- sr-users - kamailio.org

[Serusers] Have some INVITE problems
by Alex 11 Oct '05

11 Oct '05

Hi all I would like to explain some problem. When my UAC initiates INVITE the server response with 100 trying After receiving 100 trying the UAC sending again INVITE. ( HERE is the problem). RFC - 3261 ("After receiving a 1xx response, any retransmissions cease altogether "); Any help on that will be appreciated. In addition if there any way to extend the timeout between the INVITES. Thanks

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[Users] [Fwd: [Sip-implementors] TLS certificate question]
by Klaus Darilion 11 Oct '05

11 Oct '05

Hi! I asked a question on the sip-implementors list. As nobody answered I post it here again, as it is also related to ser/openser. thanks klaus -------- Original Message -------- Subject: [Sip-implementors] TLS certificate question Date: Wed, 05 Oct 2005 09:10:05 +0200 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> To: SIP Implementors <sip-implementors(a)cs.columbia.edu> Hi! I'm trying to figure out how to make a certificate for a SIP proxy. RFC3262, section 4.1 states: For NAPTR records with SIPS protocol fields, (if the server is using a site certificate), the domain name in the query and the domain name in the replacement field MUST both be valid based on the site certificate handed out by the server in the TLS exchange. Similarly, the domain name in the SRV query and the domain name in the target in the SRV record MUST both be valid based on the same site certificate. Otherwise, an attacker could modify the DNS records to contain replacement values in a different domain, and the client could not validate that this was the desired behavior or the result of an attack. I'm not sure what the phrase "...MUST both be valid based on the site certificate" means. Does it mean that all possible domains must be present in the certificate? now imagine the follwing DNS lookups: sip:user@example.com ; order pref flags service regexp replacement IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.example.com. IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.example.com IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.example.com. _sips._tcp.example.com. That lookup would return: ;; Priority Weight Port Target IN SRV 0 1 5061 server1.example.com IN SRV 0 2 5061 server2.example.com _sip._tcp.example.com. That lookup would return: ;; Priority Weight Port Target IN SRV 0 1 5060 server1.example.com IN SRV 0 2 5060 server2.example.com _sip._udp.example.com. That lookup would return: ;; Priority Weight Port Target IN SRV 0 1 5060 server1.example.com IN SRV 0 2 5060 server2.example.com Finally, a TLS connection is made with server1.example.com. IF I understand RFC3263 correct, all of the above domains must be present in the certificate, but how to do this? 1. Should I put CN=example.com into the Subject and all other domains into the Subject Alternative Name? DNS=_sips._tcp.example.com. DNS=_sip._tcp.example.com DNS=_sip._udp.example.com. DNS=server1.example.com DNS=server2.example.com 2. Should I leave the Subject empty and put all domains into the Subject Alternative Name? 3. Why is it not sufficient to use only the domain "example.com" in the certificate (putting it into the subject field)? 4. Which SIP URIs should be used to check against the domains in the certificate (mutual proxy-proxy scenario)? Is it correct to check the domain in the request URI against the certificate of the receiving proxy, and check the domain in the From: URI against the certificate of the originating proxy? Thanks for any clarifications Klaus _______________________________________________ Sip-implementors mailing list Sip-implementors(a)cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

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[Serusers] help:I can't build auth_radius module
by shikun.wang 11 Oct '05

11 Oct '05

hi, I have install radiusclient-0.3.2,and comment the related exclude modules in the main Makefile.But when I build the whole SER,it outputs something as follows: ... make[1]: Entering directory `/usr/local/src/ser-0.9.3/modules/auth_radius' make[1]: Leaving directory `/usr/local/src/ser-0.9.3/modules/auth_radius' make[1]: Entering directory `/usr/local/src/ser-0.9.3/modules/auth_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.3"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -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 -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 -c authorize.c -o authorize.o gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.3"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -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 -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 -c authrad_mod.c -o authrad_mod.o make[1]: Leaving directory `/usr/local/src/ser-0.9.3/modules/auth_radius' ... It does NOT create an auth_radius.so file. Can anyone help me with some instructions?Thank you.

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[Serusers] msilo configuration
by Teemu Harju 11 Oct '05

11 Oct '05

Hi, I've been configuring msilo for offline messaging and I've faced few problems. First... I use Xten (or now Counterpath) Eyebeam and each time user starts to type an instant message it sends a SIP MESSAGE with Content-Type: application/im-iscomposing+xml and msilo naturally stores this message also. I tried to prevent this with the following configuration but it didn't work.. if (search("Content-Type: application/im-iscomposing+xml")) { # # Do not store composing MESSAGEs # } else { # # Store offline message # } Or course this is done only for method=="MESSAGE"... Can you see why this is not working? Second... Is there a way to make the reply to the offline message appear to be coming from the user that it was sent to? I mean you can configure msilo to reply for example with user registrar(a)myserproxy.com, but that opens a new IM window for the user so it's not that good. Thanks in advance... -- Teemu Harju http://www.teemuharju.net

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[Serusers] Radius authentication and accounting issues
by deepak.chandrasekaran＠wipro.com 11 Oct '05

11 Oct '05

Hi all, I am trying to provide radius authentication and accounting in ser. I am using freeradius. Everytime I try to start ser, it crashes. Can anybody suggest some possible reasons as to why this happens? I followed radius how-to doc to configure radius server and radiusclient-ng library. I hv attached herewith my ser.cfg file. Any help would be greatly appreciated. Thanks in advance. Regards, Deepak Ser.cfg: # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=no #log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode 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" loadmodule "/usr/local/lib/ser/modules/textops.so" # RADIUS support #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" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- acc params -- #modparam("acc", "radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") #modparam("acc", "radius_missed_flag", 2) #modparam("acc", "radius_flag", 1) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- auth_radius params -- modparam("auth_radius", "radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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 if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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") { # log(1, "REGISTER: Authenticating user\n"); # Uncomment this if you want to use digest authentication # if (!www_authorize("nerd.vngncoe.wipro.com", "subscriber")) { if (!radius_www_authorize("")) { log(1, "REGISTER: challenging user\n"); www_challenge("", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

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[Serusers] how to add the auth_radius module into SER
by shikun.wang 11 Oct '05

11 Oct '05

Hi,all I want to make my ser support radius accounting,so I add: loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" into my ser.cfg,and uncomment DEFS+=-DRAD_ACC LIBS=-L$(LOCALBASE)/lib -lradiusclient in modules/acc/Makefile. But after I make the whole SER and find no auth_radius.so file under /usr/local/lib/ser/modules,neither can I start the SER for it can't load this module. Is there any thing I have not set? Any ideas?Thank you.:)

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[Serusers] CDRTool
by Cameron Beattie 11 Oct '05

11 Oct '05

I have been testing CDRTool and get the following message in the log files when browsing to www.example.com/scripts/ratingEngine.php: Cannot instantiate non-existent class: ratingtables in /var/www/CDRTool/scripts/ratingEngine.php Is this because the non commercial version of CDRTool does not provide rating or is it something I have done wrong in the setup? Any advice would be appreciated. Regards Cameron

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Re: [Serusers] denying call based on country codes
by Nhadie 10 Oct '05

10 Oct '05

Thank you for your reply. Unfortunately I send all pstn traffic to a single gateway provider. So what I really need is for ser to check if this user is in the 36countries group, and it dialled a number outside of the 36 countries then denied. I was thinking of using multiple if URI but i don't want to define a lot IF statements. Hope someone here can share something that they were able to do. > > > > > > > > > > > > > > > > > Sure... it's possible and actually pretty easy, dependent on how things > are set up. Assuming, for the sake of argument, that you have two servers > that handle PSTN calls and one handles calls to anywhere and one handles > calls to only 36 countries: > > > > I'd use the grp table for this sort of thing (although some people would > use AVPops). I'd just add the users to a particular kind of group for all > access.... > > > > In MySQL, you'd want to do something like: > > > > insert into grp > values('&lt;username&gt;','&lt;domainname&gt;','fullaccess',now()); > > > > > > > > Then, set a flag if the user is in that group... > > > > if(is_user_in(&quot;Request-URI&quot;, &quot;fullaccess&quot;)) > > { > > setflag(19); > > }; > > > > > > Then... somewhere in an INVITE check block (or wherever, really), check > for the URI and the flag. > > > > if (uri =~ &quot;^sip:00*&quot;) > > { > > if(isflagset(19)) > > { > > log(1, &quot;request for alternate 800 provider > receivedn&quot;); > > rewritehostport(&quot;10.1.1.1:5060&quot;); # PSTN for ALL > numbers > > forward(uri:host, uri:port); > > route(1); > > break; > > } > > else > > { > > log(1, &quot;request for alternate 800 provider > receivedn&quot;); > > > > rewritehostport(&quot;10.1.1.2:5060&quot;); # PSTN for only > 36 countries > > > > forward(uri:host, uri:port); > > > > route(1); > > > > break; > > }; > > > > }; > > > > > > > > Now, the trick comes when you DON'T have PSTN server separate for > different countries... > > > > THEN you might have to do some very in-depth rules based on the format of > the URI to the 36 countries in question which really isn't THAT bad, but > dependent on their numeric format for numbers, it could get annoying to > specify each one (especially if they have differing formats for land line > and mobile numbers). It's not undoable, though. > > > > N. > > > > > > On Mon, 10 Oct 2005 20:17:50 +0800, Nhadie wrote > > &gt; Hi All, > > &gt; > > &gt; I'd like to setup to groups, one that can call only around 36 > > countries and one that can call anywhere. Currently, when I see that the > URI > > starts with 00 i'll check if it belongs to the world-pstn group then it > can > > dial-out if not it's blocked. Is what I'm thinking > > possible? > > &gt; > > &gt; TIA > > &gt; > > &gt; Regards, > > &gt; Nhadie > > > > > > > > > > > > > > > > > > > > > > > > >

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RE: [Serusers] SER Calls Per Second
by Martin, Peter 10 Oct '05

10 Oct '05

Let's say, just sip proxy... Regards, Peter Martin -----Original Message----- From: Daryl Sanders [mailto:daryl.sanders@gmail.com] Sent: Monday, October 10, 2005 3:35 PM To: Martin, Peter Subject: Re: [Serusers] SER Calls Per Second This depends on if you will be proxying the media stream or if you are just handling call setup with SER. If you plan to carry the media you will need to use rtpproxy or mediaproxy along with SER. It will also matter if you are using SER as the registrar on the same box or just routing calls. In addition, the specs of the box/boxes you'll be using will have a huge effect on the number of calls. - Daryl On 10/10/05, Martin, Peter <Peter.Martin(a)ipc.com> wrote: > > > Anyone know what is the most calls per second that has been tested > > and verified using SER? > > > > Peter Martin > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Users] TLS setup
by Bogdan-Andrei Iancu 10 Oct '05

10 Oct '05

Hi everybody, finally the TLS proxy is up and running - just to remember what was about -> http://www.openser.org/pipermail/users/2005-September/000953.html The configuration is quite a simple one - it accepts exclusively requests over TLS and does relay based on RURI (keeping the TLS protocol). server info: openser.org port 5061 - SRV record is also present. The idea is to interconnect several domains via this TLS relay. If one of the interconnected servers has to deliver a request to another domain via TLS, it has just to relay it to the TLS proxy. Note that the proxy denies any local requests. How to do the redirect from your script? Before doing t_relay(), you may add something like: if ( uri=~".*(a)domain1.com" || uri=~".*(a)domain2.com") { avp_write("sip:openser.org;transport=tls","i:11"); avp_pushto("$duri","i:11"); } via DST_URI, the TLS protocol will be forced without affecting the RURI. Normally this logic should be totally transparent via NAPTR lookup, but we do not have it in OpenSER yet. Now, about the TLS part. The TLS proxy has both tls_verify and tls_require_certificate enabled. For certificates there are two ways: 1) generate your own root CA and certificate and set the CA to me to added to the proxy (tools available on CVS in tls/tools) 2) ask and I will provide a certificate signed with the proxy root CA. TLS configuration on your side: tls_certificate="path/cert.pem" tls_private_key="path/privkey.pem" tls_ca_list="path/calist.pem" tls_verify=on tls_require_certificate=on listen=tls:xxx.xxx.xxx.xxx tls_port_no=5061 For any other additional hints about TLS, please see the tutorial http://www.openser.org/docs/tls People interested in interconnection, please join (just let me know which way you prefer for certificates). First domain connected to the proxy is siphub.net - it's a local platform we are using. regards, bogdan PS: if there are people interested in testing SIP UA with TLS, let me know and I can add registrar support. Also, if somebody has ideas how to enhance the proxy config, feel free....

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RE: [Serusers] RPM's of recent SER versions
by Elad Dotan 10 Oct '05

10 Oct '05

Try from this site: http://yum.sip-router.org/fedora-4/ Elad. -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Yair Hakak Sent: Monday, October 10, 2005 10:27 PM To: serusers(a)lists.iptel.org Subject: [Serusers] RPM's of recent SER versions hello ser users list, where can i get RPM's for recent versions of SER? on berlios the last SER to have RPM's is 0.8.12. thanks, yair

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[Serusers] RPM's of recent SER versions
by Yair Hakak 10 Oct '05

10 Oct '05

hello ser users list, where can i get RPM's for recent versions of SER? on berlios the last SER to have RPM's is 0.8.12. thanks, yair

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[Serusers] SER Calls Per Second
by Martin, Peter 10 Oct '05

10 Oct '05

Anyone know what is the most calls per second that has been tested and verified using SER? Peter Martin

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[Users] tls_require_certificate
by Alexander Ph. Lintenhofer 10 Oct '05

10 Oct '05

Hi there, I have a question concerning TLS in openser: By switching tls_require_certificate to "on", the peer is forced to send his certificate for means of mutual authentication. My problem is, that the peer may be another proxy server whom I want to authenticate with its cert - but the peer might also be an user agent. In my situation I use a Snom 360 which has not the possibility to import an own user-certificate (only a CA-cert for verifying server-certs). ----------- ---------- --------- | snom 360 | <------ TLS -------> | outbound | <----- TLS -----> | inbound | ----------- server sends cert ---------- mutual AUTH --------- But when I activate tls_require_certificate=on in the openser.cfg of the outbound proxy, the snom360 can't register, because it has no user-cert. On the other hand, when I disable tls_require_certificate, the snom can register, but the security between the proxies is weak. Is there an appropriate solution for this problem ?? Maybe I didn't understand the sample configuration at all.... Thanks in advance and regards, Philipp

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[Serusers] Calling other SIP users
by Voipers Portugal 10 Oct '05

10 Oct '05

Hi, I have my SER working when i try to talk to other SIP servers. However, i cannot make calls inside my own network. I am using LDAP to authenticate my users, and I have sipldap script that circulates in the web. Everything works fine until there, i get the following output when the user exists: sip:username@host isn´t this what was supposed to happen? When i then try to (!lookup("location")) { if (!exec_dset("/etc/ser/sipldap")) { sl_send_reply("404", "Not Found"); break; } else { log(1," sipldap call"); }; }; #!lookup I get the Not Found message. But if i call the following way: username@username_IP the call works fine.. any ideia? Jose Simoes

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[Serusers] check_from
by Jason Mohler 10 Oct '05

10 Oct '05

I have a spoofed call that keeps coming in with the ANI of 000000000. How can I use check_from() and is it comes in as 000000000 to just drop the call so it does not take up my resources. Is below a reliable synatx for this? if (method==INVITE) && (!check_from()){ break; };

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[Users] Have openser make changes based on message header
by Brandon Price 10 Oct '05

10 Oct '05

First I want to thank everyone thats helped me. Here is what I am working on today. I have a set of sipura 2002, and linkys pap2 sip ata's. When you dial *67 (cid block) it tells the adapter to return a dialtone and modify the sip message header in such a way the SIP Display info is changed to "Anonymous". This doesn't set the privacy=full flag in the rpid and thus the cid is passed to the terminating end. I have tried having openser perform an action based on if from_uri =~ Anonymous but that apparently only matches to the SIP from address: not, SIP Display info. Below I have a sample bit of sip messaging.... My question is, is there any way to have openser look inside the packet and do something based on "SIP Display info"? Can I do this w / AVP's have it pull that var form the sip messaging so I can work with it later? Or heck does anyone have a solution to make the ata's pass the privacy=full flag? Message Header Via: SIP/2.0/UDP 216.120.224.163:5060;branch=z9hG4bK-f4bd2420 From: Anonymous <sip: 15184782406(a)ser1.manhattan.vtnoc.net>;tag=d5d92a6937d9f838o0 SIP Display info: Anonymous SIP from address: sip:15184782406@ser1.manhattan.vtnoc.net SIP tag: d5d92a6937d9f838o0 To: <sip:13143212222@ser1.manhattan.vtnoc.net> SIP to address: sip:13143212222@ser1.manhattan.vtnoc.net Call-ID: bae3e6fd-a92244b4@localhost CSeq: 101 INVITE Max-Forwards: 70

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[Serusers] ser replying at udp source port instead of via-port
by Atif Rasheed 10 Oct '05

10 Oct '05

Hello all, I am using ser 0.9.4 + rtpproxy. now when our SSW INVITEs SER, SER sends 100-trying to the udp-source-port where SER should reply with connection information in via or contact headers. (probably via header) Please comment. Thank you -- Atif

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[Serusers] SIp auto dialer
by Iqbal 10 Oct '05

10 Oct '05

Hi Does anyone have any recommendations, linux one is better, and would sipsak do this. All I want is a number to be dialled, and then handup once answered, no voice required. This is not for spam, just to allow me to dial up random numbers worldwide, from which I want to plot a time to completion graph, from where I can then hopfully automatically change my GW in lcr, for best route. Iqbal

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[Serusers] SER 0.9.4/ postgresql Registration problems
by H Quintana 10 Oct '05

10 Oct '05

Hi, I just updated SER from 0.9.3 to 0.9.4. In both cases I'm using Postgresql. With SER 0.9.3 the REGISTER process works fine, the REGISTER is accepted and then the record is inserted in DB. With SER 0.9.4, I've found that when the "location" is inserted or updated in the DB, SER is generating core. However when SER starts the locations list is read from DB without any problem. Any ideas? Thanks for your help, Humberto __________________________________ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs

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[Users] REGISTER expires and replication
by Helge Waastad 10 Oct '05

10 Oct '05

Hi, I have a scenario in which I have a little bit of confusion.. I have a redundant setup, two OpenSER, a MySQL cluster and replication of REGISTERs. The UAC sends expires=3600 Both OpenSER's has a max_expires=1800, min_expires=300,default_expires=300 My Sipura-1001 has default expires=3600. In the OK message sent from the OpenSER handeling the REGISTER, expires=1800 is sent back to the UAC which is OK (However the Sipura does not obey this, which also is struggeling, but that is another story....) The replicated REGISTER, hovever, uses the original expiery from the UAC...expires=3600 Do I have to rewrite the expires in the replicated message? -- mvh/best regards Helge Waastad System Engineer Smartnet

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[Serusers] SER, and the two firewall problem
by Richard Southwick 10 Oct '05

10 Oct '05

Hi all -- We're using SER as a registrar and proxy, and need to deal with the following NAT problem: * SER is proxying to asterisk, which is behind a firewall. * Our users are registering with SER, but their UAs are behind another firewall. We're looking at nathelper and rtpproxy. Two questions: * Will this combo (SER, nathelper, rtpproxy) work in this scenario to solve the NAT and pinhole issues? * Does SER have to live outside the firewall, or is there a way it can be behind the same firewall as Asterisk? Thanks for any insight, Richard

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[Serusers] [Fwd: [Sip-implementors] TLS certificate question]
by Adrian Georgescu 10 Oct '05

10 Oct '05

Juha, I host for example multiple domains for multiple users. This means that users rely on my services entirely to provide them with identity services. Ideally I would like to present the Certificate of domain X from user in domain X when I forward his calls through my proxy. Alternatively I could say I put certificate with CN=ag-projects.com in all request going outside my domains but then my white-label customers will become yellow. So something to match business with the technicalities is required, don't you think? Adrian > But then, the whole authorization thing would be nonsens. > > Just imagine a host named "sip.badguy.com". This host has a valid > certificate for its hostname. Then, this SIP proxy sends a SIP request > with the header: > From: "Klaus Darilion" <sip:klaus at darilion.com> > > Now, what is the receiving proxy interested in? Does it want to validate > the host or the sender (From header)? there are other ietf means to validate the sender. usually they involve signing of from uri with the certificate of its domain. see for example draft-rosenberg-sip-identity-privacy-00 for a good summary of the issues and problems involved. in proxy-to-proxy case, all that needs to be done is to validate the remote proxy. > IMO, I want to authenticate the sender in the From header. Thus, the > certificate would have to match the SIP domain, and not the host > name. see above. -- juha

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[Serusers] Conferencing...
by Voipers Portugal 10 Oct '05

10 Oct '05

As far as I have seen, SER itself cannot offer conferencing services, right? Therefore I have also noticed that you recommend Asterisk, SEMS or others. My doubt is, how does SER contacts with those servers? Does SER supports MGCP? I tought not, but if this is true, how do they communicate? If anyone knows where i can see more info on SER conferencing, please let me know. Regards, Jose Simoes

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[Serusers] time stamping each SIP message in a dialogue
by Iqbal 10 Oct '05

10 Oct '05

Hi I just realised that the date: parameters is not set in each SIP message, am sure this is correct, but wouldnt this be a good idea, I am trying to work out which part of my call is being delayed, and the date command would make it easier. Just compare times between each message in ngrep. Something like what mailservers do, so that you can see if the mail was on a queue for a long time. Iqbal

1 0

[Serusers] denying call based on country codes
by Nhadie 10 Oct '05

10 Oct '05

3 3

[Serusers] lcr module based on timeout
by Iqbal 10 Oct '05

10 Oct '05

Hi Can the lcr module be deployed in such as way where, if call takes x secs to complete it will go elsewhere..or perhaps even before the invite is sent, a RTT to the gateway is done, and it picks the nearest/best gateway for that prefix Iqbal

1 0

[Serusers] Testing
by Pavol Segeč 10 Oct '05

10 Oct '05

Hi all, Is there some methodology and test tools for doing load testing (performance capabilities) of SIP Servers? Thanks pavol Sorry, for this off topic.

