile.rules:81: resolve.d: No such file or directory Makefile.rules:81: route.d: No such file or directory Makefile.rules:81: route_struct.d: No such file or directory Makefile.rules:81: script_cb.d: No such file or directory Makefile.rules:81: socket_info.d: No such file or directory and also serctl is not working i am using root serctl moni is not working [root@voice ser]# serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config --- Grigory Fishilevich <g.fishilevich(a)gmail.com> wrote: > hi, > > these messages have nothing to say. i think you have > a permissions > problem, you should take a look at /var/log/messages > > ? have you ser running or not? > __________________________________ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail

2 1

[Serusers] Problems with Ser 0.9.0 and Mediaproxy
by Felipe Martins 24 May '05

24 May '05

Hi everyone, I trying to configure SER-0.9.0 to work with MediaProxy-1.3.1 but i'm experiencing some problems. I can install mediaproxy, configure his mediaproxy.ini configuration file and run it, but SER (with the configuration acording to the onsip.org guide) doesn't run. it gives me de following error: ==================== Error ================================= May 24 09:22:47 sipproxy1 /usr/local/sbin/ser[626]: error: mediaproxy/mod_init(): can't find is_from_local and/or is_uri_host_local functions. Check if domain.so is loaded May 24 09:22:47 sipproxy1 /usr/local/sbin/ser[626]: init_mod(): Error while initializing module mediaproxy ==================== End ================================= I've checked for mediaproxy.so and domain.so, and both are in the directory they should be. I've also looked up for some information at www.google.com.br but didn't find any relevant. Have anyone experienced this kind of problem ? For a complete post my SER.CFG is attached below: ==================== Ser.cfg ================================ debug=3 fork=yes log_stderror=no listen=99.99.99.99 port=5060 children=4 dns=no rev_dns=no 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.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.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("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # 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 (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ==================== End ================================= Best Regards. -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

2 1

[Serusers] Centrex Features
by Clay Bryan 24 May '05

24 May '05

Hello, I was wondering if the ser has any Centrex or pbx functionality. Like putting someone on hold and transferring a phone call and multiple lines. Thanks in advance. Wercs Communications Clay Bryan Network Administrator Wercs Communications 400 East First Casper, WY 82601 <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe r%2C+WY++82601&country=us> CBryan(a)wercs.com <mailto:CBryan@wercs.com> tel: fax: mobile: 307-233-8359 307-233-8701 307-258-7371 Add me to your address book... <https://www.plaxo.com/add_me?u=21475170292&v0=646557&k0=282754785> Want a signature like this? <http://www.plaxo.com/signature>

2 1

Re: [SPAM] [Serusers] Re: Module programming question
by Samuel Osorio Calvo 24 May '05

24 May '05

I did not make a complete review but the first thing I saw is that you should move all the module interface to the .c file. Samuel. Unclassified. >>> Daniel Corbe <daniel.junkmail(a)gmail.com> 05/23/05 05:34PM >>> Actually those were incredibly useful. Thank you. Now, when I compile my module I am getting the following error message: 16:18:13 root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc -c -o radius_cache.o radius_cache.c In file included from radius_cache.c:46: radius_cache.h:58: `NAME' undeclared here (not in a function) radius_cache.h:58: syntax error before string constant radius_cache.h:62: warning: initialization from incompatible pointer type I'm not sure what to do about that. It seems to be something internal and specific to MODULE_VERSION in the SER header. My module: http://www.rafb.net/paste/results/N0uQyJ91.html My module's header file: http://www.rafb.net/paste/results/vo3vG345.html Thanks for the help. -Daniel On 5/23/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > There's a simple module, called print that is meant to be mainly an > example. So I recommend you to take a look at its interface and to start > from there your module. > I attacht also several files that a little bit old but still they are > really useful. > > Luck! > > Samuel. > > > Unclassified. > >>> Daniel Corbe <daniel.junkmail(a)gmail.com> 05/22/05 10:41PM >>> > Here's another question: > > I got my module to compile by taking _init out and reworking the > initalization of the module into something else. > > Now when SER loads I get: > > May 22 21:34:00 proxy2 ser: ERROR: no version info in module > </usr/local/lib/ser/modules/radius_cache.so>: Undefined symbol > "module_version" > > What is module_version? How do I define it? > > Thanks. > > -Daniel > > > On 5/22/05, Daniel Corbe <daniel.junkmail(a)gmail.com> wrote: > > Hello, > > > > I have a SER module which I've written in C and I'm trying to > compile. > > When I compile it I get this error: > > > > 20:53:24 > root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc > > -shared -o radius_cache.so radius_cache.o > > radius_cache.o: In function `_init': > > radius_cache.o(.text+0x0): multiple definition of `_init' > > /usr/lib/crti.o(.init+0x0): first defined here > > > > Is there something else in SER that exports _init already? > > > > Thanks. > > > > -Daniel > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Serusers] How to realized the jabber server and ser server integration?
by zhaomin 24 May '05

24 May '05

My ser server working is good,I want to integrated the xmpp jabber server into ser service.and when I reading the doc about SIMPLE to XMMP(Jabber) gateway doc I find some confure error in it,when I correct the ser configre error .it's runing.But I can not register into ser servers .How can I resolve it ? Thanks very much ZhaoMin following is my ser.cfg,jabber server is good runing. # # configuration for Jabber module testing # (sample config file using the module with presence support) # # $ID: daniel $ # debug=9 # debug level (cmd line: -dddddddddd) #fork=yes # (cmd. line: -D) fork=no log_stderror=yes # (cmd line: -E) #log_stderror=no # (cmd line: -E) children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 listen=127.0.0.1 listen=221.202.129.165 fifo="/tmp/ser_fifo" # for more info: sip_router -h # ------------------ module loading ---------------------------------- #modules loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/jabber.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/pa.so" # ----------------- setting module-specific parameters --------------- # -- registrar params -- modparam("registrar", "default_expires", 120) modparam("registrar", "use_domain", 1) # -- usrloc params -- modparam("usrloc", "use_domain", 1) modparam("usrloc", "db_mode", 0) # -- jabber params -- modparam("jabber","db_url","sql://root:zm9898@127.0.0.1/sip_jab") modparam("jabber","jaddress","mvu.cn") modparam("jabber","jport",5222) modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","mvu.cn=*") modparam("jabber","aliases","4;aim.mvu.cn;icq.mvu.cn;msn.mvu.cn=%;yahoo.mvu.cn;") route{ #if ( !mf_process_maxfwd_header("10") ) #{ # sl_send_reply("483","To Many Hops"); # drop(); #}; if (uri=~"[@:]sip\.mvu\.cn([;:].*)*") { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { if (t_newtran()) { save("location"); log("REGISTER received -> reply okay\n"); }; if(search("registration")) { log("XJAB: Going ONline in Jabber network!!!\n"); if(jab_go_online()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; } else { log("XJAB: Going OFFline in Jabber network!!!\n"); if(jab_go_offline()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; }; break; }; if (method=="SUBSCRIBE") { if (t_newtran()) { handle_subscription("registrar"); }; break; }; if(!lookup("location")) { sl_send_reply("404","Not found"); break; }; }; if ((search("To:.*@icq\.mvu\.cn")) || (search("To:.*@\.mvu\.cn")) || (search("To:.*@msn\.mvu\.cn")) || (search("To:.*@yahoo\.mvu\.cn"))) { if (! t_newtran()) { sl_reply_error(); break; }; if (method=="MESSAGE") { log("MESSAGE received -> manage it with XJAB\n"); if(search("\n:on")) { if (jab_go_online()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not found"); }; break; }; if(search("\n:off")) { if (jab_go_offline()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not found"); }; break; }; if(search("\n:join")) { if (jab_join_jconf()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not Found"); }; break; }; if(search("\n:exit")) { if (jab_exit_jconf()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not Found"); }; break; }; if (jab_send_message()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("503","Service Unavailable"); }; break; }; if (method=="SUBSCRIBE") { handle_subscription("jabber"); break; }; log("NON_Message request received for JABBER gateway->dropt!\n"); sl_send_reply("202","Accepted"); break; }; if (!t_relay()) { sl_reply_error(); }; #forward(uri:host,uri:port); }

1 0

[Serusers] fifo_db_url
by Llanos Serna García-Conde 24 May '05

24 May '05

1 0

[Serusers] How to generate ACD in redius?
by Bharat More 24 May '05

24 May '05

Hi... I want to generate ACD (Autometic Call Destribution ) using RADIUS. how can I do that? where to manipulate the code? Or Just give me the logical structure for the authentication and RADIUS accounting. Just as ( Request...sanity....forwarding.. checks for domains...etc.) Please explain in details coz i'm going to use SER First time... ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

1 0

[Serusers] cvs help
by Mohammad Khan 24 May '05

24 May '05

Hello, Where can I get instruction to download ser from cvs? What are the tags available to download? Thanks, Mohammad Khan