2 1

[Users] caller id authentication
by jayesh nambiar 10 Oct '05

10 Oct '05

hi all, I am trying to implement this feature in my openser 0.10. It is as follows: I allow users in pstn network to dial my gateway number, wherein they get a dial tone and the gateway then forwards all the calls to openser and are processed accordingly. But now I want to authenticate these caller ids. I want to give this privilege only to my subscribers. I had thought of adding the ids in the trusted table as my gateway is in trusted table, but this does not seem to be a good solution. Can I add an extra column in the subscriber table wherin I mention the caller-id for individual users (I give only one caller_id to one user), so that if the user is authenticated then the caller- id for that user should be authenticated. Can it be possible. I was thinkin of using AVPs for this, but I dont get the idea of retreiving the caller-ids from the table to check if the caller-id is valid. Can someone help me in this regard. thanks jayesh --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

2 1

RE: [Serusers] ser web
by Elad Dotan 10 Oct '05

10 Oct '05

Hi, Thank you for your response. I'm now able to see the serweb login screen. When I put the username (admin) and password (heslo) I get a "bad username or password" error Message. Could you please assist. Thanks, Elad. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Friday, September 30, 2005 1:02 PM To: Jan Janak Cc: Elad Dotan; serusers(a)lists.iptel.org Subject: Re: [Serusers] ser web And watch the apache log files. There you will see the http requests and why apache can't find them. regards klaus Jan Janak wrote: > SERWeb should be accessible as http://<ip_of_the_host>/serweb. > > If that returns 404 then you need to create /serweb alias in Apache > configuration. You can do so by including /etc/serweb/apache.conf file > in the main Apache configuration file. > > Jan. > > On 29-09-2005 16:40, Elad Dotan wrote: > >>Hi, >>I've installed 4 RPM files on a Fedora CORE 3. >>The RPM are: >> >>ser-0.9.3-0.i386.rpm >>php-pecl-classkit-0.4-2.i386.rpm >>ser-mysql-0.9.3-0.i386.rpm >>serweb-0.9.3-0.i386.rpm >>I enabled the httpd and mysql service. >>I'm not sure what to do next. >>All the Documentation I have is related to source flies and not RPM. >>Ser is working but I don't know how to configure the serweb in order >>for it to work. >> >>I tried: http:// >>but I get an error. >> >>where can I find documentation about ser web? >>Thanks, >>Elad. >> > > >>_______________________________________________ >>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 > >

2 1

[Serusers] PSTN through CISCO
by share phone 10 Oct '05

10 Oct '05

Hi All, I have a question how to use PSTN through cisco through openser.I am intrested in the security any help is appreciated. Do any one have the CDR Report through User Interface? Regards, karun. --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

1 0

[Serusers] how to keep passwords encrypted in radchek table ??
by tulay sazak 10 Oct '05

10 Oct '05

Hello EveryBody, I am using SER version 0.9.3 and freeRADIUS version 1.0.4. I am trying to keep the user passwords as encrypted in freeRadius DB which is mySql table radcheck. When I run the Radius Request I got: auth: No User-Password or CHAP-Password attribute in the request auth: Failed to validate the user. What should I do to solve this problem? 1) in radiusd.conf ? 2) What should be the value of "Auth-Type" parameter in radcheck or radgroupchek tables ? 3) What should be the attribute for password in radcheck table ? Is it "User-Password" or "Chap-Password" or "Crypt-Password" ?? I'll appreciate very much if anyone can help. Regards Tulay Sazak __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

2 1

[Serusers] registration problems...
by Olivier Taylor 10 Oct '05

10 Oct '05

hello, I have a registration problem. I use many different uacs(Zyxel,Grandstream ata486, budgettone100,sipura 3000, alcatel st280, eyebeam,....) Ser version is 0.9.4 on Freebsd 5.4 On ser.cfg, I have : modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 1200) modparam("registrar", "max_expires", 1800) and if (nat_uac_test("19")) { if (method == "REGISTER" || !search("^Record-Route:")) { fix_nated_contact(); force_rport(); etflag(6); }; }; The problem is that after a few ReRegistration, ser stop updates od the dababase even if ReRegistration continues to come... Ok, min_expires seems not Rfc compliant, but even without that I have the same behavior. I have seen many discussions about that and tried many solutions... The 2 greps shows register messages for 2 differents user agents Any good idea? ps: Greger, isn't it to telegraphic this time? ;) U 2005/10/09 14:37:37.460238 81.241.192.55:6938 -> 82.146.123.252:5060 REGISTER sip:finalcut.be SIP/2.0..To: 3227470340<sip:3227470340@finalcut.be>..From: 3227470340<sip:3227470340@finalcut.be>;tag=ab7b0c68..Via: SIP/2.0/UDP 192.168.2.101:6938 ;branch=z9hG4bK-d87543-505047371-1--d87543-;rport..Call-ID: 5e6f6777312d2659..CSeq: 13 REGISTER..Contact: <sip:3227470340@192.168.2.101:6938>..Expires: 3600..Max-Forwards: 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: eyeBeam release 3007n stamp 17816..Authorization: Digest username="322747 0340",realm="finalcut.be",nonce="4349103d8b8ae8fadbc161b5bb1a93a299fb4f4a",u ri="sip:finalcut.be",response="e6735467abe564030707db572d6f9f1f",algorithm=M D5..Content-Length: 0.... # U 2005/10/09 14:37:37.480571 82.146.123.252:5060 -> 81.241.192.55:6938 SIP/2.0 200 OK..To: 3227470340<sip:3227470340@finalcut.be>;tag=59f1565d6938a98ef3fc4707bf9b43aa. 1234..From: 3227470340<sip:3227470340@finalcut.be>;tag=ab7b0c68..Via: SIP/2. 0/UDP 192.168.2.101:6938;branch=z9hG4bK-d87543-505047371-1--d87543-;rport=6938;rec eived=81.241.192.55..Call-ID: 5e6f6777312d2659..CSeq: 13 REGISTER..Contact: <sip:322747034 ;expires=1800"> <mailto:0@81.241.192.55:6938> 0@81.241.192.55:6938>;expires=1800, <sip:3227470340@81.241.192.55:1048>;expires=119..Server: Sip EXpress router (0.9.4 (i386/freebsd))..Content-Length: 0..Warning: 392 82.1 46.123.252:5060 "Noisy feedback tells: pid=43104 req_src_ip=81.241.192.55 req_src_port=6938 in_uri=sip:finalcut.be out_uri=sip:finalcut.be via_cnt==1".... U 217.136.0.70:53284 -> 82.146.123.252:5060 REGISTER sip:82.146.123.252;user=phone SIP/2.0..Via: SIP/2.0/UDP 192.168.1.13:5060..From: 1300 <sip:1300@82.146.123.252;user=phone>..To: 1300 <sip:1300@82.146.123.252;user= phone>..Call-ID: <mailto:43478@192.168.1.13..Contact> 43478@192.168.1.13..Contact: <sip:1300@192.168.1.13:5060;user=phone>..CSeq: 18961 REGISTER..Max-Forwards: 70..Content-Length: 0..User-Agent: ST280 (REV-A2+ 2.439-1)..Expires: 1800..Authorization: Digest username="1300",realm="finalcut.be",nonce="43490c49911fa90e11538f52e07c19509 087b505",uri="sip:82.146.123.252",algorithm=MD5,r esponse="7ac5e0f3719872d4624a86c14dc677c8".... # U 82.146.123.252:5060 -> 217.136.0.70:53284 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.13:5060;rport=53284;received=217.136.0.70..From: 1300 <sip:1300@82.146.123.252;user=phone>..To: 1300 <sip:1300@82.146.123.252;use r=phone>;tag=59f1565d6938a98ef3fc4707bf9b43aa.1e42..Call-ID: <mailto:43478@192.168.1.13..CSeq> 43478@192.168.1.13..CSeq: 18961 REGISTER..Contact: <sip:1300@217.136.0.70:53284;user=phone>;expires=1800..Serve r: Sip EXpress router (0.9.4 (i386/freebsd))..Content-Length: 0..Warning: 392 82.146.123.252:5060 "Noisy feedback tells: pid=43111 req_src_ip=217.136.0.70 req_src_port=532 84 in_uri=sip:82.146.123.252;user=phone out_uri=sip:82.146.123.252;user=phone via_cnt==1".... #

2 1

[Serusers] test mail
by Sam Lee 10 Oct '05

10 Oct '05

please ignore

1 0

[Serusers] testmail
by Sam Lee 10 Oct '05

10 Oct '05

please ignore

1 0

[Serusers] Mediaproxy must be with SER?
by Ryan Pagquil 10 Oct '05

10 Oct '05

Hi, Should I also install SER for each mediaproxy installation for balancing media traffic? I'm planning to have multiple installation of media proxy located in some part of the world to divert traffic and load balance the media traffic. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Users] Billing System
by harry gaillac 09 Oct '05

09 Oct '05

Hello, Somebody has tested freeSide billing system !? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

5 13

[Serusers] SER Codecs/Web Admin
by gorand＠dvvti.com 09 Oct '05

09 Oct '05

Can anyone answer these two questions for me. Is there a web admin which similar to pingtel or Asterisk that allows for easier admin for SER Server? Question 2. What codec formats is supported by ser?

2 1

[Users] Dumb question - can I use Oracle instead of MySQL
by Juan Carlos Castro y Castro 09 Oct '05

09 Oct '05

for the "openser" database? Perhaps via unixODBC? Thx, Juan

2 3

[Serusers] Asigning a Subscriber to a Specific rtpproxy
by Andres 09 Oct '05

09 Oct '05

Hi, Can anybody think of a way to assign a subscriber's call to a specific rtpproxy? The idea is to have one main SER server but multiple rtpproxy servers handling the media stream close to where the subs are. The SER server can be a continent away but the rtpproxy is in the same city as the users. I was thinking of using ACL (groups) somehow but not sure if it is at all possible. Any ideas? Thanks, -- Andres Network Admin http://www.telesip.net

3 4

[Users] tls_require_certificate - sorry!
by Alexander Ph. Lintenhofer 08 Oct '05

08 Oct '05

----------- ---------- --------- | snom 360 | <------ TLS -------> | outbound | <----- TLS -----> | inbound | ----------- server sends cert ---------- mutual AUTH ---------

1 0

[Users] tls_require_certificate - 2nd try
by Alexander Ph. Lintenhofer 08 Oct '05

08 Oct '05

Sorry for the bad ASCII-graphic... Another try.... ----------- ---------- --------- | snom 360 | <------ TLS -------> | outbound | <----- TLS -----> | inbound | ----------- server sends cert ---------- mutual AUTH --------- Philipp

1 0

[Serusers] Multiple SER concept
by Ryan Pagquil 08 Oct '05

08 Oct '05

Hi Guys, How does multiple SER deployed strategically in some part of the world works and what does it do with users that are registering with the same domain for all those SER's? How does calls look like when this was implemented? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Serusers] Shoot my logic full of holes....
by sip 08 Oct '05

08 Oct '05

Okay... I'm wondering if this would work as a quick hack for checking call blocking (and making it rational). In order to use AVPOps, and not rewrite a module, I've created a table called callblock, which looks exactly like the usr_preferences table (basically an AVP table), but has no primary keys (as I want to be able to add multiple values for the same username/domain/attribute). In my code, I then have a block that looks like: if(avp_db_load("$ruri/username","s:callblkna/callblock")) { if(avp_check("s:callblkna","eq/$from/gi")) { sl_send_reply("603", "Call denied"); break; }; }; if(avp_db_load("$ruri/username","s:callblkvm/callblock")) { if(avp_check("s:callblkvm","eq/$from/gi")) { route(4); break; }; }; If it's a callblkna, it then sends an error reply. If it's a callblkvm, it does a route(4) (which is straight to voicemail). Anyone see a glaring hole in this logic here? It seems to work, but I'd hate to get down the road and discover that I overlooked something glaringly obvious (which I often do). N.

2 1

[Serusers] Can 'avp_check' check whether sourceIP and destinationIP are the same subnet segment?
by Zen Kato 08 Oct '05

08 Oct '05

If 2 UAs exist on the same network segment, e.g., 192.168.0.1 and 192.168.0.2, how to check whether sourceIP and destinationIP are the same subnet segment using avpops module? If 'ser' sits on 192.168.100.1, UA1 can send 'INVITE' then UA2 rings, but when we pick up UA2's handset, these two UAs goes to a panic. I have to reboot two UAs. So, I would like to 'rewritehostport' INVITE packets to 'man-in-the-middle-sip-server', 'asterisk' when UA1 INVITEs UA2. ser asterisk |.1 |.2 192.168.100.0/24 -------------------------------- |.254 router |.254 192.168.0.0/24 --------------------------------- |.1 |.2 UA1 UA2 Regards, Zen

1 0

[Serusers] Convert SIP-T to SIP
by Jorge Cuevas 07 Oct '05

07 Oct '05

Hi, I just found out about SER, and I cannot find info on SIP-T, just that it can proxy SIP-T calls (in an older message). I want to go beyond that and convert from SIP-T to SIP, so I can use it with asterisk or directly to cisco equipment. Does anyone know if SER can be used to receive calls in SIP-T and forward them as SIP ? Can it be done with no coding, little coding, a lot of coding? Any enlightment would be really apreciated. Thanks, Jorge

1 0

[Serusers] OT - VoIP TestDay
by Joao Pereira 07 Oct '05

07 Oct '05

Hello to all, We are installing SER in the Portuguese Universities, and next Monday (17:00 GMT) we will test the connectivity between the Universities, but also from the Universities to the rest of the world. Does someone knows lines around the world for us to call and test the communication? I googled it a lot and I just found one: The weather report in Australia: sip:+61262223590@aarnet.edu.au If someone knows of more lines like this one, or want to help us in the tests by just answering the calls, please contact me (maybe out of the list because this is a bit out of topic ). Thanks Joao Pereira www.fccn.pt

3 2

[Serusers] T38 Support
by Atif Rasheed 07 Oct '05

07 Oct '05

Hello all, can SER with RTP-proxy or Media-Proxy relay T38. If supported, do we need to do some extra configuration or it will work in standard configuation. Thank you, Atif

3 3

[Serusers] SIPATH Internet setup
by gorand＠dvvti.com 07 Oct '05

07 Oct '05

I am trying to set setup SER on SIPATH. I have a WRT54g v 3.0. It Flashed fine, and followed all the instructions up to where I need to wgetthe packages. The problem is that I don't have any connectivity with this device to the internet. I don't know the proper ip configurations. My network is a 172.16.0.0/16 Network This is the ifconfig from the router. What is the proper commands to get this guy to talk to the internet. Your help would be greatly appreciated. Thanks. br0 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5C inet addr:192.168.1.1 Bcast:192.168.1.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1381 errors:0 dropped:0 overruns:0 frame:0 TX packets:511 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:103850 (101.4 KiB) TX bytes:59305 (57.9 KiB) eth0 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5C UP BROADCAST RUNNING PROMISC MULTICAST MTU:1500 Metric:1 RX packets:5403 errors:0 dropped:0 overruns:0 frame:0 TX packets:378 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:100 RX bytes:756228 (738.5 KiB) TX bytes:86750 (84.7 KiB) Interrupt:5 Base address:0x2000 eth1 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5E UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1379 errors:0 dropped:0 overruns:0 frame:4094 TX packets:785 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:100 RX bytes:123144 (120.2 KiB) TX bytes:97726 (95.4 KiB) Interrupt:4 Base address:0x1000 lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:1 errors:0 dropped:0 overruns:0 frame:0 TX packets:1 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:76 (76.0 B) TX bytes:76 (76.0 B) vlan0 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5C UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:291 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 B) TX bytes:35072 (34.2 KiB) vlan1 Link encap:Ethernet HWaddr 00:13:10:8B:D4:5C UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:87 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 B) TX bytes:51678 (50.4 KiB)

1 0

[Users] RPID and how it relates to CallerID
by Brandon Price 07 Oct '05

07 Oct '05

Can someone please explain to me how RPID works under openser? I have some experience with SIP and Asterisk, but I am having some trouble manipulating the RPID variables under openser. How does RPID relate to Callerid or simple services like private calling, or anonymous call block. How do I tell openser to send all calls from a certain user with the privacy=full flag, I'm assuming this tells the sip proxy or UA on the other end not to present the callerid? Any help greatly appreciated. -Brandon Price-

3 8

[Serusers] Where to set T1 timer (RFC 3261)?
by John Paul Morrison 07 Oct '05

07 Oct '05

Where is the T1 timer set in Ser? The RFC suggests a default of 500 ms for round-trip time. I have an ethereal trace showing Ser re-transmitting after less than 300 ms. Anyway, because of our environment I'd like to set T1 to be 1000 ms. I'm using 0.8.14. thanks

2 3

[Users] Openser not consuming the credentials
by Brandon Price 07 Oct '05

07 Oct '05

I have openser set up to remove previously authorized credentials from the message being processed by the server, using consume_credentials. Yet the authorization credentials are still being passed. Any help is as always appreciated. Thank you. The related openser.cfg portions are posted below... if (!www_authorize("", "subscriber")) { www_challenge("", "1"); break; }; consume_credentials(); save("location"); break; if I then restart openser, make a call and start tethereal I get.... tethereal -V port 5060 | grep username Authorization: Digest username="theusernamewashere",realm="ser1.manhattan.whatever.net",nonce= "43457703fa69c4dd5321a7229ea41102f2200390",uri="sip: 13143212222(a)ser1.manhattan.whatever.net",algorithm=MD5,response="fc57c2b 4897d3d7b8ad887a788bf07f6",qop=aut

3 3

[Serusers] variables in ser.cfg
by Müller Thorsten 07 Oct '05

07 Oct '05

Hi all, is there a possibility to use variables in ser.cfg (ser0.9.0) like ... set myip=192.168.1.11 set myaccount=testuser set mypasswd=testpasswd ... modparam("uac","credential","$myaccount:$myip:$mypasswd"); ... rewritehost($myip); ... Perhaps a solution would be with avpops, but I didn't get it to work in conjunction with dbtext. Has someone an example how to read variables with avpops out of a dbtext table which are not connected to a SIP-message? (I read the README for avpops!!) Any other solutions would be a help!! Would this be possible with openser's pseudo variables or another ser-version? Thanks Thorsten

2 1

[Serusers] Blocking User Agent for a call?
by Ryan Pagquil 07 Oct '05

07 Oct '05

Hi Guys, Is there a way for SER to block calls coming from a specific user-agent? I want to block calls so that uniform useragaent will be used by my users. Any modules that can do this? Thanks, Ryan

1 0

[Serusers] ACK loop in SER - Route missing problem ?
by Lambert, Micheline (Satnet) 07 Oct '05

07 Oct '05

Hi, I'm using SER with Sipura and VoIP GW. When there is no line available from the VoIP GW, the VoIP GW replies with 486 Busy Here. The Sipura receives this message and sends an ACK. The ACK has the IP addr. Of the SER. If it is ok, the ACK does not include the "Route" parameter, is it a problem from the Sipura? This cause the SIP to loop the ACK because there is no Route and with the following code, the packet becomes too big!!! ............ if (!method=="REGISTER") record_route(); # ------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------ # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); t_relay(); break; }; ...................... Same configuration and same trace 192.168.20.155 is the SIP Proxy 192.168.20.105 is the VoIP GW 192.168.20.90 is the Sipura 312 54.858597 192.168.20.90 192.168.20.155 SIP/SDP Request: INVITE sip:3353@192.168.20.155, with session description 313 54.860243 192.168.20.155 192.168.20.90 SIP Status: 407 Proxy Authentication Required 314 54.877006 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 315 54.886801 192.168.20.90 192.168.20.155 SIP/SDP Request: INVITE sip:3353@192.168.20.155, with session description 316 54.889720 192.168.20.155 192.168.20.105 SIP/SDP Request: INVITE sip:3353@192.168.20.105:5060, with session description 318 54.924666 192.168.20.105 192.168.20.155 SIP Status: 100 Trying 319 54.925866 192.168.20.155 192.168.20.90 SIP Status: 100 Trying 320 54.950257 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 321 54.951442 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 323 54.968942 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 440 55.527290 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 441 55.528528 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 442 55.547237 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 627 56.624649 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 639 56.626924 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 649 56.647321 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 912 58.719683 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 913 58.720967 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 916 58.746851 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1192 62.806510 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 1193 62.807741 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 1194 62.827148 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1417 66.892115 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 1418 66.893350 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 1419 66.907704 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1628 68.143481 192.168.20.101 192.168.20.55 SIP Thanks. Micheline _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] Sems IVR Module
by Nicky 07 Oct '05

07 Oct '05

Please help. ----- Original Message ----- From: "Nicky" <nicky(a)caliber.com.sg> To: <serusers(a)lists.iptel.org>; <sems(a)lists.iptel.org> Sent: Wednesday, September 28, 2005 10:38 PM Subject: Sems IVR Module > Hi all, > > I am testing the sems version from the Head and when i uses a SIP Phone to > call into > the IVR module for recording when I am in the middle of recording, the > program detected DTMF > signal and jump out of the script. Why is it so? as I never pressed on any > key? How can I resolve this > problem? Why is the voice signal become the DTMF signal? Is there any > adjustment that > i can do to solve this problem , as my voice goes louder the more DTMF > signal is being detected. > > Please help. > > regards, > nicky >

5 7

[Serusers] mediaproxy and private ip
by Olivier Taylor 07 Oct '05

07 Oct '05

Using the last cfg from onsip.org, i just see that now Uas are registered with their private Ip when using mediaproxy. Mediaproxy seems the best solution for scalability, that's why I want to test it. Any Idea?

2 5

[Serusers] ACK loop in SER
by Lambert, Micheline (Satnet) 07 Oct '05

07 Oct '05

Hi, I'm using SER with Sipura and VoIP GW. When there is no line available from the VoIP GW, the VoIP GW replies with 486 Busy Here. The Sipura receives this message and sends an ACK. The ACK has the IP addr. Of the SER. Is it ok, or it should be the IP address of the VoIP GW. When SER receives the ACK with its IP address 192.168.20.155, there is a loop of ACK in SER......... I attached a trace and this is my configuration. 192.168.20.155 is the SIP Proxy 192.168.20.105 is the VoIP GW 192.168.20.90 is the Sipura 312 54.858597 192.168.20.90 192.168.20.155 SIP/SDP Request: INVITE sip:3353@192.168.20.155, with session description 313 54.860243 192.168.20.155 192.168.20.90 SIP Status: 407 Proxy Authentication Required 314 54.877006 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 315 54.886801 192.168.20.90 192.168.20.155 SIP/SDP Request: INVITE sip:3353@192.168.20.155, with session description 316 54.889720 192.168.20.155 192.168.20.105 SIP/SDP Request: INVITE sip:3353@192.168.20.105:5060, with session description 318 54.924666 192.168.20.105 192.168.20.155 SIP Status: 100 Trying 319 54.925866 192.168.20.155 192.168.20.90 SIP Status: 100 Trying 320 54.950257 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 321 54.951442 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 323 54.968942 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 440 55.527290 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 441 55.528528 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 442 55.547237 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 627 56.624649 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 639 56.626924 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 649 56.647321 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 912 58.719683 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 913 58.720967 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 916 58.746851 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1192 62.806510 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 1193 62.807741 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 1194 62.827148 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1417 66.892115 192.168.20.105 192.168.20.155 SIP Status: 486 Busy Here 1418 66.893350 192.168.20.155 192.168.20.90 SIP Status: 486 Busy Here 1419 66.907704 192.168.20.90 192.168.20.155 SIP Request: ACK sip:3353@192.168.20.155 1628 68.143481 192.168.20.101 192.168.20.55 SIP Request: BYE sip:06@192.168.20.105:5060;maddr=192.168.20.105 1630 68.147489 192.168.20.55 192.168.20.105 SIP Request: BYE sip:06@192.168.20.105:5060;maddr=192.168.20.105 1631 68.174246 192.168.20.105 192.168.20.55 SIP Status: 200 OK 1632 68.176946 192.168.20.55 192.168.20.101 SIP Status: 200 OK Micheline

2 1

[Users] possibility to access dst_uri
by Daniel-Constantin Mierla 06 Oct '05

06 Oct '05

Hello, latest openser version from cvs includes a set of new functions (core and avpops) and pseudo-variable which allow to access and manage the value of dst_uri. The dst_uri field contains the address of the next hop, when the routing does not follow R-URI address. Such cases are Route header driven routing (dst_uri is set after loose_route() to the value of next Route header), contacts behind nat (dst_uri is set to the address of nat after lookup("location")) or dispatcher usage. The dst_uri has higher priority in routing over R-URI but less than explicit parameters in relaying functions (e.g., t_relay_to*()). The field was invisible from the script but it proved in the discussions on the mailing list to be important to access it. For example, to detect whether the caller and callee are behind same nat. Also, it can be checked in case of preloaded Route header to secure relaying to special resources (e.g., pstn gateway). With the new branch_route the value of dst_uri can be checked for each branch. A short summary of what was added: * core functions: - setdsturi("uri") - set the value of dst_uri - resetdsturi() - reset the value of dst_uri - isdsturiset() - test if the value of dst_uri is null * pseudo-variables - $du - value of dst_uri (added long time ago) - $dd - domain of dst_uri - $dp - port of dst_uri - $dP - transport protocol of dst_uri * avpops changes - avp_write() can read the value of dst_uri and write it in an avp - avp_pushto() can write the value of an avp in dst_uri Tutorials and dokuwiki will be updated soon. Daniel

4 10

[Users] openser to only offer ulaw
by Brandon Price 06 Oct '05

06 Oct '05

How can I specify only to offer one codec when negotiating an INVITE. Particularly G-711-ulaw?

2 2

[Serusers] Simple Proxy Server
by Zac Tolley 06 Oct '05

06 Oct '05

I want to build a server which will act purely as a SIP proxy. The idea is I will put the proxy into my DMZ, and on my internal firewall I already have a Cisco 2600, which will convert the sip traffic into something the pbx understands. Can I do this with Ser? Do I need to have MySQL installed, after all it's just a pure proxy? Can I use it to create CDR's? Any help appreciated. Zac Tolley

1 0

[Serusers] prefix
by Rick Thompson 06 Oct '05

06 Oct '05

Hi All Does anyone know how to include a # in the following command Prefix("123#4"); Because the above doesn't work. SER doesn't recognize the # and in fact errors out on it. I've also tried Prefix("123\#\4"); Prefix("123'#'4"); Prefix("123[#]4"); Thanks Richard C. Thompson VP of Technology, CCNA Virtual Communications, LLC o 954-334-4000 x 420 f 954-217-7322

2 1

[Serusers] MOH
by Xavier TRENTIN 06 Oct '05

06 Oct '05

Hi all, does anybody knows if there's a way to do something like music on hold with sems, or if somebody is currently programing a plugin for it? Thanks. B.R. Xavier

1 0

Re: [Serusers] xmlrpc calls
by zkeatts 06 Oct '05

06 Oct '05

Jan, Looking the logs hasn't revealed much telling info, but I'll go ahead and send you what I have anyways. I took the liberty of extracting this from my /var/logs/everything.log I was hoping there would be something more helpful regarding that memory allocation error. I do know, after looking at the source, that the "Can't allocate x-byte block" error is coming from my particular xmlrpc-c library. Interestingly enough I have a generic xmlrpc-c.tar.gz which does indicate very well what version it is on. Looking at the sourceforge page all of them have a particular schema which has left me wondering where I found this package. I am going to assume that it is ver 1.03.02, but I am going to go ahead and update it to see what happens, since there was a new release yesterday. I will let you know if that error persists after the update. Logs: Sep 6 09:23:00 myhost ./ser[5674]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5675]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5676]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5677]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5678]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5679]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5680]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5681]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5682]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5683]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5684]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5685]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5686]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5687]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5688]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5689]: INFO: signal 15 received Sep 6 09:23:00 myhost ./ser[5690]: INFO: signal 15 received Sep 6 09:23:22 myhost ser: read 936833521 from /dev/urandom Sep 6 09:23:22 myhost ser: seeding PRNG with 2062863074 Sep 6 09:23:22 myhost ser: test random number 923067521 Sep 6 09:23:22 myhost ser: WARNING: fix_socket_list: could not rev. resolve 63.77.68.19 Sep 6 09:23:22 myhost ./ser[5753]: Maxfwd module- initializing Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is initially 110592 Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is finally 221184 Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is initially 110592 Sep 6 09:23:22 myhost ./ser[5753]: INFO: udp_init: SO_RCVBUF is finally 221184 Sep 6 09:25:22 myhost ./ser[5770]: Binding 'testuser','sip:testUser@localhost' has expired Sep 6 09:25:48 myhost syslog-ng[1812]: STATS: dropped 0 Sep 6 09:27:23 myhost ./ser[5770]: Binding 'testuser','sip:testUser@localhost' has expired Jan Janak wrote: >This is interesting. Your server sends "Can't allocate x-byte memory >block" but this message does not come from SER, neither can I find it in >my version of xmlrpc-c library. > >What version of xmlrpc-c library do you have ? What is the >OS/distribution ? Could you also send me the log from SER ? > > thanks for helping me to debug this. > > Jan. > > >

2 4

[Serusers] Theory behind a media proxy and NATed UA
by Felipe Louback 06 Oct '05

06 Oct '05

The problem with NAT is because the IP address of the SDP body is a non routeable IP. So how does a media proxy works? I read that a media proxy put his address at the sdp body and both UA talks with the media proxy instead of each other. At the time a NAT user sends an INVITE, there is no corresponding port on the NAT. So How does a media proxy contact the NATed UA? Isn't this the same problem the called part will have to contact the NATed party? I read a couple of papers about it, but in all of them it is said that both ends talk to the media proxy and that is all. No explanation about how things work. I know it works because I am using mediaproxy. If anyone could point me how this works or where I can find a document explaining.... Thanks, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "For God so loved the world that he gave his one and only Son, that whoever believes in him shall not perish but have eternal life." John 3:16

4 10

[Serusers] to integrate Server SIP
by Ederson 06 Oct '05

06 Oct '05

as to integrate servers SIP My name is Ederson My language is Portuguesse --------------------------------- Novo Yahoo! Messenger com voz: ligações, Yahoo! Avatars, novos emoticons e muito mais. Instale agora!

1 0

[Users] rtpproxy failover
by harry gaillac 06 Oct '05

06 Oct '05

Hello, How can we use ser and rtpproxy for load-balancing and failover ? if ser is able to handle n calls per second how can we calculate the load for rtpproxy ( memory, cpu, ...) in order to send all traffic to one rtpproxy? If this one is down nathelper module is able to send traffic to an other rtpproxy . SER----------------------------- | | | rtpproxy1 rtpproxy2 rtpproxy3 via unix socket What's the best way for SER failover ? ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 5

[Users] Re: [Devel] openser and asterisk
by Bogdan-Andrei Iancu 06 Oct '05

06 Oct '05

Hi Matt, I redirected this email on the users mailing list - it's more appropriate. the idea seams ok, with couple of comments: 1) be sure that fwd to localhost is ok (instead of a routable IP) 2) doing Record-Route may be a good think. to debug tour problem, add some log("...") statements into your script to be able to trace the processing. Also a network trace (including on lo device) will be helpful to see what happens - if the messages are received, if they are sent and where. Also watch the log for potential errors. regards, bogdan Matt L. Zhu wrote: > has anyone successfully setup openser as the frontend proxy for > asterisk? here is my setup > > /etc/asterisk/sip.conf > [general] > context=default > port=5065 > bindaddr=0.0.0.0 > srvlookup=yes > > [ser] > type=user > context=proxy > host=192.168.0.10 > > then i edited openser.cfg to do something like this > > if > (uri=~"sip:[a-zA-Z\.]*@(xxx\.xxx\.com)|(192\.168\.0\.10)") { > forward( localhost, 5065 ); > break; > }; > > i connected two sipphones (wengo) in this case to openser, but calls > are not going through at all, connecting directly to asterisk works. > have anyone worked in this situation? > > thanks > > > > _______________________________________________ > Devel mailing list > Devel(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/devel >

5 10

[Serusers] A roadmap for SER
by harry gaillac 06 Oct '05

06 Oct '05

Hello, I think that most of the ser users would wish a roadmap for SER ? Does team developpers could provide it ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] www.openpbx.org
by harry gaillac 06 Oct '05

06 Oct '05

Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

4 3

Re: [Serusers] RTPproxy debug
by Joao Pereira 06 Oct '05

06 Oct '05

Hello, /etc/hosts has the two IPs (the private and the public). I m sure the phones in the private network can connect to the private IP of SER and register them selfes. And I m also sure that the phones with public IPs can connect to SERs public IP and register. But how can I check why isnt the RTP trafic passing? I know the phones are registered, I can see the INVITEs with ngrep, but I cant debug the passing of the RTP. Thanks Joao Pereira www.fccn.pt sip wrote: >Have you checked the obvious things like telnetting from inside the network to >the internal IP at port 5060 to make sure the connection is being made? > >Also... how's your /etc/hosts look? If you're using 2 IPs, SER may be trying >to use essentially 2 different domains. If the internal IP isn't resolving as >the same domain as the external (either through DNS (not recommended) or via >/etc/hosts) then SER may be refusing to authenticate because the domains differ. > >Just some things to check if you haven't already. > >N. > > > >On Tue, 04 Oct 2005 18:48:20 +0100, Joao Pereira wrote > > >>I dont know If the problem is realy of my ser.cfg, because the IP >>phones with public IPs work, but the phones with private IPs just >>ring... but I believe theres no RTP passing. Theres my SER: >> >>|--------------| >>| |-------------- private IP >>| S E R | >>| |-------------- public IP >>|--------------| >> >>Theres my ser.cfg: >> >>(...) >>listen = 193.136.xxx.xxx >>listen = 10.0.0.135 >> >>(...) >> >>alias=fccn.pt >>alias=193.136.xxx.xxx >>alias=10.0.0.135 >> >># memlog - Debugging level for final memory statistics report. Default >># is L_DBG -- memory statistics are dumped only if debug is set high. >>memlog=1 >> >># ------------------ module loading ---------------------------------- >> >>loadmodule "/usr/local/lib/ser/modules/postgres.so" >>loadmodule "/usr/local/lib/ser/modules/sl.so" >>loadmodule "/usr/local/lib/ser/modules/tm.so" >>loadmodule "/usr/local/lib/ser/modules/rr.so" >>loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >>loadmodule "/usr/local/lib/ser/modules/usrloc.so" >>loadmodule "/usr/local/lib/ser/modules/registrar.so" >> >>loadmodule "/usr/local/lib/ser/modules/textops.so" >> >>loadmodule "/usr/local/lib/ser/modules/acc.so" >> >># Uncomment this if you want digest authentication >>loadmodule "/usr/local/lib/ser/modules/auth.so" >>loadmodule "/usr/local/lib/ser/modules/auth_db.so" >> >>loadmodule "/usr/local/lib/ser/modules/exec.so" >>loadmodule "/usr/local/lib/ser/modules/group.so" >>loadmodule "/usr/local/lib/ser/modules/print.so" >>#loadmodule "/usr/local/lib/ser/modules/mediaproxy.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/xlog.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" >> >>loadmodule "/usr/local/lib/ser/modules/nathelper.so" >> >># ----------------- setting module-specific parameters --------------- >> >># -- nathelper params -- >> >># proxy nat >>modparam("nathelper", "natping_interval", 15) >>modparam("nathelper", "ping_nated_only", 1) >> >>(...) >> >># ------------------------- request routing logic ------------------- >> >># main routing logic >> >>route{ >> >> 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; >> }; >> >> # -------------------------------------------------- >> # NOTIFY Keep-Alive Section >> # -------------------------------------------------- >> if ((method=="NOTIFY") && search("^Event: keep-alive")) { >> sl_send_reply("200", "OK"); >> break; >> }; >> >> if ((method=="NOTIFY") && (uri=~"^sip:700@")) { >> sl_send_reply("200", "OK"); >> break; >> }; >> >> # -------------------------------------------------- >> # OPTIONS Section >> # -------------------------------------------------- >> if (method=="OPTIONS") { >> options_reply(); >> break; >> }; >> >> if (method=="REGISTER" && nat_uac_test("3")) { >> >> fix_nated_contact(); >> force_rport(); >> setflag(2); >> }; >> >> # -------------------------------------------------- >> # Registration Section >> # -------------------------------------------------- >> (...) >> >> # -------------------------------------------------- >> # Accounting Section >> # NOTE: We test for flag 14 because we do not want to record Click2Dial >> # entries >> # -------------------------------------------------- >> >> if ((method=="INVITE" || method=="BYE" || >>method=="CANCEL"||method=="ACK" ||method=="REGISTER") && >>!isflagset(14)) { setflag(1); }; >> >> # -------------------------------------------------- >> # NAT Tear-Down Section >> # -------------------------------------------------- >> >> if ((method == "BYE" || method == "CANCEL")) { >> unforce_rtp_proxy(); >> }; >> >> # -------------------------------------------------- >> # Record Route Section >> # -------------------------------------------------- >> # 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 >> if (!method=="REGISTER") { >> record_route(); >> }; >> >> # -------------------------------------------------- >> # Loose Route Section >> # >> # Grant route routing if route headers present >> # -------------------------------------------------- >> if (loose_route()) { >> route(2); >> break; >> }; >> >> # >> >> # -------------------------------------------------- >> # NAT Test Section #1 >> # -------------------------------------------------- >> >> if (nat_uac_test("3") && !search("^Record-Route:")) { >> force_rport(); >> >> fix_nated_contact(); >> }; >> >>(...) >> >> # -------------------------------------------------- >> # Alias Routing Section >> # -------------------------------------------------- >> >> lookup("aliases"); >> if (!uri==myself) { >> route(2); >> break; >> }; >> >>(...) >> >> # -------------------------------------------------- >> # Anonymous Call Rejection Section >> # -------------------------------------------------- >> if (isflagset(24) && (method=="INVITE") && >>search("^(f|F)rom:.*(a|A)nonymous")) { >> route(8); >> break; >> }; >> >> # -------------------------------------------------- >> # URI Compare Section >> # -------------------------------------------------- >> # Here we compare the "from" and "to" to see if the caller is dialing >> # their own extension. If so then we route to voicemail if needed >> if (method=="INVITE") { >> avp_write("$from", "i:34"); >> if (avp_check("i:34", "eq/$ruri/i")) { >> if (isflagset(31)) { >> route(5); >> break; >> } else { >> sl_send_reply("486", "Busy"); >> break; >> }; >> }; >> }; >> >>(...) >> >> # -------------------------------------------------- >> # Do Not Disturb Section >> # -------------------------------------------------- >> if (avp_db_load("$ruri/username", "s:donotdisturb")) { >> if (avp_check("s:donotdisturb", "eq/y/i")) { >> route(5); >> break; >> }; >> }; >> >> # -------------------------------------------------- >> # Call Routing Section >> # -------------------------------------------------- >> >> if (!lookup("location")){ >> >> # if flag 31 (ie voicemail) is set and we made it here this means >> # the user's phone is not registered anywhere. We'll forward to >> # voicemail after this block because we need to check the call >> # forward settings first >> if (isflagset(31)) { >> >> # flag 19 means the user has voicemail but is not online >> # so we need to remember to send to voicemail if call >> # forwarding is not enabled >> setflag(19); >> }; >> >> if (method=="INVITE") { >> >> if( !uri=~"@fccn.pt" ) { >> if (!method=="REGISTER") record_route(); >> t_relay(); >> break; >> } >> >> if (does_uri_exist()) { >> # o utilizador foi encontrado mas esta offline >> sl_send_reply("480", "Temporarily Unavailable"); >> break; >> }; >> >> sl_send_reply("404", "Utilizador nao Encontrado"); >> break; >> }; >> }; >> >> # -------------------------------------------------- >> # Call Forwarding Section >> # -------------------------------------------------- >> >> if (method=="INVITE") { >> >> # only load the forward no answer option if voice mail is not enabled >> if (!isflagset(31)) { >> if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { >> route(1); >> >> if (!avp_check("$calltype", "eq/-/i")) { >> if (avp_check("$calltype", "eq/dom/i")) { >> avp_write("dom", "$fwd_no_answer_type"); >> } else if (avp_check("$calltype", "eq/int/i")) { >> avp_write("int", "$fwd_no_answer_type"); >> } else { >> avp_write("sip", "$fwd_no_answer_type"); >> } >> >> setflag(27); >> }; >> }; >> }; >> >> if (avp_db_load("$ruri/username", "s:fwdbusy")) { >> route(1); >> >> if (!avp_check("$calltype", "eq/-/i")) { >> >> if (avp_check("$calltype", "eq/dom/i")) { >> avp_write("dom", "$fwd_busy_type"); >> } else if (avp_check("$calltype", "eq/int/i")) { >> avp_write("int", "$fwd_busy_type"); >> } else { >> avp_write("sip", "$fwd_busy_type"); >> } >> >> setflag(26); >> }; >> }; >> >> avp_pushto("$ruri", "i:99"); >> }; >> >> if (isflagset(19)) { >> # send to voicemail >> route(5); >> } else { >> route(2); >> }; >> >>} #route >> >>route[1] { >> >> avp_write("-", "$calltype"); >> >> if (uri=~"^sip:1[0-9]{10}@") { >> strip(1); >> }; >> >> lookup("aliases"); >> if (!lookup("location")) { >> if (uri=~"^sip:[0-9]{10}@") { >> >> # test for domestic PSTN number >> if (isflagset(28)) { >> avp_write("dom", "$calltype"); >> }; >> >> } else if (uri=~"^sip:011[0-9]*@") { >> >> # test for international PSTN number >> if (isflagset(29)) { >> avp_write("int", "$calltype"); >> }; >> }; >> >> break; >> }; >> >> avp_write("sip", "$calltype"); >> >>} >> >>route[2] { >> >># check for RFC1918 ip addresses >> # -- NATHELPER -- >> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && >>!search("^Route:")){ >> sl_send_reply("479", "We don't forward to private IP addresses"); >> break; >> }; >> >> #check if the NAT Flag is set >> if (isflagset(6)) { >> log(1, "LOG: flag 6 set => One of the sides is NATed -> >>force_rtp_proxy()\n"); >> if(!force_rtp_proxy()) { >> log(1, "LOG: force_rtp_prpoxy FAILED!\n"); >> } else { >> log(1, "LOG: force_rtp_proxy() succeeded!\n"); >> }; >> }; >> >> if (isflagset(25)) { >> replace("^From:(.*)>" , "From: \"Anonymous\" <sip:someone at >>anonymous.invalid>"); >> }; >> >>if (method=="INVITE" || method=="ACK") { >> force_rtp_proxy(); >> }; >> >> t_on_failure("1"); >> t_on_reply("1"); >> >> if (!t_relay()) { >> >> if (method=="INVITE" || method=="ACK") { >> unforce_rtp_proxy(); >> }; >> >> sl_reply_error(); >> }; >>} >> >># encaminhamento para o Gateway PSTN >>route[3] { >> >> if (method=="INVITE") { >> >> if (!proxy_authorize("fccn.pt", "utilizador")) { >> >> proxy_challenge("fccn.pt", "0"); >> break; >> } >> } else if (!check_from()) { >> >> log(1, "Spoofed SIP call attempt"); >> sl_send_reply("403", "Use From=ID"); >> break; >> >> } else if (!(is_from_local() || is_uri_host_local())) { >> >> sl_send_reply("403", "Please register to use our service"); >> break; >> }; >> >> # enable caller id blocking for PSTN calls >> if (isflagset(25)) { >> append_rpid_hf(); >> }; >> }; >> >> # SIP->PSTN calls get 45 seconds to timeout >> avp_write("i:45", "inv_timeout"); >> >>rewritehostport("10.0.0.135:1720"); >> >> if (method=="INVITE" || method=="ACK") { >> force_rtp_proxy(); >> }; >> >> if (isflagset(31)) { >> t_on_failure("1"); >> }; >> >> t_on_reply("1"); >> >> if (!t_relay()) { >> >> if (method=="INVITE" || method=="ACK") { >> unforce_rtp_proxy(); >> }; >> >> sl_reply_error(); >> }; >>} >> >>(...) >> >>onreply_route[1] { >> >> # Not all 2xx messages have a content body so here we >> # make sure our Content-Length > 0 to avoid a parse error >> if (status=~"(180)|(183)|2[0-9][0-9]") { >> >> if (!search("^Content-Length:\ 0")) { >> force_rtp_proxy(); >> }; >> }; >> >> if (nat_uac_test("1")) { >> fix_nated_contact(); >> }; >>} >> >>failure_route[1] { >> >> if (t_check_status("487")) { >> break; >> }; >> >> if (isflagset(26) && t_check_status("486")) { >> >> # forward busy is flag 26 >> if (avp_pushto("$ruri", "s:fwdbusy")) { >> avp_delete("s:fwdbusy"); >> append_branch(); >> resetflag(26); >> # test for domestic PSTN gateway >> if (avp_check("$fwd_busy_type", "eq/dom/i")) { >> # test for domestic PSTN gateway >> route(3); >># sl_send_reply("503", "Gateway PSTN Indisponivel-3"); >> #} else if (avp_check("$fwd_busy_type", "eq/int/i")) { >> # test for international PSTN gateway >> # route(6); >># } else { >> # default to sip call >> route(2); >> }; >> >> break; >> }; >> }; >> >> # here we can have either voicemail __OR__ forward no answer >> if (isflagset(27) && t_check_status("408")) { >> >> # forward no answer is flag 27 >> >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> append_branch(); >> resetflag(27); >> >> if (avp_check("$fwd_no_answer_type", "eq/dom/i")) { >> # test for domestic PSTN gateway >> route(3); >> #sl_send_reply("503", "Gateway PSTN Indisponivel"); >> #} else if (avp_check("$fwd_no_answer_type", "eq/int/i")) { >> # test for international PSTN gateway >> # route(6); >> #} else { >> # default to sip call >> route(2); >> }; >> >> break; >> }; >> >> } else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) { >> >> avp_delete("$voicemail"); >> route(4); >> break; >> }; >>} >> >>onreply_route[2] { >> >> log(1, "LOG: entered onreply_route[2] now... \n"); >> >> if(isflagset(6)) { # && status =~"(183)|2[0-9][0-9]") { >> log(1, "LOG: TRansaction was sent to a NATed client --> >>fix_nated_contact() and force_rtp_proxy().\n"); >> if(!fix_nated_contact()) { >> log(1, "LOG: fix_nated_contact() FAILED!\n"); >> } else { >> log(1, "LOG: fix_nated_contact() succeded.\n"); >> }; >> >> if(!force_rtp_proxy()) { >> log(1, "LOG: force_rtp_proxy() FAILED!\n"); >> } else { >> log(1, "LOG: force_rtp_proxy() succeded.\n"); >> }; >> >> append_hf("P-hint: fixed NAT contact for response\r\n"); >> >> } else if (nat_uac_test("1")) { >> log(1, "LOG: Uncaught NAT. => fix_nated_contact()\n"); >> if(!fix_nated_contact()) { >> log(1, "LOG: fix_nated_contact() FAILED!\n"); >> } else { >> log(1, "LOG: fix_nated_contact() succeded.\n"); >> }; >> }; >> >>} >> >>Sorry the long post. >>Joao Pereira >>www.fccn.pt >> >>Paul Hazlett wrote: >> >> >> >>>Joao, >>> >>>The problem is probably and error in your ser.cfg file. I'd recommend >>>you download a copy of the Getting Started document from www.onsip.org >>>and look at the section on RTPproxy. >>> >>>Regards, >>>Paul >>> >>>On 10/4/05, Joao Pereira <joao.pereira(a)fccn.pt> wrote: >>> >>> >>> >>> >>>>Hello to all, >>>>Is there any way of making debug in RTP proxy? >>>>My SER has one public IP and one private IP. The public phones can speak >>>>each other, but the private IP phones can t speak each other (they just >>>>ring, but theres no voice ) , and apears this error on SER: >>>>ERROR: send_rtpp_command: can't read reply from a RTP proxy >>>> >>>>How can I debug RTPproxy? >>>>Thanks >>>> >>>>Joao Pereira >>>>www.fccn.pt >>>> >>>> >>>> >>>>_______________________________________________ >>>>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 >> >> > > > >