2 1

Re: [Serusers] javarockx (Paul) 's ser.cfg
by Dave 24 May '05

24 May '05

Hi, In the failure_route[1], an append_branch() function is missing after avp_pushto(), which causes an error during t_relay(). Am I correct or am I missing something? Thanks, Dave --- Java Rockx <javarockx(a)gmail.com> wrote: > Sorry all for the repost but the listserv scrubbed > my attachment. Here it is > again. > > P > > # ------------- Initial global variables > > debug=3 > fork=yes > log_stderror=no > > listen=10.3.0.221 <http://10.3.0.221> > port=5060 > children=4 > server_signature=no > > # hostname matching an alias will satisfy the > condition uri==myself" > alias=sip.mycompany.com <http://sip.mycompany.com> > > # dns - Uses dns to check if it is necessary to add > a "received=" field > # to a via. Default is no. > # rev_dns - Same as dns but use reverse DNS. > dns=no > rev_dns=no > > # check_via - Turn on or off Via host checking when > forwarding replies. > # Default is no. arcane. looks for discrepancy > between name and > # ip address when forwarding replies. > check_via=no > > # syn_branch - Shall the server use stateful synonym > branches? It is > # faster but not reboot-safe. Default is yes. > syn_branch=yes > > # memlog - Debugging level for final memory > statistics report. Default > # is L_DBG -- memory statistics are dumped only if > debug is set high. > memlog=1000 > > # sip_warning - Should replies include extensive > warnings? > # By default yes, it is good for trouble-shooting. > sip_warning=no > > # fifo - FIFO special file pathname > fifo_mode=0666 > fifo="/tmp/ser_fifo" > fifo_db_url="mysql://ser@localhost/ser" > > # reply_to_via - A hint to reply modules whether > they should send reply > # to IP advertised in Via. Turned off by default, > which means that > # replies are sent to IP address from which requests > came. > reply_to_via=no > > # user | uid - uid to be used by the server. 99 = > nobody. > #uid="nobody" > > # group | gid - gid to be used by the server. 99 = > nobody. > #gid="nobody" > > # mhomed -- enable calculation of outbound > interface. > # useful on multihomed servers. > mhomed=0 > > # ------------- external 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/auth.so" > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > loadmodule "/usr/local/lib/ser/modules/acc.so" > loadmodule "/usr/local/lib/ser/modules/group.so" > loadmodule "/usr/local/lib/ser/modules/print.so" > loadmodule "/usr/local/lib/ser/modules/textops.so" > loadmodule > "/usr/local/lib/ser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > loadmodule "/usr/local/lib/ser/modules/uri.so" > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > loadmodule "/usr/local/lib/ser/modules/domain.so" > loadmodule "/usr/local/lib/ser/modules/speeddial.so" > loadmodule "/usr/local/lib/ser/modules/options.so" > loadmodule "/usr/local/lib/ser/modules/avpops.so" > loadmodule > "/usr/local/lib/ser/modules/permissions.so" > loadmodule "/usr/local/lib/ser/modules/xdz_tools.so" > > # ------------ nathelper settings > modparam("nathelper", "rtpproxy_disable", 1) > modparam("nathelper", "natping_interval", 0) > > # ------------ media proxy settings > modparam("mediaproxy", "natping_interval", 30) > modparam("mediaproxy", "mediaproxy_socket", > "/var/run/mediaproxy.sock") > modparam("mediaproxy", "sip_asymmetrics", > "/usr/local/etc/ser/sip-asymmetric-clients") > modparam("mediaproxy", "rtp_asymmetrics", > "/usr/local/etc/ser/rtp-asymmetric-clients") > > # ------------- db_url setting > modparam("acc|auth_db|domain|group|lcr|permissions|speeddial|uri_db|usrloc|xdz_tools", > "db_url", "mysql://ser@localhost/ser") > > # ------------- use_domain setting > modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", > "use_domain", 0) > > # ------------- permissions settings > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > > # ------------- accounting parameters > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 1) > modparam("acc", "log_fmt", "cdfimorstup") > modparam("acc", "log_level", 1) > modparam("acc", "failed_transactions", 1) > modparam("acc", "report_cancels", 1) > modparam("acc", "report_ack", 0) > > # allows clear text passwords in the mysql database > modparam("auth_db", "calculate_ha1", 1) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "avps_column_int", "domn") > modparam("auth_db", "avps_column_str", "uuid") > > # ------------- domain parameters > modparam("domain", "db_mode", 1) > > # ------------- registration parameters > modparam("registrar", "nat_flag", 6) > modparam("registrar", "min_expires", 300) > modparam("registrar", "max_expires", 3600) > modparam("registrar", "default_expires", 600) > modparam("registrar", "default_q", 100) > modparam("registrar", "desc_time_order", 1) > modparam("registrar", "append_branches", 0) > > # ------------- rr parameters > modparam("rr", "enable_full_lr", 0) > > # ------------- tm parameters > modparam("tm", "fr_timer", 22) > modparam("tm", "fr_inv_timer", 27) > modparam("tm", "wt_timer", 10) > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > > # ------------- usrloc parameters > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "timer_interval", 30) > modparam("usrloc", "desc_time_order", 1) > > # -------------- AVPOPS parameters > modparam("avpops", "avp_url", > "mysql://ser@localhost/ser") > modparam("avpops", "avp_table", "usr_preferences") > modparam("avpops", "avp_aliases", "voicemail=i:500") > > route { > > # > ------------------------------------------------------------------------ > # Sanity Check Section > # > ------------------------------------------------------------------------ > if (!mf_process_maxfwd_header("10")) { > log(1, "LOOP DETECTED"); > sl_send_reply("483", "Too Many Hops"); > break; > }; > > if (msg:len > max_len) { > log(1, "MESSAGE OVERFLOW"); > sl_send_reply("513", "Message Too Large"); > break; > }; > > # > ------------------------------------------------------------------------ > # Click2Dial Test > # > ------------------------------------------------------------------------ > if (src_ip==myself && (search("^From: <sip:700@") || > search("^To: === message truncated === __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/

1 0

[Serusers] SER CANCEL Processing
by a c 23 May '05

23 May '05

Seems like SER does not respond back to CANCEL Requests. Is this a issue / some config problem on my side? ngrep traces attached interface: eth0 (192.168.0.0/255.255.255.0) filter: ip and ( port 5060 ) # U 192.168.0.2:5062 -> 192.168.0.5:5060 INVITE sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voi p.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. 0.2..CSeq: 940 INVITE..Max-Forwards: 70..Content-Type: application/sdp..Use r-Agent: X-Lite release 1103m..Content-Length: 258....v=0..o=102 7659613 76 59633 IN IP4 192.168.0.2..s=X-Lite..c=IN IP4 192.168.0.2..t=0 0..m=audio 80 00 RTP/AVP 0 8 3 98 96..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpma p:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:96 telephone-event/8000..a=fm tp:96 0-15.. # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.16 8.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>. .Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: 940 INVIT E..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length: 0..Warn ing: 392 192.168.0.5:5060 "Noisy feedback tells: pid=2943 req_src_ip=192.1 68.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999 999(a)voip.com via_cnt==1".... # U 192.168.0.5:5060 -> 192.168.0.3:5060 INVITE sip:9999999999@voip.com SIP/2.0..Record-Route: <sip:192.168.0.5;ftag =3282942385;lr=on>..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bKe034.1018c5d 1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;branch=z9hG4bKE29503D9D8C F4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To : <sip:9999999999@voip.com>..Contact: <sip:102@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..CSeq: 940 INVITE..Max-For wards: 16..Content-Type: application/sdp..User-Agent: X-Lite release 1103m. .Content-Length: 258....v=0..o=102 7659613 7659633 IN IP4 192.168.0.2..s=X- Lite..c=IN IP4 192.168.0.2..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 96..a=rtpm ap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC /8000..a=rtpmap:96 telephone-event/8000..a=fmtp:96 0-15.. # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 100 Trying..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: < sip:9999999999@voip.com>..Call-Id: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192 .168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bKe034 .1018c5d1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;branch=z9hG4bKE29 503D9D8CF4A479B6D9DDB17C5CFC1..Content-Length: 0.... # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 180 Ringing..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=27378..Call-Id: 29469D19-CDF1-49ED-8245-A67E1 B1DBB88@192.168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP 192.168.0.5;branch= z9hG4bKe034.1018c5d1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;branch =z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..Date: Mon, 23 May 2005 22:15:14 G MT..Contact: sip:192.168.0.3..User-Agent: sipX/2.5.2 (WinNT)..Accept-Langua ge: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces..Content-Length: 0 .... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 180 Ringing..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=27378..Call-Id: 29469D19-CDF1-49ED-8245-A67E1 B1DBB88@192.168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP 192.168.0.2:5062;rp ort=5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..Date: Mon, 23 May 2005 22:15:14 GMT..Contact: sip:192.168.0.3..User-Agent: sipX/2.5.2 (WinNT) ..Accept-Language: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOT IFY, REGISTER, SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces..Co ntent-Length: 0.... # U 192.168.0.2:5062 -> 192.168.0.5:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voi p.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. 0.2..CSeq: 940 CANCEL..Max-Forwards: 70..User-Agent: X-Lite release 1103m.. Content-Length: 0.... # U 192.168.0.2:5062 -> 192.168.0.5:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voi p.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. 0.2..CSeq: 940 CANCEL..Max-Forwards: 70..User-Agent: X-Lite release 1103m.. Content-Length: 0.... # U 192.168.0.2:5062 -> 192.168.0.5:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.2:5062;r port;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voi p.com:5062>;tag=3282942385..To: <sip:9999999999@voip.com>..Contact: <sip:10 2@192.168.0.2:5062>..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168. 0.2..CSeq: 940 CANCEL..Max-Forwards: 70..User-Agent: X-Lite release 1103m.. Content-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;b ranch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com: 5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 ae5b6e96e979a-6e9f..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168.0 .2..CSeq: 940 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Cont ent-Length: 0..Warning: 392 192.168.0.5:5060 "Noisy feedback tells: pid=29 51 req_src_ip=192.168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999999@voip.com via_cnt==0".... # U 192.168.0.5:5060 -> 192.168.0.3:5060 CANCEL sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.5;branch =z9hG4bKe034.1018c5d1.0..From: 102 <sip:102@voip.com:5062>;tag=3282942385.. Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..To: <sip:9999999 999(a)voip.com>..CSeq: 940 CANCEL..User-Agent: Sip EXpress router(0.9.2 (i386 /linux))..Content-Length: 0.... # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 200 OK..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip: 9999999999(a)voip.com>..Call-Id: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168 .0.2..Cseq: 940 CANCEL..Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bKe034.101 8c5d1.0..Content-Length: 0.... # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 487 Request Terminated..From: 102 <sip:102@voip.com:5062>;tag=32829 42385..To: <sip:9999999999@voip.com>;tag=27378..Call-Id: 29469D19-CDF1-49ED -8245-A67E1B1DBB88@192.168.0.2..Cseq: 940 INVITE..Via: SIP/2.0/UDP 192.168. 0.5;branch=z9hG4bKe034.1018c5d1.0..Via: SIP/2.0/UDP 192.168.0.2:5062;rport= 5062;branch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..Date: Mon, 23 May 2005 22:15:44 GMT..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REG ISTER, SUBSCRIBE..User-Agent: sipX/2.5.2 (WinNT)..Accept-Language: en..Supp orted: sip-cc, sip-cc-01, timer, replaces..Contact: sip:192.168.0.3..Conten t-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.3:5060 ACK sip:9999999999@voip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.5;branch=z9 hG4bKe034.1018c5d1.0..From: 102 <sip:102@voip.com:5062>;tag=3282942385..Cal l-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88@192.168.0.2..To: <sip:9999999999 @voip.com>;tag=27378..CSeq: 940 ACK..User-Agent: Sip EXpress router(0.9.2 ( i386/linux))..Content-Length: 0.... # U 192.168.0.3:5060 -> 192.168.0.5:5060 SIP/2.0 200 OK..From: 102 <sip:102@voip.com:5062>;tag=3282942385..To: <sip: 9999999999(a)voip.com>..Call-Id: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168 .0.2..Cseq: 940 CANCEL..Contact: <sip:9999999999@192.168.0.3:5060>..Via: SI P/2.0/UDP 192.168.0.5;branch=z9hG4bKe034.1018c5d1.0..Date: Mon, 23 May 2005 22:15:44 GMT..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REG ISTER, SUBSCRIBE..User-Agent: sipX/2.5.2 (WinNT)..Accept-Language: en..Supp orted: sip-cc, sip-cc-01, timer, replaces..Content-Length: 0.... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;b ranch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com: 5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 ae5b6e96e979a-6e9f..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168.0 .2..CSeq: 940 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Cont ent-Length: 0..Warning: 392 192.168.0.5:5060 "Noisy feedback tells: pid=29 51 req_src_ip=192.168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999999@voip.com via_cnt==0".... # U 192.168.0.5:5060 -> 192.168.0.2:5062 SIP/2.0 408 Request Timeout..Via: SIP/2.0/UDP 192.168.0.2:5062;rport=5062;b ranch=z9hG4bKE29503D9D8CF4A479B6D9DDB17C5CFC1..From: 102 <sip:102@voip.com: 5062>;tag=3282942385..To: <sip:9999999999@voip.com>;tag=a6a1c5f60faecf035a1 ae5b6e96e979a-6e9f..Call-ID: 29469D19-CDF1-49ED-8245-A67E1B1DBB88(a)192.168.0 .2..CSeq: 940 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Cont ent-Length: 0..Warning: 392 192.168.0.5:5060 "Noisy feedback tells: pid=29 51 req_src_ip=192.168.0.2 req_src_port=5062 in_uri=sip:9999999999@voip.com out_uri=sip:9999999999@voip.com via_cnt==0".... __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] sip responses
by a c 23 May '05