1 0

Re: [Serusers] SerWeb login
by Karel Kozlik 06 Oct '05

06 Oct '05

Hi, please (CC) your mails to the mailing list in order to others can see the replies. No, I mean SerWeb log. You must enable it in config file. $config->enable_loging = true; Which version of serweb are you useing? Karel Voipers Portugal napsal(a): > Well, i did what you told me, and now when i press Login, i get a > blank page with nothing on in. And if i press refresh, i get the login > page again. > > When you asked for the log, is the ser log, right? Well, nothing > happened there when i try to access the login page. Any idea? > > Jose Simoes > > > > 2005/10/5, Karel Kozlik <karel(a)iptel.org>: > >>Hi, >>which version of serweb are you useing? If some old, check your php.ini >>for register_globals = on. >> >>If you have serweb 0.9.x, you are first who reporting this behaviour. >>Please enable logging, set log level to debug and send me the logfile. >> >>regards Karel >> >>Voipers Portugal napsal(a): >> >>>Hi, >>> >>>When i am trying to access serweb in http://host/ser/admin/index.php i >>>get the space to authenticate myself.. but independently from what i >>>type in username and password, when i press login nothing happens. any >>>idea? >>> >>>Thanks in advance, >>> >>>Jose Simoes >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>

1 0

[Serusers] Radius ser.conf
by Andrej Sirk 06 Oct '05

06 Oct '05

Hello, Is there an example of ser.conf using radius authentication. Thanks in advance.

1 0

[Serusers] append_hf, append_urihf (append_furihf)
by Greg Fausak 06 Oct '05

06 Oct '05

Hi, I'm about to hack up the textops module a bit, and wondered if anyone else has run into this. I'd like to create a line like: P-Asserted-ID: 14085551212(a)mydomain.com and insert this into the sip packet. I'd use append_hf(), except the number listed above is contained in the From: URI. I was going to hack a append_urihf() into a append_furihf() to accomplish the task...can I do this with an exec_dset or something else? -g -- Greg Fausak greg(a)thursday.com

6 7

[Serusers] Windows Messenger with SER
by Ana Paula Oliveira dos Santos 06 Oct '05

06 Oct '05

Hello, Somebody can help me in a doubt that I have with relation to the use of the SER with Windows Messenger? Is it necessary to include something in the code it ser.cfg, as some new configuration so that it has accepted register, required instant messages and presence through the Windows Messenger? If exists any configuration, somebody could please, send the archive ser.cfg with them? Thanks in advance, Ana.

2 1

[Serusers] SER and cpu usage
by Chia Huey Lim 06 Oct '05

06 Oct '05

Hi, I am using ser 0.9.3 and postgresql with db mode 2. There is a very serious problem when SER is binding users. Each SER PID takes around 11% to 13% of CPU, which add up to be 99.x% And at that time, no one can REGISTER, no one can INVITE etc. Anyone experience that before? Regards, Chia

3 5

[Serusers] forwarding a call to a phone number using SER
by Sam Lee 06 Oct '05

06 Oct '05

Do look at the example in onsip.org They have a sample config file for call forwarding. Sam ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of sagar Sent: Thursday, October 06, 2005 12:10 AM To: serusers(a)lists.iptel.org Subject: [Serusers] forwarding a call to a phone number using SER Hi All, I want to impliment the call forwarding in my SER. Can anybody please send me a working ser.cfg which forwards a call to any particular phone number. I am very hopefull from your side. THANKS IN ADVANCE . SAGAR. vids_cs(a)yahoo.com <http://us.f518.mail.yahoo.com/ym/Compose?To=vids_cs@yahoo.com> vids.cs(a)gmail.com <http://us.f518.mail.yahoo.com/ym/Compose?To=vids.cs@gmail.com> ________________________________ Yahoo! for Good Click here to donate <http://store.yahoo.com/redcross-donate3/> to the Hurricane Katrina relief effort.

1 0

RE: [Serusers] forwarding a call to a phone number using SER
by Sam Lee 06 Oct '05

06 Oct '05

Do look at the example in onsip.org They have a sample config file for call forwarding. Sam ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of sagar Sent: Thursday, October 06, 2005 12:10 AM To: serusers(a)lists.iptel.org Subject: [Serusers] forwarding a call to a phone number using SER Hi All, I want to impliment the call forwarding in my SER. Can anybody please send me a working ser.cfg which forwards a call to any particular phone number. I am very hopefull from your side. THANKS IN ADVANCE . SAGAR. vids_cs(a)yahoo.com <http://us.f518.mail.yahoo.com/ym/Compose?To=vids_cs@yahoo.com> vids.cs(a)gmail.com <http://us.f518.mail.yahoo.com/ym/Compose?To=vids.cs@gmail.com> ________________________________ Yahoo! for Good Click here to donate <http://store.yahoo.com/redcross-donate3/> to the Hurricane Katrina relief effort.

1 0

[Users] TLS - certificate infrastructure
by Alexander Ph. Lintenhofer 05 Oct '05

05 Oct '05

Hi everybody, I want to test openser 0.10.x and its TLS capabilities. Therefore I plan to install two proxies, sip.atlanta.com and sip.biloxi.com. Two users, alice(a)atlanta.com and sip.biloxi.com, should communicate over the two proxies secured by TLS. The UAs are snom360 phones. ------------------- ----------------- ----------------- ----------------- | alice(a)atlanta.com | <-------> | sip.atlanta.com | <-------> | sip.biloxi.com | <-------> | bob(a)biloxi.com | ------------------- ----------------- ----------------- ----------------- Mutual authentication should take place between the UAC and the outbound proxy, the two proxies and between the inbound proxy and the UAS. The problem is that I am not sure about the organisation of the certificate's infrastructure. I don't know which would be the best solution to implement. So please look at my suggestions and feel free to you make your comments. 1.. user certificate for alice(a)atlanta.com 2.. server certificate for sip.atlanta.com 3.. server certificate for sip.biloxi.com 4.. user certificate for bob.biloxi.com The root certificate is self signed (Does this work with openser?) a.) One common CA (=root) signs all components. ----------- | CA | ----------- / / \ \ / / \ \ / | | \ --- --- --- --- |1| |2| |3| |4| --- --- --- --- b.) Tow separate CAs (= each one's root) sign their proxy and UA. Mutual import of the other domains root certificate takes place. ----- ----- |CA A | |CA B | ----- ----- / \ / \ --- --- --- --- |1| |2| |3| |4| --- --- --- --- c.) One common root signs two CAs which sign their proxy and UA. ----------- | root-cert | ----------- / \ / \ ----- ----- |CA A | |CA B | ----- ----- / \ / \ --- --- --- --- |1| |2| |3| |4| --- --- --- --- Thank you very much for your help! regards, Philipp

2 1

[Serusers] Radius Authentication
by Patrick Jordan-Smith 05 Oct '05

05 Oct '05

Hi list, I have a problem where my SER is supposed to be authenticating a user out from the radius server, but it doesn't contact the radius server. When I sniff the ethernet I can see it doing DNS lookups (incl answers) on the radius server but it never actually sends radius packets. It even does PTR lookups on the IP address, but still nothing. If anyone has any ideas of why this his happening, please help me :) Thanks, Patrick

2 2

[Serusers] dummy or very complicate???
by Olivier Taylor 05 Oct '05

05 Oct '05

just a question, Am i dummy or my questions are so complicated? Or banned maybe :( I never ave an answer to my questions, very disapointing... Regards, Olivier

2 1

[Users] OpenSER not processing "200 OK" reponse to CANCEL message
by Tavis P 05 Oct '05

05 Oct '05

Using logic similar to the Onsip.org config file i'm doing call forward no-answer using a TM timer, when the call is forwarded OSER sends a cancel to the original client which responds with a "200 OK". OSER doesn't seem to process this 200 ok, generating an error and resending the CANCEL message. Running OSER in debug level 5 shows that OSER was not able to match the "200 OK" to the CANCEL message Below is a trace of exchange, the following error is recorded in the debug log -- get_hdr_field: cseq <CSeq>: <102> <CANCEL> header field type 5, name=<CSeq>, body=<102 CANCEL> DEBUG: t_reply_matching: hash 44785 label 1883536846 branch 0 DEBUG: t_reply_matching: no matching transaction exists DEBUG: t_reply_matching: failure to match a transaction DEBUG: t_check: msg id=3 global id=3 T end=(nil) .... ERROR: forward_reply: no 2nd via found in reply -- # U OSER_PROXY:15061 -> SIP_UAC:5060 CANCEL sip:200110005333@192.168.99.240:5060 SIP/2.0. Via: SIP/2.0/UDP OSER_PROXY:15061;branch=z9hG4bKb3b2.9e597b73.0. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=d478d8ef4036cbffo0. Call-ID: c818cfaf-edd9443f(a)192.168.99.66. To: <sip:200110005333@OSER_PROXY:15061>. CSeq: 102 CANCEL. Content-Length: 0. . # U SIP_UAC:5060 -> OSER_PROXY:15061 SIP/2.0 200 OK. Via: SIP/2.0/UDP OSER_PROXY:15061;branch=z9hG4bKb3b2.9e597b73.0. From: tavis <sip:200110005334@OSER_PROXY:15061>;tag=d478d8ef4036cbffo0. To: <sip:200110005333@OSER_PROXY:15061>;tag=00055e7cd6f33c1a551406a2-75f29afe. Call-ID: c818cfaf-edd9443f(a)192.168.99.66. CSeq: 102 CANCEL. Server: Cisco-CP7960G/7.5. Content-Length: 0. .

3 5

Re: [Users] openser2openser
by Alexander Ph. Lintenhofer 05 Oct '05

05 Oct '05

Hi Klaus, > Alexander Philipp Lintenhofer wrote: > >> Hi Klaus, >> >> TLS: Is this feature already tested with version 0.10.x? Is it >> necessary that >> both proxies are under the same root-CA or is it possible to define >> different > > > up to now I did not tested it, I just read the README. If I understand > it correctly, than you can import as man CA certs as you like. OK, that is also my state of information. So you import the root certificates of all trusted domains with which you want authentification. > >> trust anchors by distributing root certificates? Or do I need a >> cross-path >> mechanism to deal with this problem? > > > At the moment I'm having problems figuring out how the server > certificate must look like. The standard is X509v3. > e.g. a lookup for sip:klaus@example.net may lead to another domain > using SRV. Which domain must be in the certificate? Where in the > certificate (Subject? Subject alternative name? ...) The SRV-Request yields the resonsible sipserver of example.net. According to RFC3261 the subject of the certificate must correspond to the canonical hostname of this server. I believe that your outbound proxy exchanges his certificate with the inbound of example.net for mutual authentification. So regarding RFC 2246 both need a way to validate the other cert. -> ? regards, philipp > >> >>> proxy2proxy authentication is usually done by TLS. >>> >>> The problem is that both proxies use different nonce to >>> authenticate. You can try to set the secret on both proxies: >>> http://openser.org/docs/modules/0.10.x/auth.html#AEN62 >>> >>> regards >>> klaus >>> >>> Taras Bendik wrote: >>> >>>> Situation: >>>> client1 ----->openser1 ----> openser2 ---->client2 >>>> Both openser have same accounts (user/pass) >>>> >>>> When im not using proxy authentification it works ok. >>>> If i use it it gives me 407 >>>> >>>> i have tried to use following >>>> http://www.voice-system.ro/docs/uac/ar01s06.html#ex_auth >>>> >>>> and always goes executing this part >>>> if (isflagset(7)) { >>>> t_reply("503","Authentication failed"); >>>> break; >>>> } >>>> >>>> I look at ngrep log, and it is some thing like this >>>> ser1 -> ser2 INVITE >>>> ser2 -> ser1 AUTH Required >>>> ser1 -> ser2 INVITE with auth >>>> ser2 -> ser1 AUTH Required >>>> ser1 -> ser2 INVITE with auth >>>> ser2 -> ser1 AUTH Required >>>> It seems to me that openser1 cannot authentificate on openser2. >>>> >>>> >>>> Thanks in advance >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> >> >> > > >

1 0

[Serusers] forwarding a call to a phone number using SER
by sagar 05 Oct '05

05 Oct '05

Hi All, I want to impliment the call forwarding in my SER. Can anybody please send me a working ser.cfg which forwards a call to any particular phone number. I am very hopefull from your side. THANKS IN ADVANCE . SAGAR. vids_cs(a)yahoo.com vids.cs(a)gmail.com --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

2 1

[Serusers] Jabber module on latest stable (0.9.x) versions
by Matt L. Zhu 05 Oct '05

05 Oct '05

has anyone found resolution to this issue? Hi! Is there anybody who's succesfully running jabber module on top of a recent "stable" ser/openser version? I can't get the server running, it hangs and I have then to kill it with "kill -9". This is the error log (running on openser 0.9.5, not forking, debug level = 8): ... 0(21394) DEBUG: init_mod_child (1): avpops 0(21394) get_connection(): Connection found in the pool 0(21394) XJAB:xjab_check_workers: worker[0][pid=21403] has exited - status=0 err=21403 errno=0 0(21394) XJAB:21402:xjab_check_workers: create a new worker[0] 0(21394) XJAB:xj_worker[0]:21405: exiting - wrong parameters 0(21394) XJAB:xjab_check_workers: worker[1][pid=21404] has exited - status=0 err=21404 errno=0 0(21394) XJAB:21402:xjab_check_workers: create a new worker[1] 0(21394) XJAB:xj_worker[1]:21406: exiting - wrong parameters 0(21394) SIGCHLD received: we do not worry about grand-children I don't really understand why I get different errors on different modules lowering the debug level (3): 2(0) get_connection(): Inherited open database connections, this is not a good idea 2(0) db_init(): Could not create a connection 2(0) ERROR: child_init(-2): Error while connecting database 2(0) init_mod_child(): Error while initializing module usrloc Please note that everything is working fine if I disable jabber module I have seen jabber module hasn't been modified recently, is it possible it is now somehow not compatibile with *sql modules? -- Ing. Pietro Ravasio - pietro.ravasio at abbeynet.it Enterprise Market Manager Abbeynet S.p.A. - http://www.abbeynet.it TEL: +390702109044 GSM: +393282172055 FAX: +390702109055

1 1

[Users] rtpproxy.sock does not responding
by Zdenek Svarc 05 Oct '05

05 Oct '05

Hi all, I use OpenSER 0.9.5 and the latest rtpproxy from snapshots repository (http://www.openser.org/downloads/snapshots/rtpproxy/), 2.6.8 kernel, sarge. This error occures when i try to run openser: :ERROR: send_rtpp_command: can't connect to RTP proxy :send_rtpp_command(): proxy <unix:/var/run/rtpproxy.sock> does not responding, disable it :WARNING: rtpp_test: can't get version of the RTP proxy :WARNING: rtpp_test: support for RTP proxy <unix:/var/run/rtpproxy.sock>has been disabled temporarily Part of the config: modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("registrar", "nat_flag", 6) Of course rtpproxy is running (without any parameters, is this correct?) and /var/run/rtpproxy.sock is there. OpenSER is running under openser user and RTPproxy is running under root user. Do you have any idea? Regards, Zdenek

2 2

[Serusers] query for lcr table
by Iqbal 05 Oct '05

05 Oct '05

Hi I have a query which select the correct prefix from the lcr table, but have realised that it my be incorrect, is anyone else using something similiar for rating SELECT prefix from lcr WHERE 'sip:1234@sip.domain.com' LIKE lcr.from_uri AND '35387318220' LIKE CONCAT(lcr.prefix, '%') ORDER BY CHAR_LENGTH(lcr.prefix), lcr.priority DESC, RAND() ; Now this returns +--------+ | prefix | +--------+ | 353 | | 3538 | | 35387 | +--------+ Now I assume you should pick the one with the best match, i.e at the bottom, if so should the query not have ASC no desc in the query, rather than having to add another line which takes all result and sees length of each. Iqbal

2 2

[Users] openser2openser
by Taras Bendik 05 Oct '05

05 Oct '05

Situation: client1 ----->openser1 ----> openser2 ---->client2 Both openser have same accounts (user/pass) When im not using proxy authentification it works ok. If i use it it gives me 407 i have tried to use following http://www.voice-system.ro/docs/uac/ar01s06.html#ex_auth and always goes executing this part if (isflagset(7)) { t_reply("503","Authentication failed"); break; } I look at ngrep log, and it is some thing like this ser1 -> ser2 INVITE ser2 -> ser1 AUTH Required ser1 -> ser2 INVITE with auth ser2 -> ser1 AUTH Required ser1 -> ser2 INVITE with auth ser2 -> ser1 AUTH Required It seems to me that openser1 cannot authentificate on openser2. Thanks in advance

4 4

[Serusers] Radius client
by Andrej Sirk 05 Oct '05

05 Oct '05

Hello, i have the following error with radiusclient-ng: #radiusclient -f digest localhost -a test123 error opening radius configuration file thanks

1 0

[Serusers] Re: [Serdev] Conferencing
by Paul Hazlett 05 Oct '05

05 Oct '05

Rajeev, I do not use SEMS myself. We use Asterisk for voicemail -- but it can do conferencing as far as I know. Perhaps you can visit www.voip-info.org or www.asteriskdocs.org to find out how to configure Asterisk for conferencing. You can also check out www.onsip.org to get the Getting Started document which shows how to do call forwarding with SER. Regards, Paul On 10/5/05, rajeev.khetan(a)cem-solutions.net <rajeev.khetan(a)cem-solutions.net> wrote: > Hi Paul, > > Ok I downloded SEMS and installed it, but dont know how to use it, > I think it is related to config file, can you mail me any sample config > file to > check conferencing , voice mail, callforwarding etc features. > > Regards, > Rajeev > > Paul Hazlett wrote: > > >Rajeev, > > > >SER itself cannot do conferencing. In fact it doesn't understand > >anything about RTP streams at all. > > > >If you want to do conferencing or voice mail or other sorts of things > >then you need to look at integrating SER with other software > >applications, such as Asterisk PBX or SEMS. > > > >Regards, > >Paul > > > >On 10/4/05, rajeev.khetan(a)cem-solutions.net > ><rajeev.khetan(a)cem-solutions.net> wrote: > > > > > >>Hi, > >> > >>I want to know wheather ser proxy support conferencing ?. > >>If it support plz. send the config file for it. > >>Thanks in advance. > >> > >>Regards, > >>Rajeev > >> > >> > >>_______________________________________________ > >>Serdev mailing list > >>serdev(a)lists.iptel.org > >>http://lists.iptel.org/mailman/listinfo/serdev > >> > >> > >> > > > > > > > > > > >

1 0

[Serusers] CAll id in IM
by Abhijit A. Mahajani 05 Oct '05

05 Oct '05

Hello, I wish to send an IM to some UAC from our custom UAS,so while framing the MESSAGE request , in the Call Id field, whose Call id do i fill? Sender's or receiver's? Thanks and Regards. Abhijit

1 0

[Users] INVITE Timeout
by Helge Waastad 05 Oct '05

05 Oct '05

I just wonder if this is a script error or if it is something else: I am loading an INVITE timeout using avpops and the OpenSER terminates the call. The callflow is as follows: A -------------OpenSER------------B ---INVITE----> <---100------- ---INVITE---> <---100------ <---180------ <---180------- <Inv Timeout delay> <---408------- ---CANCEL---> <---487------ ---ACK------> <---200 OK--- <----200 OK--- ---ACK-------> ---CANCEL---> <---481------ ---CANCEL---> <---481------ Any comments? I do not want anyone to fix my problem, just a direction :-) -- mvh/best regards Helge Waastad System Engineer Smartnet

5 8

[Users] t_on_branch
by Richard Z 05 Oct '05

05 Oct '05

Hi, I am trying to use t_on_branch to identify if each branch is behind the same NAT as the caller. A branch_route is defined and check dst_uri's address. If any one branch is behind different NAT, a flag is set. The flag is checked later to call rtp/mediaproxy functions. But there is a problem. It looks like the branch_route is NOT called until a t_relay is called. That's way after the flag is checked. If that's the case, I don't see how branch_route can help any NAT situation. Thanks, Richard

3 2

[Serusers] avpradius and aliases
by Olivier Taylor 05 Oct '05

05 Oct '05

hello, I use avpradius (table preferences) for call-forwarding. When i call a subscriber, no problems, when i call an alias, no avp found for the alias? What do I wrong? regards, Olivier

1 0

[Serusers] SerWeb login
by Voipers Portugal 05 Oct '05

05 Oct '05

Hi, When i am trying to access serweb in http://host/ser/admin/index.php i get the space to authenticate myself.. but independently from what i type in username and password, when i press login nothing happens. any idea? Thanks in advance, Jose Simoes

2 1

[Serusers] Click to Send Web Based Application
by viquar syed 05 Oct '05

05 Oct '05

Hi, I am looking for some input on how to program Click to Dial web application. If someone has written or has some info on getting a sample to look at, please direct me in the right direction. Regards, Viquar Syed

3 2

[Serusers] dbtext
by Andrej Sirk 05 Oct '05

05 Oct '05

Hello, Has anybody an example of text database file. Thanks.

2 1

[Users] CDR error
by jayesh nambiar 05 Oct '05

05 Oct '05

hi everybody, I've just put my openser 0.9.5 in the production environment and it works well. For now, I jus use a query to extract the CDR as "call_date", "orig_number", "termination_number", "duration" and "src_ip". but when the volume is very large I get the following problem. The scenario is as follows. call_date | orig_number | term_number | duration | src_ip 2005-09-25 20:45:30 | 203@my_sip_server | 56924536@my_sip_server | 14:00:30 X.X.X.X 2005-09-25 20:45:30 |56924356@my_sip_server | 43234123@my_sip_server| 4:00:30 X.X.X.X I get the same number as the origination number which was the destination number some moments before, when in reality it is not possible to call from that particular number. But this only happens when the calling volume is really large. Does anyone have any idea regarding this. If you want to see the query I can do that. pls share if there is some better query to generate the CDR.. I have also downloaded the latest version 0.10.x from CVS head. Can somebody tell me how stable is it for the production environment. thanx jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

2 4

[Serusers] Sip Call Tear down
by Ritesh Jalan 05 Oct '05

05 Oct '05

Hi All We came accross a typical problem, Let me describe you through SIP IP Phone -----------> SIP Proxy --------------> MediaGateway -----------PSTN Network IP address 'A' IP address 'B' IP address 'C' Phone No. 'xxxx' Now when we make a call from SIP IP Phone to a PSTN network say xxxx no (an IVR no.). call gets establish, but in the mean while if the network of SIP IP Phone goes down, call is still going on. Is there any method through which we can check if the IP phone is UP (on Network) or not, if not then call gets disconnected. Ritesh Jalan

1 0

FW: [Serusers] voicemail route
by Rick Thompson 04 Oct '05

04 Oct '05

Aisling I am using the following to get to voicemail. failure_route[1] { xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm> <%rr> <%rs>\n"); if(t_check_status("487")) { break; }; if(method=="INVITE" && (t_check_status("486|408|404|480"))) { if(avp_db_load("$ruri", "s:mailbox")) avp_pushto("$ruri/username", "s:mailbox"); prefix("V"); rewritehostport("66.236.248.133:5060"); append_branch(); xlog("L_ALERT", "****** Transfering to Voicemail\n"); t_on_reply("1"); t_relay(); }; } This works for uid busy and uid not registered but doesn't work for ser's 30 sec time out 408 "modparam("tm", "fr_inv_timer", 30)" So if someone can help me figure out why I would appreciate it. Rick _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Aisling Sent: Tuesday, October 04, 2005 5:56 AM To: 'Greger V. Teigre'; serusers(a)lists.iptel.org Subject: RE: [Serusers] voicemail route Ok Greger, thanks for the reply. Just a quick question, for the forward to voicemail on busy scenario, I can see the callee phone sending the 486 busy message to SER and SER sends an ACK back. At this stage I then see SER send a 404 to the caller.I presume the correct sequence instead for this would be SER forwarding the INVITE to Asterisk? Many thanks. -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: 04 October 2005 06:55 To: Aisling; serusers(a)lists.iptel.org Subject: Re: [Serusers] voicemail route Aisling, I think the only way you can get further on this is to use ngrep and create a complete trace of the call. Then you have to match each of your log messages to each SIP message. sip_scenario can help you in drawing out who sent what. Remember that once you relay to Asterisk, Asterisk will get in the loop and these messages should also be relayed properly. My guess is that this has something to do with the OK or ACK at the end of call. Most likely you forget about a SIP message when reading your logs... ;-) (I've done it myself so many times) g-) ----- Original Message ----- From: Aisling <mailto:ashling.odriscoll@cit.ie> To: serusers(a)lists.iptel.org Sent: Monday, October 03, 2005 09:05 PM Subject: [Serusers] voicemail route Hello everyone, I am using the onsip call features ser.cfg and am adapting it for asterisk voicemail. This is what I currently have changed: 1) In the usr_preferences table in the ser database have an entry for user 2092. Insert into usr_preferences (username, attribute, value) values ("2092", "voicemail", "y"); 2) In Route[3] (used for call invite handling) if(avp_db_load("$ruri/username","s:voicemail")){ if(avp_check("s:voicemail", "eq/y/i")){ setflag(18); }; }; This will check if the user wants to use voicemail according to the preference that is set for them in the usr_preferences table. I they don't want to use voicemail set value to "n" 3) In failure route[1] if (call fwd on no answer is enabled{ } else if(isflagset(18) && t_check_status("408")){ route(x); break; }; 4) route[x] { acc_db_request("missed called", "missed_calls"); revert_uri(); rewritehostport("x.x.x.x:5064"); #port where asterisk is listening append_branch(); t_relay_to_udp(x.x.x.x", "5064"); break(); } I am getting a 404 sent back to the phone..I suspect this is something got to do with route 1 as I have used loads of log messages and I can see the flag being set, route x being called but after the failure route, the code jumps to route 1...This is probably because in route 3 it says t_on_failure("1") followed by route 4 followed by route 1...I just don't know what to do about it....Does anyone have any suggestions? Kindest Regards, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. _____ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] 403 No relaying
by rajesh volluru 04 Oct '05

04 Oct '05

hi I had installed SER. I get following error when I try to configure jain sip communicator to use SER. SIP/2.0 403 No relaying Call-ID: aa7c1df636a1640571eb10e920781a2b(a)192.168.1.169 CSeq: 1 REGISTER From: "Rajesh" <sip:admin@192.168.1.207:20318;transport=udp>;tag=6972371 To: "Rajesh" <sip:admin@192.168.1.207:20318;transport=udp>;tag=b51ece8fd8c195776737473dd6552d43.1682 Via: SIP/2.0/UDP 192.168.1.169:20318;branch=z9hG4bKcba4c28b1d1942c2fe848a76e481591d;rport=20318;received=68.225.20.250 P-Behind-NAT: Yes Server: Sip EXpress router (0.9.0udpfifo (i386/linux)) Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=8440 req_src_ip=68.225.20.250 req_src_port=20318 in_uri=sip:192.168.1.207:5060;transport=udp out_uri=sip:192.168.1.207:5060;transport=udp via_cnt==1" Content-Length: 0 Please help me resolve this. Thanks Rajesh --------------------------------- Yahoo! India Matrimony: Find your partner now. --------------------------------- Yahoo! India Matrimony: Find your partner now.

1 0

[Serusers] acc module and failed_transactions
by Federico Giannici 04 Oct '05

04 Oct '05

I have activated the "failed_transactions" flag in SER 0.9.4. Now the "acc" module logs the 487 errors for INVITEs, but it DOESN'T log the 404 or 480 errors. Why? Perhaps they are considered "missed calls" (i don't log missed calls)? But then the 487 should be considered missed call too... Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

5 6

[Serusers] SER Distributions
by gorand＠dvvti.com 04 Oct '05

04 Oct '05

Are there any SER Distrobutions already set up on an ISo File with all the appropriate scrips. Something similar to Asterisk@Home. Please let me know if one exists and where I can find it. Thanks

3 2

[Users] mediaproxy and registrar on two different server
by andrea 04 Oct '05

04 Oct '05

Dear all, I'm working to separate on two different servers registrar and mediaproxy. Mediaproxy server (with both mediaproxy and ser proxy process) will be the front-end and all registers and calls from Intenet will be directed to this server and then routed to internal registrar. Now, I'm working with natted clients. I've started with only one server (with only one ser) and it works fine. When I've tried to split the architecture on two server I can't fine a good configuration. In particular, when a user with a public IP calls a natted user, the RTP doesn't use mediaproxy Anyone can help me? do you have any working configurations? thanks, Andrea

2 4

[Serusers] when binding can't invite?
by Chia Huey Lim 04 Oct '05

04 Oct '05

Hi, I notice that whenever SER is binding UAs, everyone who register / invite at that time will just get stucked for a few seconds. SER will not response when it's binding UAs Is this normal? Anything can be done to improve this situation? Thanks in advance Regards, Chia

6 8

[Serusers] Re: [Serdev] Conferencing
by Paul Hazlett 04 Oct '05

04 Oct '05

Rajeev, SER itself cannot do conferencing. In fact it doesn't understand anything about RTP streams at all. If you want to do conferencing or voice mail or other sorts of things then you need to look at integrating SER with other software applications, such as Asterisk PBX or SEMS. Regards, Paul On 10/4/05, rajeev.khetan(a)cem-solutions.net <rajeev.khetan(a)cem-solutions.net> wrote: > Hi, > > I want to know wheather ser proxy support conferencing ?. > If it support plz. send the config file for it. > Thanks in advance. > > Regards, > Rajeev > > > _______________________________________________ > Serdev mailing list > serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev >

1 0

[Serusers] Radius
by Andrej Sirk 04 Oct '05

04 Oct '05

Hello, How to load radius authentication modules into ser. I have compiled ser as described in the radius howto and I can not load modules in it. I have tried lots of options but without success. When added a line (example: loadmodule "/usr/local/lib/ser/modules/auth_radius.so") in ser.cfg there was an error. Please Help.

2 1

[Serusers] relocation error with binary 0.9.4 for linux
by Christian Gerstner 04 Oct '05

04 Oct '05

Hi! I tried to run binary from package ser-0.9.4_linux_i368.tar.gz and I get: ser: relocation error: ser: symbol regexec, version GLIBC_2.3.4 not defined in file libc.so.6 with link time reference What's wrong? Is it my fault? Regards, Christian

1 0

[Serusers] voicemail route
by Aisling 04 Oct '05

04 Oct '05

Hello everyone, I am using the onsip call features ser.cfg and am adapting it for asterisk voicemail. This is what I currently have changed: 1) In the usr_preferences table in the ser database have an entry for user 2092. Insert into usr_preferences (username, attribute, value) values ("2092", "voicemail", "y"); 2) In Route[3] (used for call invite handling) if(avp_db_load("$ruri/username","s:voicemail")){ if(avp_check("s:voicemail", "eq/y/i")){ setflag(18); }; }; This will check if the user wants to use voicemail according to the preference that is set for them in the usr_preferences table. I they don't want to use voicemail set value to "n" 3) In failure route[1] if (call fwd on no answer is enabled{ } else if(isflagset(18) && t_check_status("408")){ route(x); break; }; 4) route[x] { acc_db_request("missed called", "missed_calls"); revert_uri(); rewritehostport("x.x.x.x:5064"); #port where asterisk is listening append_branch(); t_relay_to_udp(x.x.x.x", "5064"); break(); } I am getting a 404 sent back to the phone..I suspect this is something got to do with route 1 as I have used loads of log messages and I can see the flag being set, route x being called but after the failure route, the code jumps to route 1...This is probably because in route 3 it says t_on_failure("1") followed by route 4 followed by route 1...I just don't know what to do about it....Does anyone have any suggestions? Kindest Regards, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

3 4

[Serusers] Ser-Web interface errors...
by Pablo Cabezas 04 Oct '05

04 Oct '05

Hi! I have access to the admin web page (http://localhost/serweb/admin/index.php) with the default user "admin". I can see the users I have created with the "serctl add..." wish. The problem comes when clicking in the hyperlink "account" for one of the users in the list (in the administration web page). A new page is opened, and in the upper section appears the errors: # Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied in /var/lib/serweb/html/functions.php on line 315 # Warning: Cannot modify header information - headers already sent by (output started at /var/lib/serweb/html/functions.php:315) in /var/lib/serweb/html/page.php on line 35 And written with red font, appears this message too: # DB Error: no such table # sorry -- cannot open write fifo I have uncommented the line "fifo_mode=0660" at the /etc/ser/ser.cfg, as it was told here at the list, but it goes wrong. I have changes permissions in /tmp/ser_fifo, but then the web page doesnt get opened... I would appreciate your help in this point, because I suppose that creating user accounts and managing them is the only required step to intercomunicate softphones in a simple LAN, isnt it? Thanks! Pablo

2 2

Re: [Serusers] Ser-Web interface errors...
by Karel Kozlik 04 Oct '05

04 Oct '05

Hi, please (CC) your mails to the mailing list in order to others can see the replies. Coments are inline. > Ok Karel, that went ok! > > Now, when accessing to http://localhost/serweb/user...) with a user > account, appears the error message "DB Error: No such table". > > What table does it refer to? I thought that launching the initial wish > to create the ser tables should be enough in order to intialize all > these things related to the mySql database... Yes, it is enought. The problem may be in incopatible versions of ser and serweb. Try run: ser_mysql.sh reinstall If it will not help, try enable logging in serweb and see the log file. Here you will see the name of missing table. Karel > > Thanks! > Pablo > > >> From: Karel Kozlik <karel(a)iptel.org> >> To: Pablo Cabezas <paolochan(a)hotmail.com> >> CC: serusers(a)lists.iptel.org >> Subject: Re: [Serusers] Ser-Web interface errors... >> Date: Tue, 04 Oct 2005 11:09:09 +0200 >> >> Hi, >> try set fifo_mode=0666 in your ser.cfg >> >> Karel >> >> Pablo Cabezas napsal(a): >> >>> Hi! >>> >>> I have access to the admin web page >>> (http://localhost/serweb/admin/index.php) with the default user >>> "admin". I can see the users I have created with the "serctl add..." >>> wish. >>> >>> The problem comes when clicking in the hyperlink "account" for one of >>> the users in the list (in the administration web page). A new page is >>> opened, and in the upper section appears the errors: >>> >>> >>> # Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission >>> denied in /var/lib/serweb/html/functions.php on line 315 >>> >>> # Warning: Cannot modify header information - headers already sent >>> by (output started at /var/lib/serweb/html/functions.php:315) in >>> /var/lib/serweb/html/page.php on line 35 >>> >>> >>> >>> And written with red font, appears this message too: >>> >>> # DB Error: no such table >>> # sorry -- cannot open write fifo >>> >>> >>> I have uncommented the line "fifo_mode=0660" at the /etc/ser/ser.cfg, >>> as it was told here at the list, but it goes wrong. I have changes >>> permissions in /tmp/ser_fifo, but then the web page doesnt get opened... >>> >>> I would appreciate your help in this point, because I suppose that >>> creating user accounts and managing them is the only required step to >>> intercomunicate softphones in a simple LAN, isnt it? >>> >>> Thanks! >>> Pablo >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Serusers] BYE->INVITE
by Xavier TRENTIN 04 Oct '05

04 Oct '05

Hi all, I'd like to implement that kind of scenario: I have 3 user agents UA1 UA2 and UA3 and SER running. UA1 calls UA2. When UA2 sends a BYE request to UA1, I'd like that SER catch this request and initiate an INVITE request from UA1 to UA3, so that UA1 calls now UA3. Is there a way to do that? Thanks. Regards. Xavier

1 0

[Serusers] 483 too many hops
by Chia Huey Lim 04 Oct '05

04 Oct '05

Hi, I am using the standard code for sanity check in the routing script: 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; }; And once I start the SER, after 10 minutes, there are around 10% of the users get "483 Too Many Hops" from SER upon re-registration. Subsequently SER is binding them out as their register sip messages are not received in time. Is there any way to eliminate this issue? Will it be routing script problem? Thanks. Regards, Chia

1 0

[Serusers] Multiple domain
by Anders Hansson 04 Oct '05

04 Oct '05

Hi, Can anyone give me a description how to set up SER 0.8.14 to handle multiple domains. How do I use MySQL subscriber table for authentication and how is location table used for routing. Thanks in advance, Anders

1 0

[Serusers] SER Calling Card Platform
by gorand＠dvvti.com 04 Oct '05

04 Oct '05

Is there a third party application that can be used in conjunction with ser to enable a calling card platform through ser?