23 May '05

when SER gets a provisional response {100/180/..}, how do I figure out if the response is a 100 / 180 / ? What variable holds the response code? thank you ac __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] duplicate invite
by Ser Users 23 May '05

23 May '05

hello, I am using ser 0.8.14 with tm and acc modules and I have multiple "failure_route" blocks. when the call setup takes longer the client sends a second invite with the same call-id which I see in accounting that is not logged (it is absorbed by tm module as a retransmission I think) everything is OK, but the main route block is still executed for the retransmitted invite. is there a way I can configure ser not to execute the main routing block for the retransmitted invite ? I have tried: --------------------------------------- if (!t_newtran()) { break; } OR if(t_lookup_request) { break; } --------------------------------------- in the main route block but I get: "WARNING: script writer didn't release transaction" and a "server error" at client side. is 0.8.14 the latest "stable" version of ser ? thanks, Razvan Radu

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[Serusers] Re: [Serdev] Bug? Inconsistent return of loose_route function
by Jan Janak 23 May '05

23 May '05

Hello, comments inline. On 23-05-2005 18:57, Cesc Santasusana wrote: > Hi, > > I found what i think is a bug in the return value of loose_route() ... > bear with me because it is a rather long email. > > First, the scenario: > > phone1 ----- SER1 ----- SER2 ----- phone2 > > p1 to s1 and p2 to s2 use UDP > s1 to s2 uses tcp or tls > > The bug comes into place when there is a change of protocol in ser, > from UDP into TCP or TLS. > We do not use a preloaded route set from the phones. > What ser does is it adds 2 route headers, using the r2 parameter. > Route: <route1>, <route2> > > CASE A: -------------------------------------------- > In this scenario, say p1 places a call to p2, going through s1 and s2, > creating the route set with the record-route. > Now, let's say p2 generates the BYE message. > When the BYE gets to s1, the last hop, in the config we do: > if( loose_route () ) { > t_relay(); > } > But, and here is the inconsistency, loose_route (remember we are using > TCP/TLS, so there are 2 routes in one header field) returns FALSE. > > CASE B: -------------------------------------------- > If this experiment is repeated using ALL UDP, the route headers when they > get to s1 contain only one route, and the call to loose_route() returns .... TRUE! > > > Unless the config file does not allow blind sip message relaying, both cases end > up working ... the BYE gets to the final destination. But if the config file does not > accept relaying ... the request-uri in the BYE is not taken as a local domain ... and > thus the message is rejected ... (To Alex Mack: could this be the solution to your problem? > the BYEs not reaching the phone? add some log output and check it ... ) > > ================================================================ > So, this is not consistent. Now, what is the solution? > I think that both case A and B should return the same, but what, true or false? Yes, they should return the same value in this case and it should be true (see below). It looks like you discovered a bug. > As I see it, the solution causing less trouble is that both return true. So, the last proxy > considers this BYE as being loose_routed. > For this ... change the code in: > rr::loose_route.c:: static inline int after_loose(struct sip_msg* _m, int preloaded) > ..... > if (enable_double_rr && is_2rr(&puri.params)) { > .... > if (res > 0) { /* No next route found */ > DBG("after_loose: No next URI found\n"); > return (preloaded ? NOT_RR_DRIVEN : RR_DRIVEN); > or simply > return (preloaded RR_DRIVEN); > > > And now ... the other option is that both return false ... but this would make big changes. > Correct me if i am wrong, but wouldn't this be the compliant solution? > Considering that the r-uri is not part of the route set, when the last proxy processes > all route headers and there are no more left, there is no loose routing done. The > destination is taken from the r-uri, not the route set ... thus loose_route should return > false. > Or I am getting it all wrong? rfc and goal of loose_route() ? The meaning of the return value of loose_route is as follows (or should be): 1 - Record routing was used to route the message 0 - Record routing was not (usually requests establishing a dialog or transactional requests). So the function should return true in the mentioned case, because record routing was used although the message does not contain any Route header fields anymore. In other words the purpose of loose_route return value is to distinguish mid-dialog requests (such as ACK or BYE) from dialog-establishing requests (such as INVITE). The reason why we have to do this is that we typically do not apply all the logic in the script to mid-dialog requests (because you should not change the request-uri of the request anymore once the dialog has been established -- changes caused by loose_route do not count). You may also wonder why don't we use the contents of the Request-URI for this -- there should be the Contact value of phone1 in the Request-URI in mid-dialog requests and the value should not match uri==myself so all we need to do is to enclose all the processing logic within if (uri == myself) condition. This would work in a perfect world without the need to be backwards compatible with strict routers. In our world we have to be backwards compatible with strict routers and we have to deal with broken implementations too (that strip ;lr parameters). Due to backwards compatibility with strict routers it can happen that the Request-URI will contain the URI of the proxy that is currently processing the request and subsequent if (uri==myself) would be true. Jan.

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[Serusers] Bug? Inconsistent return of loose_route function
by Cesc Santasusana 23 May '05

23 May '05

Hi, I found what i think is a bug in the return value of loose_route() ... bear with me because it is a rather long email. First, the scenario: phone1 ----- SER1 ----- SER2 ----- phone2 p1 to s1 and p2 to s2 use UDP s1 to s2 uses tcp or tls The bug comes into place when there is a change of protocol in ser, from UDP into TCP or TLS. We do not use a preloaded route set from the phones. What ser does is it adds 2 route headers, using the r2 parameter. Route: <route1>, <route2> CASE A: -------------------------------------------- In this scenario, say p1 places a call to p2, going through s1 and s2, creating the route set with the record-route. Now, let's say p2 generates the BYE message. When the BYE gets to s1, the last hop, in the config we do: if( loose_route () ) { t_relay(); } But, and here is the inconsistency, loose_route (remember we are using TCP/TLS, so there are 2 routes in one header field) returns FALSE. CASE B: -------------------------------------------- If this experiment is repeated using ALL UDP, the route headers when they get to s1 contain only one route, and the call to loose_route() returns .... TRUE! Unless the config file does not allow blind sip message relaying, both cases end up working ... the BYE gets to the final destination. But if the config file does not accept relaying ... the request-uri in the BYE is not taken as a local domain ... and thus the message is rejected ... (To Alex Mack: could this be the solution to your problem? the BYEs not reaching the phone? add some log output and check it ... ) ================================================================ So, this is not consistent. Now, what is the solution? I think that both case A and B should return the same, but what, true or false? As I see it, the solution causing less trouble is that both return true. So, the last proxy considers this BYE as being loose_routed. For this ... change the code in: rr::loose_route.c:: static inline int after_loose(struct sip_msg* _m, int preloaded) ..... if (enable_double_rr && is_2rr(&puri.params)) { .... if (res > 0) { /* No next route found */ DBG("after_loose: No next URI found\n"); return (preloaded ? NOT_RR_DRIVEN : RR_DRIVEN); or simply return (preloaded RR_DRIVEN); And now ... the other option is that both return false ... but this would make big changes. Correct me if i am wrong, but wouldn't this be the compliant solution? Considering that the r-uri is not part of the route set, when the last proxy processes all route headers and there are no more left, there is no loose routing done. The destination is taken from the r-uri, not the route set ... thus loose_route should return false. Or I am getting it all wrong? rfc and goal of loose_route() ? Regards, Cesc Unclassified

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[Serusers] Audio Problems on SER !!!
by Felipe Martins 23 May '05

23 May '05

Hi everybody, I'm experiencing some problems with the audio of my server. My SER is configured with mysql auth and was working very well the whole friday, there were 6 people talking to each other within our private network, but today, I can't make a decent call, because the noise is very loud at every soft or hard phone I try internally. What May be that ?? Is there documented some codec audio problem about that ? Could it be a problem in my network ? Another question, I have ser worknig at a private network, with his Default Route to one of my internet gateway, My tests machine are connected via a hub just for tests, and this hub to my private network, which leaves to the internet. When I test it internallty there no problem occurs, but when I swich off the eth cable that leads to the internet (I mean the one that is my default route) SER stop authenticating my users, and I can't make a internal call. Is this a bug ?? Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

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Re: [Serusers] UAC module (backport to 0.9.0)
by info＠beeplove.com 23 May '05