2 1

RE: [Serusers] Problem/Question regarding mediaproxy
by Ronald Voermans 04 Oct '05

04 Oct '05

Again, Below is the output of /var/log/messages: Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 192.168.0.146:5004 (RTP) (will return to 192.168.0.146:5004) Sep 26 12:17:57 ser mediaproxy[2876]: session 00-00147-52dbda11-7c108ad21(a)tpsip01.ipvoice.enertel.nl: called signed in from 10.166.38.109:37528 (RTP) (will return to 10.166.38.109:37528) Regards, Ronald ________________________________ Van: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] Namens Ronald Voermans Verzonden: maandag 26 september 2005 8:41 Aan: serusers(a)lists.iptel.org Onderwerp: [Serusers] Problem/Question regarding mediaproxy Hello, I'm using Ser in combination with Mediaproxy. The situation is as follow: My SER has a private ip address (10.254.254.1), and my UAs also have private IP addresses, they can all see SER, and SER can see the UAs. Next, our company has a Voice Interconnect with a telco. (it's using Cirpack as softswitch). SER can see Cirpack and vice versa. However, the UAs cannot see the Cirpack. So I want to use Mediaproxy to overcome this. I can make outbound calls, but when calling inbound, I sometimes have one-way-voice of no voice at all. I think the problem is related to Cirpack, but I have a question thouh: - When calling mediaproxy, how does mediaproxy decide which one is the caller and which one is the called? Is this based on the From and To header? Because, if i call from PSTN -> SIP, i get serveral thousands (!) of messages saying that the called person signed in (no caller signs in in mediaproxy). This could be due to the fact that the from and to headers are incorrect. My ser.cfg is the one from onsip.org (pstn with nat). I can post configs/traces if someone needs them! Regards, Ronald

2 1

[Serusers] SER Server Behind NAT
by Abdul Lateef 04 Oct '05

04 Oct '05

Hi all, Is it possible to use SER server behind NAT with some modification in router. I have my Linux box connected to our Office Lan with SpeedTouch Router, the Router is having public IP Address. I want to install SER in my linux box to make full test. If possible how i can do it? -------- Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

2 1

[Serusers] SER + rtpproxy
by KiFuChan 04 Oct '05

04 Oct '05

Hi all.. My test environment is: SER use publice IP address at 163.25.xxx.xxx User A and B use private IP address at 192.168.1.xxx and 192.168.1.xxx( Publice IP is 220.135.xxx.xxx) While A call B, I check the content of SDP that A send to B have rewrite from "IPv4 192.168.1.xxx" to "IPv4 220.135.xxx.xxx" However, the session still cannot create successful.( I use Windows Messenger to be User Agent) Please give me some advices, thanks! Caxton

1 0

[Users] b2bua + patch
by harry gaillac 03 Oct '05

03 Oct '05

Hello, I tried to compile b2bua (vovida) with patch patch-sip::b2b::AuthAgent.cxx on Debian Sarge 3.1 without success. Thanks for help harry c -o obj.debug.Linux.i686/AuthAgent.o AuthAgent.cxx AuthAgent.cxx:338: ANSI C++ forbids declaration `callFailed' with no type AuthAgent.cxx:338: new declaration `int AuthAgent::callFailed()' AuthAgent.hxx:116: ambiguates old declaration `void AuthAgent::callFailed()' AuthAgent.cxx: In method `int AuthAgent::callFailed()': AuthAgent.cxx:366: warning: control reaches end of non-void function `AuthAgent::callFailed()' AuthAgent.cxx: At top level: AuthAgent.cxx:420: new declaration `void AuthAgent::callFailed()' AuthAgent.cxx:338: ambiguates old declaration `int AuthAgent::callFailed()' AuthAgent.cxx: In method `void AuthAgent::receivedStatus(Vocal::UA::UaBase &, const Sptr<Vocal::SipMsg> &)': AuthAgent.cxx:510: `BT_SERVICE_UNAVAIL' undeclared (first use this function) AuthAgent.cxx:510: (Each undeclared identifier is reported only once AuthAgent.cxx:510: for each function it appears in.) make[1]: *** [obj.debug.Linux.i686/AuthAgent.o] Error 1 make[1]: Leaving directory `/tmp/sip-1.5.0/sip/b2b' make: *** [b2bua] Error 2 ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 2

[Users] Behavioural Difference Question - Regarding SER/OpenSER
by Tavis P 03 Oct '05

03 Oct '05

With SER i had a config with some logic that would, in a certain situation, route a call to itself, eventually timeout and then route the call elsewhere (cancelling the transaction with the initial client). In this situation SER would add One VIA header per message passing through its script (meaning there would be two VIA headers pointing to the SER proxy when the message was sent out), however when the call timed out SER would send the CANCEL message directly to the endpoint instead of using the VIA header fields as one would expect them to be used (ie, first sending the message back to itself and then onto the destination). With OpenSER, using the same config, in the same situation it would send the CANCEL first to itself and then to the endpoint. I was just curious if there is a configuration option that controls this behaviour (it seems an optimization of sorts, if the next via is to myself then skip it) or is it a core difference? Tavis

2 2

[Serusers] Modules writing and overhead.
by sip 03 Oct '05

03 Oct '05

Well, I ended up muddling through the module API long enough to write a module to handle a specific scenario. Some of our UAs have *69 capability (call return here in the US of A), some don't... so I wrote a module that takes data from our ACC log (using a raw query since the regular query doesn't have the LIKE functionality) and finds out who the last user to send an invite to the currently dialing user was and connects the call accordingly. I understand that this may be a little like trimming a mustache with a weedwacker, but the severely limited set of available database manipulation operations available through existing modules made this somewhat useful (to me, anyway). The question really comes down to... what sort of overhead does it incur to do something like this? Granted, not everyone and his brother shall be calling *69 on an even semi-regular basis among my userbase, so it won't even process much... but should I expect the bottleneck to be (as I expect it is) the database server and its query speeds? N.

5 5

[Serusers] windows messenger problems
by syed ahmad 03 Oct '05

03 Oct '05

hello i am mumtaz i want to identify password software plz if u sent the link thanks ok; --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

1 0

[Serusers] How to install "rtpproxy"?
by KiFuChan 03 Oct '05

03 Oct '05

Hi everyone.. I have downloaded the rtpproxy but I don't know how to install it on linux. There are some file which "CVS", " main.c", "Makefile", " Makefile.gnu", " myqueue.h" and "README" in the rtpproxy folder. I input the command "./configure" and "make&make install" which can not work. Please give me a hand, thanks. Caxton

2 1

[Users] accounting missing 'BYE'
by jayesh nambiar 03 Oct '05

03 Oct '05

hi everybody, Can somebody pls help me with the following issue. I collect my CDR using the following query: SELECT t1.from_uri as orig_number, t1.to_uri as term_number, TIMEDIFF(t2.time, t1.time) as duration, t1.timestamp as calldate FROM acc t1, acc t2 WHERE t1.sip_callid = t2.sip_callid AND ((t1.fromtag = t2.fromtag and t1.totag = t2.totag) OR (t1.fromtag = t2.totag and t1.totag = t2.fromtag)) AND t1.sip_method='INVITE' AND t2.sip_method='BYE' This query works fine and gives the call duration between an INVITE and BYE discarding the re-invites. But recently in the peak calling time, I had to restart my openser for some technical reasons, and I think I might have lost some BYEs during this. Can somebody pls help me to extract the calls from acc table which has an INVITE but no BYE. Will be very thanful to you, jayesh. --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

2 1

[Users] checking route return codes
by Tavis P 03 Oct '05

03 Oct '05

i'm having a hard time trying to figure out how i can check the return code on a route, below are some examples of what i've tried. route(20); if ($rc == 1) { exit; } and if (route(20) == 1) { exit; } both generate a "parse error (1469,15-17): syntax error" on startup tavis

2 1

[Users] Softphone can't parse the message from Openser
by Andréa Giordanna 03 Oct '05

03 Oct '05

Hi, I am using avp_db_load("$ruri","i:2/preferences") && avp_pushto("$Contact/reply","i:2") to put a Contact header in the response 3XX to a INVITE message. The message is created and sent, with the new Contact header. But the softphone can't parse the message. What would be the problem? Any suggestions? Thanks, Andrea Giordanna _______________________________________________________ Novo Yahoo! Messenger com voz: ligações, Yahoo! Avatars, novos emoticons e muito mais. Instale agora! www.yahoo.com.br/messenger/

2 1

RE: [Users] TLS
by Helge Waastad 03 Oct '05

03 Oct '05

Hi again, I've attached the core dump. Br hw -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: 29. september 2005 19:37 To: Helge Waastad Cc: users(a)openser.org Subject: Re: [Users] TLS Hi Helge, looks like a double free - since the mem functions used by TLS are embedded, it quite hard to trace it. BTW, do you have the backtrace of the core? it will help until I'm able to reproduce the crash. regards, bogdan Helge Waastad wrote: >Hi, >sometimes when I retstart OpenSER eith tls support, I get: > >BUG: qm_free: freeing already freed pointer, first free: tls/tls_init.c: >ser_free(207) - aborting > > >br hw > > >

2 1

[Users] outgoing packet
by ranveer kunal 03 Oct '05

03 Oct '05

Hi, I have a small problem : I am running my openser on a system(linux/freebsd) with a loopback interface(non routable) with some ip (10.102.4.139), now the incoming packet come to openser with this IP (forwarded in l2 mode), it excepts the packet and sends the reply, but the srcip of replied packet is loopback ip, instead of the routable ip of the system. Can it be configured to send the packet out using the routable ip. Regards, Ranveer.

3 3

[Serusers] Serweb - User page...
by Pablo Cabezas 03 Oct '05

03 Oct '05

Hi! When accessing to the "user management page" this message appears at the top of the page: Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied in /var/lib/serweb/html/functions.php on line 315 Warning: Cannot modify header information - headers already sent by (output started at /var/lib/serweb/html/functions.php:315) in /var/lib/serweb/html/page.php on line 35 Permission troubles... Anything that could help? Thanks!! Pablo

2 1

[Serusers] DB User error...
by Pablo Cabezas 03 Oct '05

03 Oct '05

Hi, I have already installed SER and mySQL into my Debian system, so I have installed serweb to make administration tastks a bit easier ;-) The problem now is that when I try to access to "http://localhost/serweb" and try to log in with the default user/password (or any other xD), it always tells me "DB Error: connect failed" and "Bad username or password". It seems as if there were no possible connection between SER and the mySql service running in the background. I have tried with ser/heslo, admin/heslo... What I am doing wrong? Thanks, Pablo

3 2

[Serusers] i cannot get acc stop message!
by zhangshuai 03 Oct '05

03 Oct '05

Dear all, I use SER 0.8.14, radiusclient 0.4.3 and remote radius server: freeradius 1.0.4. I want radius to do the authentication and accounting. Now I can only receive acc start message and none stop message in /usr/local/var/log/radius/radacct/.. : Mon Oct 3 16:10:22 2005 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 1 User-Name = "41001(a)localhost.localdomain" Calling-Station-Id = "sip:41001@localhost.localdomain" Called-Station-Id = "sip:41002@localhost.localdomain" Sip-Translated-Request-URI = "sip:41002@202.204.118.219:5060" Acct-Session-Id = "D7046CD5-B8EB-4F17-A926-EE7F4A4C52FB(a)218.97.252.37" Sip-To-Tag = "523922641" Sip-From-Tag = "3394148029" Sip-Cseq = "9206" NAS-IP-Address = 218.97.252.34 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 218.97.252.34 Acct-Unique-Session-Id = "3b5c2410e3d50ac7" Timestamp = 1128370222 How come? Did anyone know this problem? Here is my ser.cfg: # ----------- 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) listen=218.97.252.34 #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/acc.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/uri_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) # -- 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 params -- modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "early_media", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "radius_missed_flag", 3) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "radius_missed_flag", 3) # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "radius_flag", 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")) { log(1,"LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { log(1,"LOG: Message too big"); 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()) { if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n"); setflag(1); }; 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=~"localhost.localdomain") { if (method=="REGISTER") { log(1, "REGISTER: Authenticating user\n"); # Uncomment this if you want to use digest authentication if (!radius_www_authorize("localhost.localdomain")) { log(1, "REGISTER: challenging user\n"); www_challenge("localhost.localdomain", "0"); break; }; save("location"); break; }; if (method=="INVITE") { log(1, "INVITE MESSAGE RECEIVED - START ACCOUNTING\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { log (1, "%method - STOP ACCOUNTING\n"); setflag(1); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; setflag(3); # 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(); }; } Many thanks for your help! Shuai

1 0

[Serusers] Error: I won't send a reply for ACK
by Fernando Patzlaff 03 Oct '05

03 Oct '05

Hi all, I have problem with my SER. It's work fine on the version 0.8.14. I upgrade its to 0.9.4, but when I make a call from a gateway to another gateway bypassing the SER the call is ended in 15 seconds. The message appears on tethreal is: [root@sv02 ser]# tethereal -R sip Capturing on eth0 11.426429 10.128.16.3 -> 10.128.0.11 SIP/SDP Request: INVITE sip:3354@10.128.0.11:5070, with session description 11.428174 10.128.0.11 -> 10.128.16.3 SIP Status: 401 Unauthorized 11.505784 10.128.16.3 -> 10.128.0.11 SIP Request: ACK sip:3354@10.128.0.11:5070 11.528044 10.128.16.3 -> 10.128.0.11 SIP/SDP Request: INVITE sip:3354@10.128.0.11:5070, with session description 0(3732) -> encaminhando para o gv01 asdasdasd 11.531864 10.128.0.11 -> 10.128.16.3 SIP Status: 100 trying -- your call is important to us 11.532865 10.128.0.11 -> 10.128.0.2 SIP/SDP Request: INVITE sip:3354@10.128.0.11:5070, with session description 11.591058 10.128.0.2 -> 10.128.0.11 SIP Status: 100 Trying 13.129865 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 183 Session Progress, with session description 13.130850 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 183 Session Progress, with session description 13.242219 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 180 Ringing, with session description 13.242898 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 180 Ringing, with session description 13.257439 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 200 OK, with session description 13.258149 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 200 OK, with session description 0(3732) ACC: transaction answered: method=INVITE, i-uri=sip:3354@10.128.0.11:5070, o-uri=sip:3354@10.128.0.11:5070, call_id=call-805E8372-0630-DA11-0214-E(a)10.128.16.3, from=<sip:10.128.16.3>;tag=a801003-12, code=200 13.406006 10.128.16.3 -> 10.128.0.11 SIP Request: ACK sip:10.128.0.11:5070;ftag=a801003-12;lr=on 0(3732) ACC: request acknowledged: method=ACK, i-uri=sip:10.128.0.11:5070;ftag=a801003-12;lr=on, o-uri=sip:10.128.0.11:5070;ftag=a801003-12;lr=on, call_id=call-805E8372-0630-DA11-0214-E(a)10.128.16.3, from=<sip:10.128.16.3>;tag=a801003-12, code=200 0(3732) Warning: sl_send_reply: I won't send a reply for ACK!! 13.755012 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 200 OK, with session description 13.756575 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 200 OK, with session description 14.755003 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 200 OK, with session description 14.756304 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 200 OK, with session description 16.755340 10.128.0.2 -> 10.128.0.11 SIP/SDP Status: 200 OK, with session description 16.756279 10.128.0.11 -> 10.128.16.3 SIP/SDP Status: 200 OK, with session description 45.262004 10.128.0.2 -> 10.128.0.11 SIP Request: BYE sip:10.128.0.11:5070;ftag=a801003-12;lr=on 45.262719 10.128.0.11 -> 10.128.16.3 SIP Request: BYE sip:10.128.16.3 45.336383 10.128.16.3 -> 10.128.0.11 SIP Status: 200 OK 45.336871 10.128.0.11 -> 10.128.0.2 SIP Status: 200 OK Do they have an idea? Thanks, Fernando Patzlaff

2 1

[Serusers] preload_udomain Error
by Ryan Pagquil 03 Oct '05

03 Oct '05

Hi guys, I got these errors whenever I restarted SER. What is the meaning of these? 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 123 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 108 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 100 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 130 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 104 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 124 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 118 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 128 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 112 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 111 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 110 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 109 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad contact record in table aliases 0(0) preload_udomain: ERROR: for username 132 Could this be the problem with my aliases that are not been used by SER? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

2 8

[Serusers] Location of MSN 4.7 ~ 5
by Ronald Wiplinger 01 Oct '05

01 Oct '05

Can any point me to a location, where these older MSN still available? Which one would you suggest to use for SER tests? bye Ronald Wiplinger

2 2

FW: [Serusers] ser and * voicemail
by Rick Thompson 01 Oct '05

01 Oct '05

I have tethereal on my ser server but didn't see a 408. _____ From: Leon Sun [mailto:leon.sun@keywestcommunications.com] Sent: Friday, September 30, 2005 5:41 PM To: 'Leon Sun'; rthompson(a)vir2com.com Cc: 'Mark Aiken' Subject: RE: [Serusers] ser and * voicemail Hi, guys, I used Ethereal to trace SIP and found code 408 when it's time out. I believe we need focus 408 rather than 487. Sorry Rick, for last e-mail. Regards Leon _____ From: Leon Sun [mailto:leon.sun@keywestcommunications.com] Sent: Friday, September 30, 2005 2:33 PM To: 'rthompson(a)vir2com.com' Cc: 'Mark Aiken' Subject: RE: [Serusers] ser and * voicemail >From Mark: SER normally sends a 487 when the INVITE timer runs out so you would need to trigger the voicemail on that event. We have our feature server handing timeouts not SER (our SER timeout is set to a very large value) so I'm not sure the best way to proceed. Does SER call the failure_route for a locally generated 487 timeout? If so then rather than the 'break' you have there now just forward to vm. I would set a different t_on_failure instead of reusing "1" though, so you dont keep forwarding if the vm fails. Mark _____ From: Rick Thompson [mailto:rthompson@vir2com.com] Sent: Friday, September 30, 2005 2:27 PM To: 'Leon Sun' Subject: RE: [Serusers] ser and * voicemail Leon Thanks for the reply to my post I see what you are doing here, but that part has been taken care of. When I get a failure on "INVITE" of 486, 404, 408 or 480 I look in the mysql database and find the voicemail box for that uri and replace the sip contact then prefix it with a "V" and relay it to asterisk. Over on the asterisk side, I route the call to the voicemail context and delete the "V" and send it to voicemail. All this works except when ser times out on the "INVITE" and for some reason it doesn't route to asterisk. That's the problem I'm having. Rick _____ From: Leon Sun [mailto:leon.sun@keywestcommunications.com] Sent: Friday, September 30, 2005 4:36 PM To: rthompson(a)vir2com.com; serusers(a)lists.iptel.org Subject: RE: [Serusers] ser and * voicemail Rick, I had same problem before and I gave it up since I didn't get any answer from list. I am using another way(tricky but working) to do voice mail. Hope it can help you if you can't fix it. 1. Check location in your routing parts before relay, if not, forward to Asterisk. 2. set up unconditional forward in ATA as 00 + ATA'DID. Make a route in SER and point 00* to Asterisk. 3. Strip 00 in Asterisk and send it to voicemail2(${EXTEN}) Regards Leon Sun _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Rick Thompson Sent: Friday, September 30, 2005 12:58 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser and * voicemail Hi All I'm working with a ser script, written by someone that's gone now, that routes inbound calls to an asterisk server for voicemail. The failure_route[1], sends calls to asterisk and the IVR plays if the ua is unreachable (not in location) "404", "408" or the ua is busy "486" but it doesn't when the inv time exceeds 30 sec (rings for 30 sec or more). The call just stops ringing and 10 sec later gets a fast busy. Any ideas from anyone would be greatly appreciated. Here is the code I'm working with. failure_route[1] { xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm> <%rr> <%rs>\n"); if(t_check_status("487")) { break; }; if(method=="INVITE" && (t_check_status("486|408|404|480"))) { if(avp_db_load("$ruri", "s:mailbox")) avp_pushto("$ruri/username", "s:mailbox"); prefix("V"); rewritehostport("A.B.C.D:5060"); append_branch(); xlog("L_ALERT", "****** Transfering to Voicemail\n"); t_on_reply("1"); t_relay(); }; } Thanks Rick