23 May '05

Does this module work on ser-0.9.0? Or, was you guys able to make it work on ser-0.9.0 If you were able to make it worked, would you please share the tricks, Thanks, Mohammad Original Message: ----------------- From: Michael Ulitskiy mdu113(a)acedsl.com Date: Mon, 23 May 2005 12:01:40 -0400 To: daniel(a)voice-system.ro, serusers(a)iptel.org Subject: Re: [Serusers] UAC module (backport to 0.9.0) On Sunday 22 May 2005 04:51 am, Daniel-Constantin Mierla wrote: > Hello, > > the issue with CSeq is known and it is stated in the documentation (see > the note there): > > http://www.voice-system.ro/docs/uac/ar01s04.html#uac_auth Thanks. Actually I read that docs, but somehow overlooked that note. Thanks for pointing it out. > Daniel Michael > > On 05/16/05 22:05, Michael Ulitskiy wrote: > > >Hello, > > > >I succeeded to make UAC module calculate credentials and > >resend message with the following simple patch on t_reply.c > >of tm module: > > > >--- t_reply.c.bak 2005-05-13 14:57:25.000000000 -0400 > >+++ t_reply.c 2005-05-13 15:16:45.000000000 -0400 > >@@ -761,6 +761,10 @@ > > a callback; save branch count to be able to determine > > later if new branches were initiated */ > > branch_cnt=Trans->nr_of_outgoings; > >+ /* also append the current reply to the transaction to > >+ * make it available in failure routes - a kind of "fake" > >+ * save of the final reply per branch */ > >+ Trans->uac[branch].reply = reply; > > > > /* run ON_FAILURE handlers ( route and callbacks) */ > > if ( has_tran_tmcbs( Trans, TMCB_ON_FAILURE_RO|TMCB_ON_FAILURE) > >@@ -769,6 +773,11 @@ > > picked_branch==branch?reply:Trans->uac[picked_branch].reply, > > picked_code); > > } > >+ > >+ /* now reset it; after the failure logic, the reply may > >+ * not be stored any more and we don't want to keep into > >+ * transaction some broken reference */ > >+ Trans->uac[branch].reply = 0; > > > > /* look if the callback perhaps replied transaction; it also > > covers the case in which a transaction is replied localy > > > >It's actually taken from cvs-head. > >Now ser.cfg written below works as expected with just one > >important issue. Authentication doesn't work anyway because > >ser doesn't increase CSeq number when resending message > >(INVITE) with credentials. I'm trying to authenticate > >to asterisk and in case of asterisk it immediately replies > >"503 Service Unavailable" if it receives INVITE with the same CSeq. > >I thought it could be asterisk-specific problem, but RFC3261 in > >section "8.1.3.5 Processing 4xx Responses" says: > >"new request constitutes a new transaction and SHOULD have the same > >value of the Call-ID, To, and From of the previous request, but the CSeq > >should contain a new sequence number that is one higher than the previous." > >So I guess this is not an asterisk-specific. > >It is my understanding that in order for this to work ser must increment > >CSeq when authenticating to UAS, but decrement CSeq in all subsequent > >in-dialog messages that will be sent to call originator including BYEs. > >This, in turn, requires for ser to be call-statefull which is not the case. > >Conclusion: uac authentication in ser is not possible. > >Please correct me if I'm wrong (I honestly want to be wrong on this matter:)) > >Thank you, > > > >Michael > > > >On Sunday 15 May 2005 12:36 am, you wrote: > > > > > >>Michael Ulitskiy wrote: > >> > >> > >> > >>>Hello, > >>> > >>>Has anyone succeeded in getting UAC authentication to work? > >>>I'm doing the following: > >>> > >>>route { > >>> resetflag(1); > >>> t_on_failure("1"); > >>> route(1); > >>>} > >>> > >>>route[1] > >>>{ > >>> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") { > >>> sl_send_reply("479", "We don't forward to private IP addresses"); > >>> break; > >>> }; > >>> if (!t_relay()) { > >>> sl_reply_error(); > >>> }; > >>>} > >>> > >>>failure_route[1] { > >>> # authentication reply received? > >>> if ( t_check_status("401|407") ) { > >>> if (!isflagset(1) && uac_auth()) { > >>> setflag(1); > >>> t_on_failure("1"); > >>> append_branch(); > >>> route(1); > >>> } else { > >>> t_reply("500","Error occurred"); > >>> } > >>> break; > >>> } > >>> > >>>} > >>> > >>>When uac_auth() is called I get the following in the: > >>>0(28973) DEBUG:uac:uac_auth: picked reply is (nil), code 407 > >>>0(28973) BUG:uac:uac_auth: empty reply on picked branch > >>> > >>>Any suggestions or ideas? > >>>Thank you, > >>> > >>>Michael > >>> > >>> > >>> > >>> > >>> > >>I am also getting same thing with different msg .. > >>BUG:uac:uac_auth: empty reply on picked branch > >> > >>Anybody know how to overcome this? > >> > >>Mohammad > >> > >> > >> > >> > > > >_______________________________________________ > >Serusers mailing list > >Serusers(a)iptel.org > >http://mail.iptel.org/mailman/listinfo/serusers > > > > > > > _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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Re: [Serusers] Advice needed
by Michael Ulitskiy 23 May '05

23 May '05

Yes, I guess it's another option. Not that I like it much though :) BTW I haven't seen providers that support it. 2all: Have you seen any? Thanks, Michael On Monday 23 May 2005 04:50 am, you wrote: > just an idea: > > does any PSTN provider supports TLS? If some does you have your auth > problem solved with the TLS support from SER. > > Samuel > > > Unclassified. > >>> "Greger V. Teigre" <greger(a)teigre.com> 05/22/05 08:57AM >>> > See inline. > > Michael Ulitskiy wrote: > > On Saturday 21 May 2005 02:31 am, you wrote: > >> I would say SER is what you need, except that you struggle with the > >> authentication. You have the following scenarios: > >> 1. PSTN termination with IP-based access control (easiest) > >> 2. PSTN termination with authentication of all INVITEs (yes, that's > >> the UAC module. You should contact the maintainer, Ramona-Elena > >> Modroiu about the status. I thought it was reported to work, but > >> haven't tried myself) > >> 3. PSTN termination with registration and authentication of > REGISTER > >> (but not INVITEs). Use sipsak to generate a REGISTER for your box. > >> > >> #2 requires that all INVITEs are sent twice and is not a very good > >> option. I would seek out PSTN providers who will give you #1. > >> g-) > > > > UAC module doesn't work and I think won't work unless ser is made > > call-statefull, 'cause it needs to adjust cseq within dialog. I > > posted my findings to this list > > several days ago (UAC module (backport to 0.9.0). Nobody replied so > I > > guess > > nobody knows the way to make it work. > > I saw your post on serusers, yes, but not on serdev. Because you cannot > make > a module work, doesn't mean it doesn't work for all, so as I said, if > you > have found a bug, post it to serdev (preferably) or directly to the > maintainer. That's the way open source software work... > > > As for ip auth I guess it's just not good enough. UDP invites don't > > require any handshake it's not hard at all to spoof ip address. I > > believe sending 2 invites worth the security it actually adds. > > Yes, but you can also do TCP. > > > Also I don't understand what you mean by #3. Taking ip address from > > authenticated REGISTER and then doing IP auth on that? > > No, using sipsak to actually do a REGISTER on behalf of your ser. No IP > > auth, basically it makes your ser a registered client of the GW. Of > course, > if INVITEs still must be authenticated, you are back to the UAC module > > problem. > > g-) > > > > Thanks, > > > > Michael > > > >> Michael Ulitskiy wrote: > >>> Hello, > >>> > >>> I'd like ask for advice on what is in your opinion the best > solution > >>> in the following scenario. > >>> I have a bunch of sip servers (asterisk boxes as my users need pbx > >>> functionality) that can make sip call to each other and my PSTN > >>> gateway. Now I want to purchase PSTN terminitaion in several > >>> different markets (and probably more in the future). All those > >>> terminations will require authentication. > >>> I want all my boxes when they see non-local call to send it to a > >>> central routing server that would determine where this call should > >>> be sent and authenticate to the appropriate provider so that I > don't > >>> have to configure all credentials on all asterisk boxes. Also I > want > >>> it not to deal with the media at all. All media streams should go > >>> directly from asterisk box to the PSTN termination provider. > >>> So basically it should be central SIP router that is able to > >>> authenticate calls if neccessary. > >>> I thought I could do it with SER and its UAC module, but it > appears > >>> UAC module doesn't work and probably won't work (see my previous > >>> post in this list about UAC backport to 0.9.0). > >>> Also I don't want to use asterisk in this place as asterisk always > >>> wants to stay in media path and I'd really like to avoid of > getting > >>> into hassle with re-invites. > >>> So the question is what are my options and what you would advice > >>> as a solution. Are there any software out there that can do it > >>> (preferably open-source, of course) or what else you could suggest > >>> to do to get desired results. > >>> Thanks a lot, > >> > >> > > > > -- > > See you later, > > Michael > > > > > > ------------------------------------------------------- > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Re: sip load balancing with 0.9.0 + crash
by a c 23 May '05

23 May '05

hello, I am getting a core dump when using the dispatcher module..Could someone help? I am using http://lists.iptel.org/pipermail/serdev/2004-August/002592.html to setup the config. thank you ac > > --config file-- > # $Id$ > # sample config file for dispatcher module > # > > debug=9 # debug level (cmd line: > -dddddddddd) > fork=no > log_stderror=yes # (cmd line: -E) > > children=2 > check_via=no # (cmd. line: -v) > dns=off # (cmd. line: -r) > rev_dns=off # (cmd. line: -R) > port=5060 > > # for more info: sip_router -h > > # ------------------ module loading > ---------------------------------- > > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > loadmodule "/usr/local/lib/ser/modules/sl.so" > # loadmodule "/usr/local/lib/ser/modules/tm.so" > loadmodule > "/usr/local/lib/ser/modules/dispatcher.so" > > # ----------------- setting module-specific > parameters --------------- > # -- dispatcher params -- > > modparam("dispatcher", "list_file", > "/usr/local/etc/ser/dispatcher.list") > # modparam("dispatcher", "force_dst", 1) > > route{ > if ( !mf_process_maxfwd_header("10") ) > { > sl_send_reply("483","To Many Hops"); > drop(); > }; > > ds_select_dst("1", "0"); > > forward(uri:host, uri:port); > # t_relay(); > } > > --dispatcher.list-- > # $Id$ > # dispathcher destination sets > > #servers > 1 sip:192.168.0.2:5060 > 1 sip:192.168.0.3:5060 > > --error output-- > >ser -cf new2.cfg > > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816402c frag. 0x8164014 (size=128) on 1 -th hit > 0(7246) loading module > /usr/local/lib/ser/modules/maxfwd.so > 0(7246) qm_malloc(0x812fc20, 16) called from > sr_module.c: register_module(136) > 0(7246) qm_malloc(0x812fc20, 16) returns address > 0x81640dc frag. 0x81640c4 (size=16) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816411c frag. 0x8164104 (size=128) on 1 -th hit > 0(7246) loading module > /usr/local/lib/ser/modules/sl.so > 0(7246) qm_malloc(0x812fc20, 16) called from > sr_module.c: register_module(136) > 0(7246) qm_malloc(0x812fc20, 16) returns address > 0x81641cc frag. 0x81641b4 (size=16) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816420c frag. 0x81641f4 (size=128) on 1 -th hit > 0(7246) loading module > /usr/local/lib/ser/modules/dispatcher.so > 0(7246) qm_malloc(0x812fc20, 16) called from > sr_module.c: register_module(136) > 0(7246) qm_malloc(0x812fc20, 16) returns address > 0x81642bc frag. 0x81642a4 (size=16) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x81642fc frag. 0x81642e4 (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x81643ac frag. 0x8164394 (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816445c frag. 0x8164444 (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 13) called from > modparam.c: set_mod_param_regex(88) > 0(7246) qm_malloc(0x812fc20, 16) returns address > 0x816450c frag. 0x81644f4 (size=16) on 1 -th hit > 0(7246) set_mod_param_regex: dispatcher matches > module dispatcher > 0(7246) set_mod_param_regex: found <list_file> in > module dispatcher > [/usr/local/lib/ser/modules/dispatcher.so] > 0(7246) qm_free(0x812fc20, 0x816450c), called from > modparam.c: set_mod_param_regex(154) > 0(7246) qm_free: freeing frag. 0x81644f4 alloc'ed > from modparam.c: set_mod_param_regex(88) > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x816454c frag. 0x8164534 (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x81645fc frag. 0x81645e4 (size=128) on 1 -th hit > 0(7246) find_export: found > <mf_process_maxfwd_header> in module maxfwd > [/usr/local/lib/ser/modules/maxfwd.so] > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x81646ac frag. 0x8164694 (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 20) called from > route_struct.c: mk_elem(70) > 0(7246) qm_malloc(0x812fc20, 20) returns address > 0x81646fc frag. 0x81646e4 (size=20) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 20) called from > route_struct.c: mk_exp(54) > 0(7246) qm_malloc(0x812fc20, 20) returns address > 0x8164740 frag. 0x8164728 (size=20) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164784 frag. 0x816476c (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164834 frag. 0x816481c (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x81648e4 frag. 0x81648cc (size=128) on 1 -th hit > 0(7246) find_export: found <sl_send_reply> in > module sl_module [/usr/local/lib/ser/modules/sl.so] > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x8164994 frag. 0x816497c (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x81649e4 frag. 0x81649cc (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x8164a34 frag. 0x8164a1c (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164a84 frag. 0x8164a6c (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164b34 frag. 0x8164b1c (size=128) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 128) called from > cfg.lex: addstr(572) > 0(7246) qm_malloc(0x812fc20, 128) returns address > 0x8164be4 frag. 0x8164bcc (size=128) on 1 -th hit > 0(7246) find_export: found <ds_select_dst> in > module dispatcher > [/usr/local/lib/ser/modules/dispatcher.so] > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x8164c94 frag. 0x8164c7c (size=32) on 1 -th hit > 0(7246) qm_malloc(0x812fc20, 32) called from > route_struct.c: mk_action(89) > 0(7246) qm_malloc(0x812fc20, 32) returns address > 0x8164ce4 frag. 0x8164ccc (size=32) on 1 -th hit > 0(7246) routing table 0: > 0(7246) if ( 0(7246) NOT( 0(7246) > external_module_call( 0(7246) f_ptr<0x4001a060> > 0(7246) , "10" 0(7246) ); 0(7246) > external_module_call( 0(7246) f_ptr<0x4001a060> > 0(7246) , "10" 0(7246) ); 0(7246) ) 0(7246) ) { > 0(7246) external_module_call( 0(7246) > f_ptr<0x4016ba20> 0(7246) , "483" 0(7246) , "To Many > Hops" ... WARNING: no fork mode and more than one listen address found(will use only the the first one)^M config file ok, exiting...^M 0(7246) DISPATCHER: destroy module ...^M 0(7246) qm_free(0x812fc20, (nil)), called from dispatch.c: ds_destroy_list(347) ^M 0(7246) BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting^M Aborted (core dumped)^M __________________________________ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail

3 3

[Serusers] Re: SIP load balancing cont..
by a c 23 May '05

23 May '05

to clarify the question, how do you tell what messages SER is sending out {without using ngrep}? I would like to get a printout everytime it sends 'trying', 'ringing', / 200 OK. What would the config look like? Is this possible? alex --- a c <lra101(a)yahoo.com> wrote: > > this worked with t_check_status(). One thing I could > not figure out is how to get a log printout when I > receive a 200 OK. How do I know if I received a 200 > OK? > I can't put this logic in the main route block. > > thank you > ac > --- a c <lra101(a)yahoo.com> wrote: > > > > I need to route to a failure route when there is a > > certain kind of failure. For example, If I don't > > receive a 'Trying' back on the initial attempt{no > > response to Invite}, then I would route using the > > failure route. For all other failures {i.e. 404, > no > > answer, 486, etc..}, I don't try any other routes. > > How > > can this be configured? > > > > thank you > > ac > > > > > > > > Discover Yahoo! > > Have fun online with music videos, cool games, IM > > and more. Check it out! > > http://discover.yahoo.com/online.html > > > > > > __________________________________ > Do you Yahoo!? > Make Yahoo! your home page > http://www.yahoo.com/r/hs > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

Re: [Serusers] UAC module (backport to 0.9.0)
by Michael Ulitskiy 23 May '05

23 May '05

Hello, I succeeded to make UAC module calculate credentials and resend message with the following simple patch on t_reply.c of tm module: --- t_reply.c.bak 2005-05-13 14:57:25.000000000 -0400 +++ t_reply.c 2005-05-13 15:16:45.000000000 -0400 @@ -761,6 +761,10 @@ a callback; save branch count to be able to determine later if new branches were initiated */ branch_cnt=Trans->nr_of_outgoings; + /* also append the current reply to the transaction to + * make it available in failure routes - a kind of "fake" + * save of the final reply per branch */ + Trans->uac[branch].reply = reply; /* run ON_FAILURE handlers ( route and callbacks) */ if ( has_tran_tmcbs( Trans, TMCB_ON_FAILURE_RO|TMCB_ON_FAILURE) @@ -769,6 +773,11 @@ picked_branch==branch?reply:Trans->uac[picked_branch].reply, picked_code); } + + /* now reset it; after the failure logic, the reply may + * not be stored any more and we don't want to keep into + * transaction some broken reference */ + Trans->uac[branch].reply = 0; /* look if the callback perhaps replied transaction; it also covers the case in which a transaction is replied localy It's actually taken from cvs-head. Now ser.cfg written below works as expected with just one important issue. Authentication doesn't work anyway because ser doesn't increase CSeq number when resending message (INVITE) with credentials. I'm trying to authenticate to asterisk and in case of asterisk it immediately replies "503 Service Unavailable" if it receives INVITE with the same CSeq. I thought it could be asterisk-specific problem, but RFC3261 in section "8.1.3.5 Processing 4xx Responses" says: "new request constitutes a new transaction and SHOULD have the same value of the Call-ID, To, and From of the previous request, but the CSeq should contain a new sequence number that is one higher than the previous." So I guess this is not an asterisk-specific. It is my understanding that in order for this to work ser must increment CSeq when authenticating to UAS, but decrement CSeq in all subsequent in-dialog messages that will be sent to call originator including BYEs. This, in turn, requires for ser to be call-statefull which is not the case. Conclusion: uac authentication in ser is not possible. Please correct me if I'm wrong (I honestly want to be wrong on this matter:)) Thank you, Michael On Sunday 15 May 2005 12:36 am, you wrote: > Michael Ulitskiy wrote: > > >Hello, > > > >Has anyone succeeded in getting UAC authentication to work? > >I'm doing the following: > > > >route { > > resetflag(1); > > t_on_failure("1"); > > route(1); > >} > > > >route[1] > >{ > > if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)") { > > sl_send_reply("479", "We don't forward to private IP addresses"); > > break; > > }; > > if (!t_relay()) { > > sl_reply_error(); > > }; > >} > > > >failure_route[1] { > > # authentication reply received? > > if ( t_check_status("401|407") ) { > > if (!isflagset(1) && uac_auth()) { > > setflag(1); > > t_on_failure("1"); > > append_branch(); > > route(1); > > } else { > > t_reply("500","Error occurred"); > > } > > break; > > } > > > >} > > > >When uac_auth() is called I get the following in the: > > 0(28973) DEBUG:uac:uac_auth: picked reply is (nil), code 407 > > 0(28973) BUG:uac:uac_auth: empty reply on picked branch > > > >Any suggestions or ideas? > >Thank you, > > > >Michael > > > > > > > I am also getting same thing with different msg .. > BUG:uac:uac_auth: empty reply on picked branch > > Anybody know how to overcome this? > > Mohammad > >

3 3

Re: [SPAM] [Serusers] Re: Module programming question
by Daniel Corbe 23 May '05

23 May '05

Actually those were incredibly useful. Thank you. Now, when I compile my module I am getting the following error message: 16:18:13 root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc -c -o radius_cache.o radius_cache.c In file included from radius_cache.c:46: radius_cache.h:58: `NAME' undeclared here (not in a function) radius_cache.h:58: syntax error before string constant radius_cache.h:62: warning: initialization from incompatible pointer type I'm not sure what to do about that. It seems to be something internal and specific to MODULE_VERSION in the SER header. My module: http://www.rafb.net/paste/results/N0uQyJ91.html My module's header file: http://www.rafb.net/paste/results/vo3vG345.html Thanks for the help. -Daniel On 5/23/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > There's a simple module, called print that is meant to be mainly an > example. So I recommend you to take a look at its interface and to start > from there your module. > I attacht also several files that a little bit old but still they are > really useful. > > Luck! > > Samuel. > > > Unclassified. > >>> Daniel Corbe <daniel.junkmail(a)gmail.com> 05/22/05 10:41PM >>> > Here's another question: > > I got my module to compile by taking _init out and reworking the > initalization of the module into something else. > > Now when SER loads I get: > > May 22 21:34:00 proxy2 ser: ERROR: no version info in module > </usr/local/lib/ser/modules/radius_cache.so>: Undefined symbol > "module_version" > > What is module_version? How do I define it? > > Thanks. > > -Daniel > > > On 5/22/05, Daniel Corbe <daniel.junkmail(a)gmail.com> wrote: > > Hello, > > > > I have a SER module which I've written in C and I'm trying to > compile. > > When I compile it I get this error: > > > > 20:53:24 > root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc > > -shared -o radius_cache.so radius_cache.o > > radius_cache.o: In function `_init': > > radius_cache.o(.text+0x0): multiple definition of `_init' > > /usr/lib/crti.o(.init+0x0): first defined here > > > > Is there something else in SER that exports _init already? > > > > Thanks. > > > > -Daniel > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Serusers] Module programming question
by Daniel Corbe 23 May '05

23 May '05

Hello, I have a SER module which I've written in C and I'm trying to compile. When I compile it I get this error: 20:53:24 root#proxy2:/usr/local/src/ser-0.9.0/modules/radius_cache>gcc -shared -o radius_cache.so radius_cache.o radius_cache.o: In function `_init': radius_cache.o(.text+0x0): multiple definition of `_init' /usr/lib/crti.o(.init+0x0): first defined here Is there something else in SER that exports _init already? Thanks. -Daniel

2 2

[Serusers] Re: SIP load balancing cont..
by a c 23 May '05

23 May '05

this worked with t_check_status(). One thing I could not figure out is how to get a log printout when I receive a 200 OK. How do I know if I received a 200 OK? I can't put this logic in the main route block. thank you ac --- a c <lra101(a)yahoo.com> wrote: > > I need to route to a failure route when there is a > certain kind of failure. For example, If I don't > receive a 'Trying' back on the initial attempt{no > response to Invite}, then I would route using the > failure route. For all other failures {i.e. 404, no > answer, 486, etc..}, I don't try any other routes. > How > can this be configured? > > thank you > ac > > > > Discover Yahoo! > Have fun online with music videos, cool games, IM > and more. Check it out! > http://discover.yahoo.com/online.html > __________________________________ Do you Yahoo!? Make Yahoo! your home page http://www.yahoo.com/r/hs

1 0

[Serusers] session Timers
by Jorge Crichigno 23 May '05

23 May '05

Hi all, I have a a doubt related with RFC 4028 - Session Timers in the Session Initiation Protocol (SIP). I am developing a new ser module that supports this RFC. While doing tests, I found the following situation: phone1(addpac) SER phone 2 (grandstream) Invite -> <-422 (Lower Min-SE) Ack -> Invite -> Invite -> <-481 (no such call) <-481 (no such call) What I saw was that, in the second Invite, the addpac sends the Invite with the the "to tag" that was attached by SER before (422 response). I belive that this is the reason of the 481 response from grandstream. I think it is a bug of Addpac. Is that correct? Thanks in advance!

2 2

[Serusers] Bug in ucontact.c?
by Francesco Bottà 23 May '05

23 May '05

Hi all, I've a question about removing alias from memory and DB. I've a SER instance in WRITE_BACK DB Mode (db_mode=2) and timer_interval param set to 10 seconds. When I should remove an alias from memory&DB (with 'serctl rm alias 'username'), in WRITE_BACK mode, the expire time of this cancelled alias go to something like 'expires=-1116854859' and increase in time but never disappear from memory neither from my DB...indeed, if I set WRITE-THROUGH mode (db_mode=1) all works ok, and the cancelled alias disappears immediately from mem and DB.... I've this SER ver installed: ser -V version: ser 0.9.2 (i386/linux) flags: STATS: Off, EXTRA_DEBUG, 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 11:28:47 May 6 2005 with gcc 3.3 Any hints? What I'm missing? Many thanx Verbal

1 0

Re: [Serusers] error in compile ser-0.9.0
by Kamran Ahmad 23 May '05

23 May '05

i am using bison-2.0.tar.gz and flex-2.5.4a.tar.gz but unable to solve this issue #ps -A 8961 ? 00:00:00 ser <defunct> 8962 ? 00:00:00 ser <defunct> 8965 ? 00:00:00 ser <defunct> 8966 ? 00:00:00 ser <defunct> 8968 pts/1 00:00:00 ps # serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config --- Grigory Fishilevich <g.fishilevich(a)gmail.com> wrote: > hi, > > these messages have nothing to say. i think you have > a permissions > problem, you should take a look at /var/log/messages > > ? have you ser running or not? > __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

1 0

[Serusers] SUBSCRIBE -> BAD REQEST (Instant Communication Application + PA Module)
by Jaroslaw Gawron 23 May '05

23 May '05

Hello Everybody I am trying to create Instant Communication Application based on MS RTC 1.2 library. I am also using SER 0.8.14 as a SIP server. Voice communication works perfect, messages works perfect but I have a big problem with sending user's status. When I tray to send SUBSCRIBE message I get BAD REQEST response. T 10.1.1.62:3749 -> 10.1.3.10:5060 [AP] SUBSCRIBE sip:pszymanek@sippbx.bb.wasko.pl SIP/2.0 Via: SIP/2.0/TCP 10.1.1.62:8718 Max-Forwards: 70 From: "yarek" <sip:yarek@sippbx.bb.wasko.pl>;tag=d290de95c2124ceca42a748963dd04e9;epid=885 a5b4691 To: <sip:pszymanek@sippbx.bb.wasko.pl> Call-ID: fb0ac597ee2b480f9315e88a9e6bcb06(a)10.1.1.62 CSeq: 2 SUBSCRIBE..Contact: <sip:yarek@sippbx.bb.wasko.pl:8718;maddr=10.1.1.62;transport=tcp> User-Agent: RTC/1.2..Expires: 0 Event: presence..Content-Length: 0 # T 10.1.3.10:5060 -> 10.1.1.62:3749 [AP] SIP/2.0 400 Bad Request Via: SIP/2.0/TCP 10.1.1.62:8718 From: "yarek" <sip:yarek@sippbx.bb.wasko.pl>;tag=d290de95c2124 ceca42a748963dd04e9;epid=885a5b4691 To: <sip:pszymanek@sippbx.bb.wasko.pl>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-3fbe Call-ID: fb0ac597ee2b480f9315e88a9e6bcb06(a)10.1.1.62 CSeq: 2 SUBSCRIBE..Error while parsing headersServer: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 10.1.3.10:5060 "Noisy feedback tells: pid=13753 req_src_ip= 10.1.1.62 req_src_port=3749 in_uri=sip:pszymanek@sippbx.bb.wasko.pl out_uri=sip:pszymanek@sippbx.bb.wasko.pl via_cnt==1" I configured the presence module, added the lines to ser.cfg if (method=="SUBSCRIBE") { log(1, "Subscribe\n"); if (t_newtran()) { log(1, "Registrar\n"); handle_subscription("registrar"); }; break; }; But it still doesn't work. Has anyone faced a similar problem ? Why it doesn't work ? Is the PA module compatible with MS RTC. Why this masseges are in bad format. Any ideas are very welcome. Best regards Jarosław Gawron email: J.Gawron(a)wasko.pl

1 0

[Serusers] Append a header field with dynamic data
by Michael Stier 23 May '05

23 May '05

Hi, I want to append a new header field on a 183 - session progress response with dynamic data: P-hint: dynamic data The string "dynamic data" e. g. could be generated by an external script. Is there any way to reach this inside ser.cfg? My imagination would be something like this: if (status=~"183") { exec_msg('TEMPSTRING=`cat /tmp/string.txt`'); append_hf("P-hint: $TEMPSTRING"); }; Many greetings Michael

2 1

Re: [Serusers] sip load balancing
by a c 23 May '05

23 May '05

hello Iqbal, just need the SIP messages load balance across servers. No RTP. thank you ac --- Iqbal <iqbal(a)gigo.co.uk> wrote: > mediaproxy can do it using dns. Am assuming you want > the rtp stream load > balanced, if its the sip messages, search throuygh > the archives there is > along discussion bout howto use LVS and load > balancing....in short > doesnt seem to work, cause SIP uses udp > > Iqbal > > a c wrote: > > >hello, > > > > got a question on SER. Could SER do load balance > >across SIP servers? > > > >for example: > > > >SIP carrier --> SER --> Media/App Servers > {multiple} > > > >Are there any Configuration examples I can look at? > > > >thank you > >ac > > > > > > > >__________________________________ > >Yahoo! Mail Mobile > >Take Yahoo! Mail with you! Check email on your > mobile phone. > >http://mobile.yahoo.com/learn/mail > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > >. > > > > > > > __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250

8 13

Re: [Serusers] Advice needed
by Samuel Osorio Calvo 23 May '05

23 May '05

just an idea: does any PSTN provider supports TLS? If some does you have your auth problem solved with the TLS support from SER. Samuel Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 05/22/05 08:57AM >>> See inline. Michael Ulitskiy wrote: > On Saturday 21 May 2005 02:31 am, you wrote: >> I would say SER is what you need, except that you struggle with the >> authentication. You have the following scenarios: >> 1. PSTN termination with IP-based access control (easiest) >> 2. PSTN termination with authentication of all INVITEs (yes, that's >> the UAC module. You should contact the maintainer, Ramona-Elena >> Modroiu about the status. I thought it was reported to work, but >> haven't tried myself) >> 3. PSTN termination with registration and authentication of REGISTER >> (but not INVITEs). Use sipsak to generate a REGISTER for your box. >> >> #2 requires that all INVITEs are sent twice and is not a very good >> option. I would seek out PSTN providers who will give you #1. >> g-) > > UAC module doesn't work and I think won't work unless ser is made > call-statefull, 'cause it needs to adjust cseq within dialog. I > posted my findings to this list > several days ago (UAC module (backport to 0.9.0). Nobody replied so I > guess > nobody knows the way to make it work. I saw your post on serusers, yes, but not on serdev. Because you cannot make a module work, doesn't mean it doesn't work for all, so as I said, if you have found a bug, post it to serdev (preferably) or directly to the maintainer. That's the way open source software work... > As for ip auth I guess it's just not good enough. UDP invites don't > require any handshake it's not hard at all to spoof ip address. I > believe sending 2 invites worth the security it actually adds. Yes, but you can also do TCP. > Also I don't understand what you mean by #3. Taking ip address from > authenticated REGISTER and then doing IP auth on that? No, using sipsak to actually do a REGISTER on behalf of your ser. No IP auth, basically it makes your ser a registered client of the GW. Of course, if INVITEs still must be authenticated, you are back to the UAC module problem. g-) > Thanks, > > Michael > >> Michael Ulitskiy wrote: >>> Hello, >>> >>> I'd like ask for advice on what is in your opinion the best solution >>> in the following scenario. >>> I have a bunch of sip servers (asterisk boxes as my users need pbx >>> functionality) that can make sip call to each other and my PSTN >>> gateway. Now I want to purchase PSTN terminitaion in several >>> different markets (and probably more in the future). All those >>> terminations will require authentication. >>> I want all my boxes when they see non-local call to send it to a >>> central routing server that would determine where this call should >>> be sent and authenticate to the appropriate provider so that I don't >>> have to configure all credentials on all asterisk boxes. Also I want >>> it not to deal with the media at all. All media streams should go >>> directly from asterisk box to the PSTN termination provider. >>> So basically it should be central SIP router that is able to >>> authenticate calls if neccessary. >>> I thought I could do it with SER and its UAC module, but it appears >>> UAC module doesn't work and probably won't work (see my previous >>> post in this list about UAC backport to 0.9.0). >>> Also I don't want to use asterisk in this place as asterisk always >>> wants to stay in media path and I'd really like to avoid of getting >>> into hassle with re-invites. >>> So the question is what are my options and what you would advice >>> as a solution. Are there any software out there that can do it >>> (preferably open-source, of course) or what else you could suggest >>> to do to get desired results. >>> Thanks a lot, >> >> > > -- > See you later, > Michael > > > ------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] exec module question
by Samuel Osorio Calvo 23 May '05

23 May '05

Hi, I am not sure if SER config file get the error code from the exit() call of your script but I just think so. Assuming it gets the return code, youn should be aware that only three possibilities can be checked from the config file: ret>0 --> config command was successfull ret<0 --> ERROR but: ret=0 --> drop message!!! So, if you return 0 from your script, the message will be dropped, just af it a break had occur... To allow this structure: if (exec_msg("/usr/local/bin/check_register.pl")) { break; } else { if (!radius_www_authorize("")) { You should return a positive value if the scrpit ended OK or a negative value if an error occurred. BTW, using exec|* commands can be a worse solution.......just take a look at older posts about this subject. Hope it helps, Samuel. Unclassified. >>> Daniel Corbe <daniel.junkmail(a)gmail.com> 05/20/05 05:31PM >>> Hello, I've noticed that 90% of my SIP traffic is REGISTER requests. I am trying to implement some sort of caching mechanism (which will inherently be faster than doing a RADIUS lookup on every REGISTER request) using exec and a perl script which does a lookup from a memory-cached file. I'm using exec_msg() to call the perl script; however it is not behaving as I would expect. Here's the code snippet from my SER config. if (exec_msg("/usr/local/bin/check_register.pl")) { break; } else { if (!radius_www_authorize("")) { I am just in preliminary testing stage so I have the perl script merely exiting with 0 status. exec documentation is not clear as to whether it uses the shell return codes to determine the exit status of exec_msg If I am exit(0)ing from the perl script I would expect the if( to fail, not break and do the radius_www_authorize. Is that not correct? Thanks for the help! Regards, Daniel Corbe _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Why does the memory occupied by SER process not decrese?
by lemon 23 May '05