2 1

[Serusers] SER Operator Pack lisence?
by Roy Sigurd Karlsbakk 01 Oct '05

01 Oct '05

hi i just came by http://www.iptelorg.de/SOP.php, and it's quite clearly SER, but then, on the downloads site, it tells me "This site offers only commercial products and services . Please follow the links below if you wish to find support related to respective software". although their stuff looks good, can you really make GPLed code a "commercial product"? roy

2 1

FW: [Serusers] ser and * voicemail
by Rick Thompson 30 Sep '05

30 Sep '05

Mark Thanks for the re-post What I see in ngrep and var/log/messages is a "CANCEL" from the ua and a failure "487" on a reply. Don't know which happens first. Even if I take out the "break" if and add "487" to the if(method== statement, it goes to the asterisk server but vm doesn't play, no audio, and it drops the call anyway. I think I'll try what you suggested about creating another t_on_failure[] and test for it within the main routing block. Rick _____ From: Mark Aiken [mailto:aiken.mark@gmail.com] Sent: Friday, September 30, 2005 5:22 PM To: Leon Sun Cc: rthompson(a)vir2com.com; serusers(a)lists.iptel.org Subject: Re: [Serusers] ser and * voicemail SER normally sends a 487 when the INVITE timer runs out so you would need to trigger the voicemail on that event. We have our feature server handing timeouts not SER (our SER timeout is set to a very large value) so I'm not sure the best way to proceed. Does SER call the failure_route for a locally generated 487 timeout? If so then rather than the 'break' you have there now just forward to vm. I would set a different t_on_failure instead of reusing "1" though, so you dont keep forwarding if the vm fails. Mark On 9/30/05, Leon Sun <leon.sun(a)keywestcommunications.com> wrote: Rick, I had same problem before and I gave it up since I didn't get any answer from list. I am using another way(tricky but working) to do voice mail. Hope it can help you if you can't fix it. 1. Check location in your routing parts before relay, if not, forward to Asterisk. 2. set up unconditional forward in ATA as 00 + ATA'DID. Make a route in SER and point 00* to Asterisk. 3. Strip 00 in Asterisk and send it to voicemail2(${EXTEN}) Regards Leon Sun _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org <mailto:nces@iptel.org> ] On Behalf Of Rick Thompson Sent: Friday, September 30, 2005 12:58 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser and * voicemail Hi All I'm working with a ser script, written by someone that's gone now, that routes inbound calls to an asterisk server for voicemail. The failure_route[1], sends calls to asterisk and the IVR plays if the ua is unreachable (not in location) "404", "408" or the ua is busy "486" but it doesn't when the inv time exceeds 30 sec (rings for 30 sec or more). The call just stops ringing and 10 sec later gets a fast busy. Any ideas from anyone would be greatly appreciated. Here is the code I'm working with. failure_route[1] { xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm> <%rr> <%rs>\n"); if(t_check_status("487")) { break; }; if(method=="INVITE" && (t_check_status("486|408|404|480"))) { if(avp_db_load("$ruri", "s:mailbox")) avp_pushto("$ruri/username", "s:mailbox"); prefix("V"); rewritehostport("A.B.C.D:5060"); append_branch(); xlog("L_ALERT", "****** Transfering to Voicemail\n"); t_on_reply("1"); t_relay(); }; } Thanks Rick _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] ser and * voicemail
by Rick Thompson 30 Sep '05

30 Sep '05

Hi All I'm working with a ser script, written by someone that's gone now, that routes inbound calls to an asterisk server for voicemail. The failure_route[1], sends calls to asterisk and the IVR plays if the ua is unreachable (not in location) "404", "408" or the ua is busy "486" but it doesn't when the inv time exceeds 30 sec (rings for 30 sec or more). The call just stops ringing and 10 sec later gets a fast busy. Any ideas from anyone would be greatly appreciated. Here is the code I'm working with. failure_route[1] { xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm> <%rr> <%rs>\n"); if(t_check_status("487")) { break; }; if(method=="INVITE" && (t_check_status("486|408|404|480"))) { if(avp_db_load("$ruri", "s:mailbox")) avp_pushto("$ruri/username", "s:mailbox"); prefix("V"); rewritehostport("A.B.C.D:5060"); append_branch(); xlog("L_ALERT", "****** Transfering to Voicemail\n"); t_on_reply("1"); t_relay(); }; } Thanks Rick

3 2

[Serusers] loose routing of BYE from UDP to TCP
by Mark Aiken 30 Sep '05

30 Sep '05

Hi, I'm having problems getting SER to (loose) route requests from UDP to TCP. I'm not sure if the UA is at fault here or some SER config issue. SER receives an INVITE over TCP, record_route() is called and then the request is relayed via UDP (t_relay_to_udp) to the UA. The INVITE relayed to the UDP UA now has 2 Record-Route headers added by SER. One has the transport=tcp parameter. Record-Route: <sip:xx.xx.xx.xx;r2=on;ftag=xyz;lr=on> Record-Route: <sip:xx.xx.xx.xx;transport=tcp;r2=on;ftag=xyz> When the UA sends the BYE to SER, it has the 2 Route headers like so (on a single line): Route: <sip:xx.xx.xx.xx;r2=on;ftag=xyz;lr=on>,<sip:xx.xx.xx.xx ;transport=tcp;r2=on;ftag=xyz;lr=on> The SER script simply calls t_relay() in the loose_route section of the script. I expected t_relay() to relay the BYE via TCP, since the Route header has transport=tcp, but it sends the BYE via UDP instead. Anyone seen this problem before? Do I need to check for transport=tcp and call t_relay_to_tcp, rather than using t_relay after loose_route()? Mark

5 13

Re: [Serusers] Windows to Linux
by zkeatts 30 Sep '05

30 Sep '05

I am probably not the best person to ask that, since I have been brute forcing my way through - i.e. reading and debugging code without consulting a manual. There are tons of resources on the net for coding in C and since you already have 3 years of experience I am sure that moving to ANSI C won't be too hard. If you have coded in JAVA or C++ then it should be a breeze. If anything you should read up on pointers and memory allocation, because those two seem to catch most new coders the most. The GDB suggestion is also excellent. I have found it to be one of the most useful debugging tools that I have ever used since it gives a nice stack backtrace with line numbers etc. good luck, Zach Keatts Software Engineer Nuvio.com viquar syed wrote: > Hey Zach, > First off, thanks for your reply. Yours is the first one I got beside > the OUT OF OFFICE notices. > I plan on getting the installation up. I want to get down to Stack > level, but I guess learning ANSI C would be my first step, have your > foud anything useful towards this ? > Again I appreciate your input. > > Thanks > Viq > DBA/Developer > Cordia Communications > Orlando FL > 407-313-7015 > > On 9/30/05, *zkeatts* <zkeatts(a)nuvio.com <mailto:zkeatts@nuvio.com>> > wrote: > > I am in a similar situation, as far as the ANSI C goes. What has > helped > me is to jump in, find a bug and try to figure out what is going on. > Granted you might not be as lucky as me on your first try in finding > bugs, but trust me even if you don't there is plenty to look > at. I can > promise you it will be confusing and infuriating, but slowly > things will > begin to make sense. > > Start by setting up a basic SER installation on a Linux box. Once you > get it set up and working then you can try out different features. > Don't be afraid to send messages to this group, I have found them > very > helpful and attentive to all levels of questions. > > good luck, > > Zach Keatts > Software Engineer > Nuvio.com <http://Nuvio.com> > > viquar syed wrote: > > > Hi, > > My company wants me to hop on to Linux/Unix from Bill Gates, and to > > make it even better I am suppose to get upto speed writing > modules to > > enhance our SER deployment. > > What experience do I have writing in ANSI C "0", "ZERO", "nada", > > BUT, I have been programming for over three yrs now so the logic is > > not an issue. > > My question, whats the best way to start, I am currently going > through > > the documentation on SER Admin, need to have a plan. > > I know no one likes newbees, but hey any help would be highly > appreciated. > > Thanks > > Viq > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

1 0

[Serusers] Windows to Linux
by viquar syed 30 Sep '05

30 Sep '05

Hi, My company wants me to hop on to Linux/Unix from Bill Gates, and to make it even better I am suppose to get upto speed writing modules to enhance our SER deployment. What experience do I have writing in ANSI C "0", "ZERO", "nada", BUT, I have been programming for over three yrs now so the logic is not an issue. My question, whats the best way to start, I am currently going through the documentation on SER Admin, need to have a plan. I know no one likes newbees, but hey any help would be highly appreciated. Thanks Viq

3 2

[Serusers] rtpproxy and B2BUA
by Nhadie 30 Sep '05

30 Sep '05

Hi, I read somewhere that B2BUA will helpm with regards to some BYE being absent sometimes. Would an rtpproxy do the same job as a B2BUA as far as making sure the BYE won't get lost? TIA Regards, Nhadie

1 0

Re: [Serusers] serweb question
by Karel Kozlik 30 Sep '05

30 Sep '05

HI, first: please (CC) the mailing list in order to others can see answers. Try enable logging in serweb (config variable $config->enable_loging = true) and see the log file. There are more descriptive errror messages. You may also try set log level to debug. Karel Nhadie napsal(a): > I also get this error when I was able to logged under > http://www.mydomain.com/serweb > > My Account > DB Error: unknown error, file: > /usr/local/serweb/application_layer/apu_aliases.php:112 > > Karel Kozlik wrote: > >> Hello, >> are you able to check content of your mysql tables? Check table >> admin_privileges if contain these records: >> >> username | domain | priv_name | priv_value >> ------------------------------------------------------- >> admin | your.domain | is_admin | 1 >> admin | your.domain | change_privileges | 1 >> >> >> Karel >> >> Nhadie napsal(a): >> >>> Hi All, >>> >>> I can login using admin and other users when I'm at >>> http://www.mydomain.com/serweb/ >>> but when I use the login interface at >>> http://www.mydomain.com/serweb/admin I can't login as admin/heslo >>> >>> What could I be missing here? TIA >>> >>> Regards, >>> Nhadie >>> >> >> > >

2 2

[Serusers] Errors during startup
by Ronald Wiplinger 30 Sep '05

30 Sep '05

Sep 29 19:44:11 dns ser: WARNING: fix_socket_list: could not rev. resolve 192.168.0.254 Sep 29 19:44:11 dns ser: WARNING: fix_socket_list: could not rev. resolve 192.168.0.254 Sep 29 19:44:11 dns /usr/local/sbin/ser[19346]: Maxfwd module- initializing Sep 29 19:44:11 dns /usr/local/sbin/ser[19346]: new_connection(): Access denied for user: 'serro@localhost' (Using password: YES) Sep 29 19:44:11 dns /usr/local/sbin/ser[19346]: db_init(): Could not create a connection Sep 29 19:44:11 dns /usr/local/sbin/ser[19346]: ERROR: auth_db_ver: unable to open database connection Sep 29 19:44:11 dns /usr/local/sbin/ser[19346]: auth_db:str_fixup(): Error while querying table version Sep 29 19:44:11 dns /usr/local/sbin/ser[19346]: ERROR: fix_expr : fix_actions error Q: 1. Where do the Warning 192.168.0.254 come from? My network does not use 192.168.0.x 2. Access denied is most likely the error for all following errors. Note the quote marks, they are around the entire name and not 'serro'@'localhost' I am not sure if that makes a difference. The access to the database is for me not clear in the moment. We have a user ser with password serpassword and a user serro with password serropassword. We have also admin with the password adminpassword. I don't find the user/passwords in the config files. My ser.cfg looks like that: grep -v ^[$,#] ser.cfg /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://ser:serpassword@localhost/ser") modparam("auth", "secret", "myphrase") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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 if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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==trace-intl.net) { if (method=="REGISTER") { if (!www_authorize("trace-intl.net", "subscriber")) { www_challenge("trace-intl.net", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

2 2

[Serusers] ser web
by Elad Dotan 30 Sep '05

30 Sep '05

Hi, I've installed 4 RPM files on a Fedora CORE 3. The RPM are: ser-0.9.3-0.i386.rpm php-pecl-classkit-0.4-2.i386.rpm ser-mysql-0.9.3-0.i386.rpm serweb-0.9.3-0.i386.rpm I enabled the httpd and mysql service. I'm not sure what to do next. All the Documentation I have is related to source flies and not RPM. Ser is working but I don't know how to configure the serweb in order for it to work. I tried: http:// but I get an error. where can I find documentation about ser web? Thanks, Elad.

3 2

[Serusers] Branch result code selection during serial forking - OpenSER does it?
by Arek Bekiersz 30 Sep '05

30 Sep '05

Hi, I've once asked Serusers list about problem with sequential forking and failure_routes[]. SER pickes lower result code from branches. This is how "tm" module is done (t_check_status(), ...). Mail was from 05 September 2005, titled: "t_check_status() - why it is checking for winning reply in failure_routes?" Received many answers, thanks for it. I was responded by Klaus Darilion on 06/09 that SER policy is kept according to RFC - we pick LOWEST response code. So it is impossible in example to make sequential (serial) routing based on "486" to third branch, when first branch responded "404" and second branch responded "486". There will always be "404". I was thinking yesterday about writing just another alternative function for t_check_status() and t_pick_branch() in tm module, to do what I need. That is: checking if error code is something useful, that tells me what happened downstream in current branch (busy? timed out?). But today I was reading this article on OpenSER: http://openser.org/pipermail/users/2005-June/000004.html Seems it is possible to use OpenSER to make sequential forking with more than one failure_route[]? There is fix to tm module that pickes result only from last set of forked branches? Am I correct? Thanks for comments. -- Regards, Arek Bekiersz

1 0

[Serusers] Re: Re: please help me about how to config radiusclient!
by zhangshuai 30 Sep '05

30 Sep '05

Dear Ludovic Zipp-Fuhrer, Thank you very much!. They begin to work! And now I can move on with acc. >In radiusclient.conf you have to set the bind parameters with the IP of >your client. I got similar errors because I don't set this parameter >correctly. I hope it helps you. > > >zhangshuai a écrit : >> Dear Ludovic Zipp-Fuhrer, >> >> >> What do you mean exactly? Whose configuration files? How to check it? >> >> The number "DA61FC29" in the error message is the hexadecimal value of the ip address of radius server. >> >> >> >> >>>did you set the good bind addresses in the configuration files? >>> >>> >>> >>>zhangshuai a écrit : >>> >>>>Hi all, >>>> >>>> >>>>I wish to use softphone X-lite on SER and radius in db_mode. When I installed both freeradius 1.0.4 and radiusclient 0.4.3 on the same server which ser 0.8.14 has installed on, I could log in X-lite through radius authorization. And now I want radiusclient (on server A) to send the request to a remote radius server (on server B) to realize authorization. I have changed those config files: /usr/local/etc/radiusclient/servers, /usr/local/etc/radiusclient/radiusclient.conf and /usr/local/etc/raddb/clients.conf as the document (http://www.iptel.org/ser/doc/ser_radius/ser_radius.html#AEN193) did. Then I test the radius server as the above document said on server A: radclient -f digest server B auth testing123 and I received expected replied message from server B. >>>> >>>>But when I log in x-lite, there are error messages: >>>> >>>>Sep 29 17:11:06 localhost ser[4408]: rc_ip_hostname: couldn't look up host by addr: DA61FC29 >>>>Sep 29 17:11:06 localhost ser[4408]: rc_send_server: no reply from RADIUS server unknown:1812 >>>> >>>>How come? Did I mis-config anything? >>>> >>>>Many thanks for your reply! >>>> >>>> >>>>Shuai >>>>http://www.goldentek.biz >>>> >>>> >>>>_______________________________________________ >>>>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 > = = = = = = = = = = = = = = = = = = = =

1 0

[Serusers] No accounting records logged by radius in mysql; Unsupported Acct-Status-Type = 15
by Cameron Beattie 30 Sep '05

30 Sep '05

I have configured ser with radius accounting and I see accounting records being generated successfully in the file system. I have also configured accounting to mysql. Successful calls are logged to mysql correctly but unsuccessful calls are not. When I run radius -X and watch what happens I see the following: rlm_sql (sql): Unsupported Acct-Status-Type = 15 modcall[accounting]: module "sql" returns noop for request 1 dictionary.ser contains the following line: VALUE Acct-Status-Type Failed 15 # RFC2866, acc I have added the contents of dictionary.ser to the radiusclient/dictionary and radiusclient-ng/dictionary files. I have included dictionary.ser in raddb/dictionary. I'm sure there's a configuration step I've missed. Could anyone suggest some other things to check? Thanks Cameron

5 4

[Serusers] SERWEB for LInux
by Kashif Meraj 30 Sep '05

30 Sep '05

HI all from where i can download WEBSER for LInux box. REgards -- Kashif Meraj

3 2

[Serusers] Proxy Behaviour on a parallel fork
by jim.pafford＠comcast.net 30 Sep '05

30 Sep '05

In ser 0.9.3 I am seeing the following problem with a parallel forking scenario. The SER is sending the final Request ACK to the wrong location every time. Is there a way to fix this. See below: Endpoint (xyz(a)1.1.1.1) sends Invite(abc123(a)proxy.com) to the SER. SER looks up location and then sends the following three messages: TRYING back to (xyz(a)1.1.1.1) INVITE to abc123(a)2.2.2.2 INVITE to abc123(a)3.3.3.3 This looks good so far. SER then gets back ringing from both endpoints and sends along to xyz(a)1.1.1.1 abc123(a)3.3.3.3 answers the call and sends back OK SER then sends OK to xyz(a)1.1.1.1 - Still good SER then sends CANCEL to abc123(a)2.2.2.2 abc123(a)2.2.2.2 responds with 200 Canceling - so far so good. Now xyz(a)1.1.1.1 sends the ACK to (abc123(a)proxy.com) - Still looks good. But now after looking up location for abc123 SER sends the ACK to the wrong endpoint abc123(a)2.2.2.2 What am I doing wrong? SER always sends the ACK back to the first address in the list as shown by serctl ul show abc123. Is there a way to correct this so that SER knows the correct endpoint to relay the ACK to? Seems like it should understand which endpoint sent back the OK to the original INVITE and then send the ACK to that endpoint and not the first one in the list after a location lookup. thanks, Jim

4 3

Re: [Serusers] please help me about how to config radiusclient!
by zhangshuai 30 Sep '05

30 Sep '05

Dear Ludovic Zipp-Fuhrer, What do you mean exactly? Whose configuration files? How to check it? The number "DA61FC29" in the error message is the hexadecimal value of the ip address of radius server. >did you set the good bind addresses in the configuration files? > > > >zhangshuai a écrit : >> Hi all, >> >> >> I wish to use softphone X-lite on SER and radius in db_mode. When I installed both freeradius 1.0.4 and radiusclient 0.4.3 on the same server which ser 0.8.14 has installed on, I could log in X-lite through radius authorization. And now I want radiusclient (on server A) to send the request to a remote radius server (on server B) to realize authorization. I have changed those config files: /usr/local/etc/radiusclient/servers, /usr/local/etc/radiusclient/radiusclient.conf and /usr/local/etc/raddb/clients.conf as the document (http://www.iptel.org/ser/doc/ser_radius/ser_radius.html#AEN193) did. Then I test the radius server as the above document said on server A: radclient -f digest server B auth testing123 and I received expected replied message from server B. >> >> But when I log in x-lite, there are error messages: >> >> Sep 29 17:11:06 localhost ser[4408]: rc_ip_hostname: couldn't look up host by addr: DA61FC29 >> Sep 29 17:11:06 localhost ser[4408]: rc_send_server: no reply from RADIUS server unknown:1812 >> >> How come? Did I mis-config anything? >> >> Many thanks for your reply! >> >> >> Shuai >> http://www.goldentek.biz >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)iptel.org >> http://mail.iptel.org/mailman/listinfo/serusers >> >