23 May '05

I am using ser 0.8.14. I wrote a program to test the stability of SER. The scenario like this caller----SER----callee. Caller program will send INVITE continuously.Callee program answer each call proxyed by SER. And then Caller program send BYE to finish the call. I run 'top' tool in linux to monitor the usage of memory and cup by SER. I found that with the call establishing speed increasing, the memory occupied by SER process increasing too.After I shut down the test program, the memory ocupied by SER process didn't decrease. Can anyone tell me why? And Does is mean memory leak? Best regards! Karman Wu ---------------------------------------------- 21CNÓÊÏäÈ«ÃæÉýG http://mail.21cn.com/huodong/0504/mail/ ²ÊÆÁÊÖ»ú°×ËÍÀ²!žÏ¿ìÀŽÄÃ! http://qipai.g.21cn.com ÖÇ»ÛŽóÌôÕœŸÍÔÚµÍŒÛ¶á±Š http://super.21cn.com/ ÌåÑéÁíÀàÔŒ»á£¬žÐÊÜ±ðÑùÀËÂþ http://y.21cn.com/club/ Ð¡ÁéÍš¶ÌÐÅÖÐÐÄ£¬¶ÌÐÅ÷ÈÁŠÎÞŒ« http://pas.21cn.com/ È«ÄÜÁÄ2005°æÉÁÁÁµÇ³¡£¡ http://callme.21cn.com

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[Serusers] call PSTN -> SIP
by Kostas Marneris 23 May '05

23 May '05

Hello, I want to make a call from PSTN to a SIP phone. I use a cisco 5300 for answering/handling the PSTN number and then forward the call to a SER box (0.8.14) . POTS -> c5300 --> SER -> SIP softphone My conf on c5300: ! dial-peer voice 1003 voip destination-pattern 12345 session protocol sipv2 session target sip-server codec g711ulaw ! sip-ua sip-server ipv4:X.Y.Z.W ! What is the appropriate configuration on ser.cfg in order to handle this call and forward it to sip:userX@mydomain.com ?? Any hint about this ?? I've already done the opposite scenario (SIP->PSTN) : ser.cfg: if (uri =~ "^sip:2*") { rewritehostport("1.2.3.4:5060"); forward(uri:host, uri:port); break; }; thanks for any help, Kostas --

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Re: [Serusers] error in compile ser-0.9.0
by Kamran Ahmad 23 May '05

23 May '05

how to solve this issue i am root ser is working but #ps -A 3793 ? 00:00:00 ser <defunct> --- Grigory Fishilevich <g.fishilevich(a)gmail.com> wrote: > hi, > > these messages have nothing to say. i think you have > a permissions > problem, you should take a look at /var/log/messages > > ? have you ser running or not? > __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

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[Serusers] Basic Billing For SER 0.8.14 (MySql Based)
by Ozan Blotter 23 May '05

23 May '05

Does anyone know a free or lowcost billing solution which works with SER, before i've tried with some commercials ones but they do not worked with SER and they were Radius Supported ones so was creating its own database. All i need is a few simple PHP files so i can add/remove users from there and a add table for rating depending on prefix and it should use SER's database which is already created. And do sample calculations for me, enter user id/loginname/callerid or whatever and show me caller/callee/duration/time/date/cost and under all a grand total with tax space. So simple but i do not know how to do in PHP. Even some examples or good URL's to read are fine for me. I use SER 0.8.14 but may use 0.9.0 or newer if needed. Thanks, Ozan Blotter

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[Serusers] admin errors in serweb
by Llanos Serna García-Conde 22 May '05

22 May '05

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[Serusers] ser/mysql configuration
by Kim Culhan 22 May '05

22 May '05

After installing mysql 5.0.3 and running ser_mysql.sh then mysql> select * from user; Returns: ERROR 1146 (42S02): Table 'ser.user' doesn't exist Any help is grealty appreciated -kim -- w8hdkim(a)gmail.com

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[Serusers] error in compile ser-0.9.0
by Kamran Ahmad 22 May '05

22 May '05

hi list any one know what could be the problem here is the trace i cannot use serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config make_and_install_output.txt -------------------------------- Makefile.rules:81: action.d: No such file or directory Makefile.rules:81: crc.d: No such file or directory Makefile.rules:81: daemonize.d: No such file or directory Makefile.rules:81: data_lump.d: No such file or directory Makefile.rules:81: data_lump_rpl.d: No such file or directory Makefile.rules:81: dprint.d: No such file or directory Makefile.rules:81: dset.d: No such file or directory Makefile.rules:81: error.d: No such file or directory Makefile.rules:81: fifo_server.d: No such file or directory Makefile.rules:81: flags.d: No such file or directory Makefile.rules:81: forward.d: No such file or directory Makefile.rules:81: hash_func.d: No such file or directory Makefile.rules:81: ip_addr.d: No such file or directory Makefile.rules:81: main.d: No such file or directory Makefile.rules:81: md5.d: No such file or directory Makefile.rules:81: md5utils.d: No such file or directory Makefile.rules:81: modparam.d: No such file or directory Makefile.rules:81: msg_translator.d: No such file or directory Makefile.rules:81: pass_fd.d: No such file or directory Makefile.rules:81: proxy.d: No such file or directory Makefile.rules:81: qvalue.d: No such file or directory Makefile.rules:81: re.d: No such file or directory Makefile.rules:81: receive.d: No such file or directory Makefile.rules:81: resolve.d: No such file or directory Makefile.rules:81: route.d: No such file or directory Makefile.rules:81: route_struct.d: No such file or directory Makefile.rules:81: script_cb.d: No such file or directory Makefile.rules:81: socket_info.d: No such file or directory Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

3 2

[Serusers] uac module for ser-0.9.0
by Mohammad Khan 22 May '05

22 May '05

Hello, would anybody please tell me where can I get right version of uac module for ser-0.9.0. I downoaded cvs snapshot of ser-0.10.99 can I use uac of this snapshot on ser-0.9.0? Mohammad

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Re: [Serusers] javarockx (Paul) 's ser.cfg
by Dave 22 May '05

22 May '05

Thank you much !! --- Java Rockx <javarockx(a)gmail.com> wrote: > Sorry all for the repost but the listserv scrubbed > my attachment. Here it is > again. > > P > > # ------------- Initial global variables > > debug=3 > fork=yes > log_stderror=no > > listen=10.3.0.221 <http://10.3.0.221> > port=5060 > children=4 > server_signature=no > > # hostname matching an alias will satisfy the > condition uri==myself" > alias=sip.mycompany.com <http://sip.mycompany.com> > > # dns - Uses dns to check if it is necessary to add > a "received=" field > # to a via. Default is no. > # rev_dns - Same as dns but use reverse DNS. > dns=no > rev_dns=no > > # check_via - Turn on or off Via host checking when > forwarding replies. > # Default is no. arcane. looks for discrepancy > between name and > # ip address when forwarding replies. > check_via=no > > # syn_branch - Shall the server use stateful synonym > branches? It is > # faster but not reboot-safe. Default is yes. > syn_branch=yes > > # memlog - Debugging level for final memory > statistics report. Default > # is L_DBG -- memory statistics are dumped only if > debug is set high. > memlog=1000 > > # sip_warning - Should replies include extensive > warnings? > # By default yes, it is good for trouble-shooting. > sip_warning=no > > # fifo - FIFO special file pathname > fifo_mode=0666 > fifo="/tmp/ser_fifo" > fifo_db_url="mysql://ser@localhost/ser" > > # reply_to_via - A hint to reply modules whether > they should send reply > # to IP advertised in Via. Turned off by default, > which means that > # replies are sent to IP address from which requests > came. > reply_to_via=no > > # user | uid - uid to be used by the server. 99 = > nobody. > #uid="nobody" > > # group | gid - gid to be used by the server. 99 = > nobody. > #gid="nobody" > > # mhomed -- enable calculation of outbound > interface. > # useful on multihomed servers. > mhomed=0 > > # ------------- external 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/auth.so" > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > loadmodule "/usr/local/lib/ser/modules/acc.so" > loadmodule "/usr/local/lib/ser/modules/group.so" > loadmodule "/usr/local/lib/ser/modules/print.so" > loadmodule "/usr/local/lib/ser/modules/textops.so" > loadmodule > "/usr/local/lib/ser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > loadmodule "/usr/local/lib/ser/modules/uri.so" > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > loadmodule "/usr/local/lib/ser/modules/domain.so" > loadmodule "/usr/local/lib/ser/modules/speeddial.so" > loadmodule "/usr/local/lib/ser/modules/options.so" > loadmodule "/usr/local/lib/ser/modules/avpops.so" > loadmodule > "/usr/local/lib/ser/modules/permissions.so" > loadmodule "/usr/local/lib/ser/modules/xdz_tools.so" > > # ------------ nathelper settings > modparam("nathelper", "rtpproxy_disable", 1) > modparam("nathelper", "natping_interval", 0) > > # ------------ media proxy settings > modparam("mediaproxy", "natping_interval", 30) > modparam("mediaproxy", "mediaproxy_socket", > "/var/run/mediaproxy.sock") > modparam("mediaproxy", "sip_asymmetrics", > "/usr/local/etc/ser/sip-asymmetric-clients") > modparam("mediaproxy", "rtp_asymmetrics", > "/usr/local/etc/ser/rtp-asymmetric-clients") > > # ------------- db_url setting > modparam("acc|auth_db|domain|group|lcr|permissions|speeddial|uri_db|usrloc|xdz_tools", > "db_url", "mysql://ser@localhost/ser") > > # ------------- use_domain setting > modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", > "use_domain", 0) > > # ------------- permissions settings > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > > # ------------- accounting parameters > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 1) > modparam("acc", "log_fmt", "cdfimorstup") > modparam("acc", "log_level", 1) > modparam("acc", "failed_transactions", 1) > modparam("acc", "report_cancels", 1) > modparam("acc", "report_ack", 0) > > # allows clear text passwords in the mysql database > modparam("auth_db", "calculate_ha1", 1) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "avps_column_int", "domn") > modparam("auth_db", "avps_column_str", "uuid") > > # ------------- domain parameters > modparam("domain", "db_mode", 1) > > # ------------- registration parameters > modparam("registrar", "nat_flag", 6) > modparam("registrar", "min_expires", 300) > modparam("registrar", "max_expires", 3600) > modparam("registrar", "default_expires", 600) > modparam("registrar", "default_q", 100) > modparam("registrar", "desc_time_order", 1) > modparam("registrar", "append_branches", 0) > > # ------------- rr parameters > modparam("rr", "enable_full_lr", 0) > > # ------------- tm parameters > modparam("tm", "fr_timer", 22) > modparam("tm", "fr_inv_timer", 27) > modparam("tm", "wt_timer", 10) > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > > # ------------- usrloc parameters > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "timer_interval", 30) > modparam("usrloc", "desc_time_order", 1) > > # -------------- AVPOPS parameters > modparam("avpops", "avp_url", > "mysql://ser@localhost/ser") > modparam("avpops", "avp_table", "usr_preferences") > modparam("avpops", "avp_aliases", "voicemail=i:500") > > route { > > # > ------------------------------------------------------------------------ > # Sanity Check Section > # > ------------------------------------------------------------------------ > if (!mf_process_maxfwd_header("10")) { > log(1, "LOOP DETECTED"); > sl_send_reply("483", "Too Many Hops"); > break; > }; > > if (msg:len > max_len) { > log(1, "MESSAGE OVERFLOW"); > sl_send_reply("513", "Message Too Large"); > break; > }; > > # > ------------------------------------------------------------------------ > # Click2Dial Test > # > ------------------------------------------------------------------------ > if (src_ip==myself && (search("^From: <sip:700@") || > search("^To: === message truncated === __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] javarockx (Paul) 's ser.cfg
by Dave 22 May '05

22 May '05

Gentlemen: I saw a posting from Javarockx http://lists.iptel.org/pipermail/serusers/2005-February/015687.html mentioning that he wants to repost his reworked ser.cfg. My search did not give me any positive results. Can someone (Paul?) please email it to the list? Thanks, Dave __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250

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Re: [Serusers] Advice needed
by Michael Ulitskiy 21 May '05

21 May '05

Can it do what I described? Documentation isn't very good on that web site, but according to what I saw I doubt it. Also what does it mean SER based PBX? SER+voicemail? Thanks, Michael On Friday 20 May 2005 07:20 pm, you wrote: > > we have an all in one ser solution that might be usefull for you > http://www.wifi.com.ar/english/voip.html > > regards, > > ____________________________________________________________________________ > > Jaime Garcia Ghirelli http://www.brujula.net > jaime(a)fonosip.com http://www.fonosip.com > http://www.wifi.com.ar > > ---------- Forwarded message ---------- > Date: Fri, 20 May 2005 15:49:35 -0400 > From: Michael Ulitskiy <mdu113(a)acedsl.com> > To: serusers(a)lists.iptel.org > Subject: [Serusers] Advice needed > > Hello, > > I'd like ask for advice on what is in your opinion the best solution > in the following scenario. > I have a bunch of sip servers (asterisk boxes as my users need pbx > functionality) that can make sip call to each other and my PSTN > gateway. Now I want to purchase PSTN terminitaion in several > different markets (and probably more in the future). All those > terminations will require authentication. > I want all my boxes when they see non-local call to send it to a > central routing server that would determine where this call should > be sent and authenticate to the appropriate provider so that I don't > have to configure all credentials on all asterisk boxes. Also I want > it not to deal with the media at all. All media streams should go directly > from asterisk box to the PSTN termination provider. > So basically it should be central SIP router that is able to authenticate > calls if neccessary. > I thought I could do it with SER and its UAC module, but it appears > UAC module doesn't work and probably won't work (see my previous > post in this list about UAC backport to 0.9.0). > Also I don't want to use asterisk in this place as asterisk always wants to > stay in media path and I'd really like to avoid of getting into hassle with > re-invites. > So the question is what are my options and what you would advice > as a solution. Are there any software out there that can do it (preferably > open-source, of course) or what else you could suggest to do to get > desired results. > Thanks a lot, > -- > See you later, > Michael > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > -- See you later, Michael

2 3

[Serusers] is serweb mature enough ?
by Lucas Aimaretto 21 May '05

21 May '05

Hi all, Is anybody using serweb ? Is it mature ?? Do you guys recomend using it ? What can I do with serweb that I can not do with the console ? And what can I do with the console that i can not with serweb ? best regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.13 - Release Date: 19/05/2005

2 1

[Serusers] semget failed
by Jorge Crichigno 21 May '05

21 May '05

Hello, I have a problem when starting SER 0.9.0 an a linux ppc64. The trace is as follow: ... unixsock_register_cmd: New command (t_uac_cancel) registered 0(0) unixsock_register_cmd: New command (t_hash) registered 0(0) unixsock_register_cmd: New command (t_reply) registered 0(0) DEBUG: lock_initialize: lock initialization started 0(0) ERROR: lock_set_init (SYSV): semget (..., 251, 0700) failed: Invalid argument 0(0) DEBUG: lock_initialize: entry semaphore initialization failure: Invalid argument 0(0) DEBUG: lock_initialize: first time semaphore allocation failure 0(0) INFO: semaphore arrays of size 250 allocated ... This error was already reported on april, 2005 and Jan Janak suggested changing SEMMSL value in include files. I did that, but the error continues... Do someone know something about this problem? Thank you!

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[Serusers] Solaris 10 box with SER 0.8.12
by ivan veizaga 21 May '05

21 May '05

Hello Can anybody please help me with the following problem: I have a Solaris 10 (x86) box with SER 0.8.12 preinstalled by default, when I run the SER with the default configuration file (ser.cfg.sample) everything works well (even with MySQL for authentication), but when I try to load the acc.so module appears the following error: # Mar 31 16:48:17 enterprise ser[14579]: ERROR: load_module: could not open module </usr/sfw/lib/ser/modules/acc.so>: ld.so.1: /usr/sfw/sbin/ser: fatal: relocation error: file /usr/sfw/lib/ser/modules/acc.so: symbol set_timer: referenced symbol not found Mar 31 16:48:17 enterprise ser[14579]: parse error (87,13-45): failed to load module Thank you very much for your time. Sorry if my english is bad. Regards. Ivan. _________________________________________________________ Do You Yahoo!? Información de Estados Unidos y América Latina, en Yahoo! Noticias. Visítanos en http://noticias.espanol.yahoo.com

2 1

[Serusers] exec module question
by Daniel Corbe 21 May '05

21 May '05

Hello, I've noticed that 90% of my SIP traffic is REGISTER requests. I am trying to implement some sort of caching mechanism (which will inherently be faster than doing a RADIUS lookup on every REGISTER request) using exec and a perl script which does a lookup from a memory-cached file. I'm using exec_msg() to call the perl script; however it is not behaving as I would expect. Here's the code snippet from my SER config. if (exec_msg("/usr/local/bin/check_register.pl")) { break; } else { if (!radius_www_authorize("")) { I am just in preliminary testing stage so I have the perl script merely exiting with 0 status. exec documentation is not clear as to whether it uses the shell return codes to determine the exit status of exec_msg If I am exit(0)ing from the perl script I would expect the if( to fail, not break and do the radius_www_authorize. Is that not correct? Thanks for the help! Regards, Daniel Corbe

3 2

[Serusers] VoIp Billing Solution SIP server compatible!!!
by Ollie Gejevski 21 May '05

21 May '05

Hey guys, I notice quite a few people are looking for a Billing solution. Please contact me as I can help. We already have few companies from this board use our solution with great success. We are Cisco as well as SIP server ( SER, SNOM ) compatible. Check out our website HYPERLINK "http://www.cyneric.com"www.cyneric.com Please contact me HYPERLINK "mailto:ollie.g@cyneric.com"ollie.g(a)cyneric.com for more information. All the best Ollie -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.13 - Release Date: 5/19/2005

2 1

[Serusers] Postgres does not compile with GCC 4.0.x
by Boris 21 May '05

21 May '05

After downloading the latest SER from CVS.. .It seems when trying to compile the postgres module with GCC 4.0.x and Glibc 2.3.5, I get the following error gcc -fPIC -DPIC -g -O -funroll-loops -Wcast-align -Wall -DNAME='"ser"' -DVERSION='"0.10.99-dev6"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/pgsql/include -I/usr/include/postgresql -c dbase.c -o dbase.o dbase.c: In function 'db_init': dbase.c:245: error: invalid lvalue in assignment make: *** [dbase.o] Error 1 Any ideas on how to fix this? I looked at the source on line 245 and it seems that one line is trying to do multiple things. Any ideas on how to fix this?

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Re: [Serusers] Advice needed
by Greger V. Teigre 21 May '05

21 May '05

BTW, the UAC doc is here (and contact info for maintainers): http://www.voice-system.ro/docs/uac/ g-) Greger V. Teigre wrote: > I would say SER is what you need, except that you struggle with the > authentication. You have the following scenarios: > 1. PSTN termination with IP-based access control (easiest) > 2. PSTN termination with authentication of all INVITEs (yes, that's > the UAC module. You should contact the maintainer, Ramona-Elena > Modroiu about the status. I thought it was reported to work, but > haven't tried myself) > 3. PSTN termination with registration and authentication of REGISTER > (but not INVITEs). Use sipsak to generate a REGISTER for your box. > > #2 requires that all INVITEs are sent twice and is not a very good > option. I would seek out PSTN providers who will give you #1. > g-) > > > Michael Ulitskiy wrote: >> Hello, >> >> I'd like ask for advice on what is in your opinion the best solution >> in the following scenario. >> I have a bunch of sip servers (asterisk boxes as my users need pbx >> functionality) that can make sip call to each other and my PSTN >> gateway. Now I want to purchase PSTN terminitaion in several >> different markets (and probably more in the future). All those >> terminations will require authentication. >> I want all my boxes when they see non-local call to send it to a >> central routing server that would determine where this call should >> be sent and authenticate to the appropriate provider so that I don't >> have to configure all credentials on all asterisk boxes. Also I want >> it not to deal with the media at all. All media streams should go >> directly from asterisk box to the PSTN termination provider. >> So basically it should be central SIP router that is able to >> authenticate calls if neccessary. >> I thought I could do it with SER and its UAC module, but it appears >> UAC module doesn't work and probably won't work (see my previous >> post in this list about UAC backport to 0.9.0). >> Also I don't want to use asterisk in this place as asterisk always >> wants to stay in media path and I'd really like to avoid of getting >> into hassle with re-invites. >> So the question is what are my options and what you would advice >> as a solution. Are there any software out there that can do it >> (preferably open-source, of course) or what else you could suggest to >> do to get desired results. >> Thanks a lot,

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[Serusers] step by step ser 0.9.0 Please Help.... Urgent...
by Bharat More 21 May '05

21 May '05

Hi All ... I want to install ser 0.9.0 MYSQL and redius for AAA purpose.Can anybody tell me step by step workout for this. Also the media proxy and the useragent addition....and then how to make the call from mediaproxy to desktop sip phone.... Thanks in advance for any hint... Bharat. ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

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