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Re: [Serusers] Proxy Behaviour on a parallel fork
by jim.pafford＠comcast.net 29 Sep '05

29 Sep '05

Thank you very much. You are absolutely correct. Removed the lookup and relied on loose-route and problem is solved. Jim -------------- Original message -------------- > > > jim.pafford(a)comcast.net wrote: > > In ser 0.9.3 I am seeing the following problem with a parallel forking > > scenario. The SER is sending the final Request ACK to the wrong location > > every time. Is there a way to fix this. See below: > > > > Endpoint (xyz(a)1.1.1.1 ) sends > > Invite(abc123(a)proxy.com ) to the SER. > > SER looks up location and then sends the following three messages: > > TRYING back to (xyz(a)1.1.1.1 ) > > INVITE to abc123(a)2.2.2.2 > > INVITE to abc123(a)3.3.3.3 > > > > This looks good so far. > > SER then gets back ringing from both endpoints and sends along to > > xyz(a)1.1.1.1 > > > > abc123(a)3.3.3.3 answers the call and sends back OK > > SER then sends OK to xyz(a)1.1.1.1 - Still good > > SER then sends CANCEL to abc123(a)2.2.2.2 > > abc123(a)2.2.2.2 responds with 200 Canceling - so > > far so good. > > > > Now xyz(a)1.1.1.1 sends the ACK to (abc123(a)proxy.com > > ) - Still looks good. > > > > But now after looking up location for abc123 SER sends the ACK to the > > Do you use lookup(location) for ACK? This is not necessary. It should be > handled in loose_route section. > > klaus > > > > wrong endpoint abc123(a)2.2.2.2 > > > > What am I doing wrong? SER always sends the ACK back to the first > > address in the list as shown by serctl ul show abc123. Is there a way > > to correct this so that SER knows the correct endpoint to relay the ACK > > to? Seems like it should understand which endpoint sent back the OK to > > the original INVITE and then send the ACK to that endpoint and not the > > first one in the list after a location lookup. > > > > thanks, > > Jim > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers

1 0

RE: [Serusers] multi-domain support with one SER-2
by Lambert, Micheline (Satnet) 29 Sep '05

29 Sep '05

Yes, I have added two domains into domain table. I created them with serctl domain add >From the database ser, it is there. mysql>select * from domain; +----------------+---------------------+ | domain | last_modified | +----------------+---------------------+ | 192.168.20.155 | 0000-00-00 00:00:00 | | 192.168.20.55 | 0000-00-00 00:00:00 | +----------------+---------------------+ 2 rows in set (0.00 sec) mysql> from the serctl command, I have this error: #serctl domain show 500 command 'domain_dump' not available # Micheline -----Original Message----- From: Alberto [mailto:alberto.ipt@telefonica.net] Sent: Thursday, September 29, 2005 9:50 AM To: Lambert, Micheline (Satnet); serusers(a)lists.iptel.org Subject: Re: [Serusers] multi-domain support with one SER-2 Hi, I am new user too. Have you add twos domains into "domain" table? Regards -- Alberto ----- Original Message ----- From: "Lambert, Micheline (Satnet)" <Lambert.M(a)emssatnet.com> To: <serusers(a)lists.iptel.org> Sent: Thursday, September 29, 2005 3:21 PM Subject: [Serusers] multi-domain support with one SER-2 > Hi, > > > I'm new with SER since August 2005. > > > I'm using ser v0.9.3 and mysql v4.1.13. > > > I use SER with two domains: 192.168.20.55 and 192.168.20.155. > > Some users are configured in subscribe table of SER database with domain > 192.168.20.55 and > > other users are configured in the same subscribe table of SER database > with > domain 192.168.20.155. > > > > I understand that user from domain 192.168.20.55 can not talk to users in > domain 192.168.20.155. Is it right? > > > > I have the following problem: > > Users from domain 192.168.20.55 can make calls to PBX/PSTN but users from > domain 192.168.20.155 can not, > > There is the message 403 Forbidden sent to the users. > > > > Here is the sequence: > > <- INVITE > > -> 407 Proxy Authentication Required > > <-ACK > > <-INVITE > > ->100 Trying > > ->403 Forbidden > > <-ACK > > ->403 Forbidden > > <-ACK > > ->403 Forbidden > > <-ACK > > > > I would like to know if there is a need to have two databases for the > multi-domain support with one SER? > > > Micheline Lambert > > I included a part of my ser.cfg file > > > > # > > # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ > > # > > # simple quick-start config script > > # > > > > ... > > > > listen=192.168.20.55 > > listen=192.168.20.155 # support multi-domains > > > > > > # ------------------ 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/textops.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/uri_db.so" > > > > # load the group module to use : is_useer_in() function > > loadmodule "/usr/local/lib/ser/modules/group.so" > > > > # load the acc module ffor accounting > > loadmodule "/usr/local/lib/ser/modules/acc.so" > > > > #new module for multi-domain support > > loadmodule "/usr/local/lib/ser/modules/domain.so" > > > > # ----------------- setting module-specific parameters --------------- > > > > # -- usrloc params -- > > > > > > modparam("auth_db|uri_db|usrloc", "db_url", > "mysql://ser:heslo@localhost/ser") > > > > modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") > > > > modparam("usrloc|registrar", "use_domain", 1) > > > > # -- auth params -- > > # Uncomment if you are using auth module > > # > > modparam("auth_db", "calculate_ha1", 1) > > # > > # 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_mode", 0) > > # Uncomment this if you want to use SQL database > > # for persistent storage and comment the previous line > > modparam("usrloc", "db_mode", 2) > > > > > > # -- rr params -- > > # add value to ;lr param to make some broken UAs happy > > modparam("rr", "enable_full_lr", 1) > > > > > > # -- acc params -- > > # set the reporting log level > > modparam("acc", "log_level", 1) > > # number of fflag which will be used fffor accounting; if a message > > # is labeled with this flag, its completion status will be reported > > modparam("acc", "log_flag", 1) > > > > modparam("acc", "db_flag", 1) > > modparam("acc", "report_cancels", 1) > > > > # ------------------------- request routing logic ------------------- > > > > # main routing logic > > > > route{ > > > > # ------------------------------------------------------------------ > > # Sanity Check Section > > # > ------------------------------------------------------------------ > > # initial sanity checks -- messages with > > # max_forwards==0, or excessively long requests > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483","Too Many Hops"); > > break; > > }; > > if (msg:len >= 2048 ) { > > sl_send_reply("513", "Message too big"); > > break; > > }; > > > > > > # ------------------------------------------------------------------ > > # labeled all transactions for accounting > > # > ------------------------------------------------------------------ > > log(1, "++++ labeled all transactions for accounting\n"); > > setflag(1); > > > > # ------------------------------------------------------------------ > > # Record Route Section > > # > ------------------------------------------------------------------ > > # 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 > > if (!method=="REGISTER") record_route(); > > > > # ------------------------------------------------------------------ > > # Loose Route Section > > # > ------------------------------------------------------------------ > > # subsequent messages withing a dialog should take the > > # path determined by record-routing > > if (loose_route()) { > > # mark routing logic in request > > append_hf("P-hint: rr-enforced\r\n"); > > route(1); > > break; > > }; > > > > # ------------------------------------------------------------------ > > # Call Type Processing > > # > ------------------------------------------------------------------ > > if (!uri==myself) { > > # mark routing logic in request > > append_hf("P-hint: outbound uri!=myself\r\n"); > > route(1); > > 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=="INVITE") { > > route(3); > > break; > > } else if (method=="REGISTER") { > > route(2); > > break; > > }; > > > > lookup("aliases"); > > if (!uri==myself) { /* myself = my IP address */ > > append_hf("P-hint: outbound alias - > uri!=myself\r\n"); > > route(1); /* relay the message */ > > break; > > }; > > > > log(1, "++++ route other SIP > messages???????????????\n"); > > route(1); > > }; > > } > > > > route[1] > > { > > if (!t_relay()) { > > sl_reply_error(); > > }; > > } > > > > > > route[2] > > { > > sl_send_reply("100", "Trying"); > > > > log(1, "==== www_authorize\n"); > > # validate the authentication of the user > > if (!www_authorize("", "subscriber")) { > > log(1, "==== send 401 Unauthorized\n"); > > www_challenge("", "0"); # send back 401 > Unauthorized message > > break; # ask for another > registration with auth. > > }; > > > > if (!check_to()) { > > sl_send_reply("401", "Unauthorized 2"); > > break; > > }; > > > > log(1, "==== remove Authorization line\n"); > > consume_credentials(); # remove Authorization digest info from > message > > > > # save the user in the location table > > log(1, "==== save location and send Ok\n"); > > if (!save("location")) { # save the user in the location table, > send Ok message > > sl_reply_error(); > > }; > > > > } > > > > > > route[3] > > { > > if (!uri=~"sip:\+?[0-9]+@.*") { > > log(1, "---- call cannot be serverd here - not > numeric uri\n"); > > sl_send_reply("403", "Call cannot be served here"); > > break; > > } > > > > # > > # validate if Incoming call > > # > > if (src_ip==192.168.20.105) { # address of VoIP Gateway > > # > > # INCOMING call from VoIP Gateway > > # > > log(1, "---- Incoming calls from VoIP > Gateway\n"); > > if (lookup("location")) { > > # > > # dest user registered ! > > # > > log(1, "---- dest user registered - > relay the message\n"); > > route(1); > > break; > > } else { > > # > > # dest user not registered (not part of > location table) > > # > > log(1, "---- dest user NOT > registered > \n"); > > sl_send_reply("403", "User not Found"); > > break; > > }; > > }; > > > > # > > # Calls received from ATA > > # > > log(1, "---- call received from ATA\n"); > > > > # > > # validate the authentication of dest user > > # > > log(1, "---- proxy_authorize()\n"); > > if (!proxy_authorize("", "subscriber")) { > > log(1, "---- src user NOT authenticated\n"); > > proxy_challenge("", "0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > log(1, "---- src user authenticated\n"); > > # > > # remove Authorization line if necessary > > # > > log(1, "---- remove proxy-authorization line\n"); > > consume_credentials(); > > > > > > # > > # is it neccessary ???? > > # > > lookup("aliases"); > > if (uri!=myself) { > > log(1, "---- Relay message because uri != > myself\n"); > > append_hf("P-hint: outbound alias - > uri!=myself\r\n"); > > route(1); > > break; > > }; > > > > # > > # Validate destination calls > > # > > if (uri=~"sip:31[0-9]*@.*") { # first digits = "31" following > with 0-9 > > # > > # Local IP calls > > # > > log(1, "---- local IP calls\n"); > > if (is_user_in("From", "local")) { # source user > part of local group? > > # > > # src part of local group, relay the > message > > # > > log(1, "---- src part of local > group\n"); > > if (lookup("location")) { # dest user > registered ? > > # > > # User registered in > location table > > # > > log(1, "---- dest user > registered\n"); > > log(1, "---- dest = > local > ATA\n"); > > route(1); /* relay the > message */ > > break; > > } else { > > log(1, "---- dest user > NOT registered \n"); > > sl_send_reply("403", "User > not Found"); > > break; > > }; > > } else { > > log(1, "---- src NOT part of local > group\n"); > > sl_send_reply("403", "No Permission for > local calls"); > > break; > > }; > > }; > > > > # > > # Local PBX calls > > # > > if (uri=~"sip:3[0-9]*@.*") { # first digit = "3" following with > 0-9 > > # > > # Local PBX calls > > # > > log(1, "---- local PBX calls\n"); > > if (is_user_in("From", "local")) { # source user > part of local group? > > # > > # src part of local group, relay the > message > > # > > log(1, "---- src part of local > group\n"); > > log(1, "---- forward message to VoIP > Gateway\n"); > > rewritehostport("192.168.20.105:5060"); > > forward(192.168.20.105, 5060); > > break; > > } else { > > log(1, "---- src NOT part of local > group\n"); > > sl_send_reply("403", "No Permission for > local calls"); > > break; > > }; > > }; > > > > if (uri=~"sip:9[2-9][0-9]*@.*") { /* first digit = "9" > following > with 2-9 */ > > # > > # Free PSTN calls > > # > > log(1, "---- free PSTN calls\n"); > > if (is_user_in("From", "free-pstn")) { # source > user > part of free_pstn group? > > # > > # src part of free-pstn group, forward > the message > > # > > log(1, "---- forward message to VoIP > Gateway\n"); > > rewritehostport("192.168.20.105:5060"); > > forward(192.168.20.105, 5060); > > break; > > } else { > > log(1, "---- src NOT part of > free_pstn group\n"); > > sl_send_reply("403", "No Permission for > free PSTN calls"); > > break; > > }; > > }; > > > > if (uri=~"sip:91[2-9][0-9]*@.*") { /* first digit = "91" > following with 2-9 */ > > # > > # Long distance PSTN calls > > # > > log(1, "---- long distance calls\n"); > > if (is_user_in("From", "ld")) { # source user part > of long_dist group? > > # > > # src part of long_dist group, forward > the message > > # > > log(1, "---- forward message to VoIP > Gateway\n"); > > rewritehostport("192.168.20.105:5060"); > > forward(192.168.20.105, 5060); > > break; > > } else { > > log(1, "---- src NOT part of long > ddistance (ld) group\n"); > > sl_send_reply("403", "No Permission for > long distance calls"); > > break; > > }; > > }; > > > > if (uri=~"sip:9011[0-9]*@.*") { /* first digit = "9011" > following with 2-9 */ > > # > > # International calls > > # > > log(1, "---- international calls\n"); > > if (is_user_in("From", "int")) { # source user part > of int group? > > # > > # src part of international group, > forward the message > > # > > log(1, "---- forward message to VoIP > Gateway\n"); > > rewritehostport("192.168.20.105:5060"); > > forward(192.168.20.105, 5060); > > break; > > } else { > > log(1, "---- src NOT part of > international (int) group\n"); > > sl_send_reply("403", "No Permission for > international calls"); > > break; > > }; > > }; > > > > # > > # Invalid calls > > # > > log(1, "---- call NOT authorized\n"); > > sl_send_reply("403", "Call not Authorized"); > > > > } > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] TLS
by Helge Waastad 29 Sep '05

29 Sep '05

Hi, sometimes when I retstart OpenSER eith tls support, I get: BUG: qm_free: freeing already freed pointer, first free: tls/tls_init.c: ser_free(207) - aborting br hw -- mvh/best regards Helge Waastad System Engineer Smartnet (+47)67830017

2 1

[Users] TLS testing
by Bogdan-Andrei Iancu 29 Sep '05

29 Sep '05

Hi everybody, To make more easier the testing of the new TLS code, I would like to know if there are people interested to get involved. My idea is to set a OpenSER proxy only with TLS as a central node for interdomain/interservers TLS connectivity. Everybody will be free to connect to the server and to route through calls to the other participants. I will take care about setting the proxy, providing TLS certificates and info about how to connect. By doing this, we will be able to test the TLS as conformity and robustness (number of calls, and number or connections). Again, if there are people interested in this, please let me know in order to start the setup. regards, Bogdan

2 3

[Serusers] SER and SN2300 inalp
by GT 29 Sep '05

29 Sep '05

Hello everybody, Anyone using SER with Inalp gateways SN2300 ? Any experience ? Thanks, gtel

4 5

[Serusers] Forwarding to a CISCO 1760 with FXO
by Nhadie 29 Sep '05

29 Sep '05

Hi All, I have this on my SIP server; # Local PSTN Destination forward to CISCO 1760 if ((uri=~"^(sip:)?02[0-9]*@................etc..etc.. if (is_user_in("From", "local-pstn")) { rewritehostport("12.34.56.78:5060"); consume_credentials(); t_relay_to_udp("12.34.56.78", "5060"); # World PSTN Destination forward to gateway provider } else if ((uri=~"^(sip:)?00[0-9]*@......etc.etc.. { if (is_user_in("From", "world-pstn")) { strip(2); prefix("00000#"); rewritehostport("87.65.43.21:5060"); consume_credentials(); t_relay_to_udp("87.65.43.21", "5060"); The forwarding to the gateway provider works no problem, but the one going to local pstn, rings on the pstn line but when I answer it session will be cut, busy signal. Is this a problem on the CISCO 1760 or in my configuration? TIA Regards, Nhadie

1 0

[Serusers] multi-domain support with one SER-2
by Lambert, Micheline (Satnet) 29 Sep '05

29 Sep '05

Hi, I'm new with SER since August 2005. I'm using ser v0.9.3 and mysql v4.1.13. I use SER with two domains: 192.168.20.55 and 192.168.20.155. Some users are configured in subscribe table of SER database with domain 192.168.20.55 and other users are configured in the same subscribe table of SER database with domain 192.168.20.155. I understand that user from domain 192.168.20.55 can not talk to users in domain 192.168.20.155. Is it right? I have the following problem: Users from domain 192.168.20.55 can make calls to PBX/PSTN but users from domain 192.168.20.155 can not, There is the message 403 Forbidden sent to the users. Here is the sequence: <- INVITE -> 407 Proxy Authentication Required <-ACK <-INVITE ->100 Trying ->403 Forbidden <-ACK ->403 Forbidden <-ACK ->403 Forbidden <-ACK I would like to know if there is a need to have two databases for the multi-domain support with one SER? Micheline Lambert I included a part of my ser.cfg file # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # ... listen=192.168.20.55 listen=192.168.20.155 # support multi-domains # ------------------ 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/textops.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/uri_db.so" # load the group module to use : is_useer_in() function loadmodule "/usr/local/lib/ser/modules/group.so" # load the acc module ffor accounting loadmodule "/usr/local/lib/ser/modules/acc.so" #new module for multi-domain support loadmodule "/usr/local/lib/ser/modules/domain.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc|registrar", "use_domain", 1) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", 1) # # 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_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of fflag which will be used fffor accounting; if a message # is labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("acc", "report_cancels", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # ------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # ------------------------------------------------------------------ # labeled all transactions for accounting # ------------------------------------------------------------------ log(1, "++++ labeled all transactions for accounting\n"); setflag(1); # ------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------ # 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 if (!method=="REGISTER") record_route(); # ------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------ # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; # ------------------------------------------------------------------ # Call Type Processing # ------------------------------------------------------------------ if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound uri!=myself\r\n"); route(1); 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=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!uri==myself) { /* myself = my IP address */ append_hf("P-hint: outbound alias - uri!=myself\r\n"); route(1); /* relay the message */ break; }; log(1, "++++ route other SIP messages???????????????\n"); route(1); }; } route[1] { if (!t_relay()) { sl_reply_error(); }; } route[2] { sl_send_reply("100", "Trying"); log(1, "==== www_authorize\n"); # validate the authentication of the user if (!www_authorize("", "subscriber")) { log(1, "==== send 401 Unauthorized\n"); www_challenge("", "0"); # send back 401 Unauthorized message break; # ask for another registration with auth. }; if (!check_to()) { sl_send_reply("401", "Unauthorized 2"); break; }; log(1, "==== remove Authorization line\n"); consume_credentials(); # remove Authorization digest info from message # save the user in the location table log(1, "==== save location and send Ok\n"); if (!save("location")) { # save the user in the location table, send Ok message sl_reply_error(); }; } route[3] { if (!uri=~"sip:\+?[0-9]+@.*") { log(1, "---- call cannot be serverd here - not numeric uri\n"); sl_send_reply("403", "Call cannot be served here"); break; } # # validate if Incoming call # if (src_ip==192.168.20.105) { # address of VoIP Gateway # # INCOMING call from VoIP Gateway # log(1, "---- Incoming calls from VoIP Gateway\n"); if (lookup("location")) { # # dest user registered ! # log(1, "---- dest user registered - relay the message\n"); route(1); break; } else { # # dest user not registered (not part of location table) # log(1, "---- dest user NOT registered \n"); sl_send_reply("403", "User not Found"); break; }; }; # # Calls received from ATA # log(1, "---- call received from ATA\n"); # # validate the authentication of dest user # log(1, "---- proxy_authorize()\n"); if (!proxy_authorize("", "subscriber")) { log(1, "---- src user NOT authenticated\n"); proxy_challenge("", "0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; log(1, "---- src user authenticated\n"); # # remove Authorization line if necessary # log(1, "---- remove proxy-authorization line\n"); consume_credentials(); # # is it neccessary ???? # lookup("aliases"); if (uri!=myself) { log(1, "---- Relay message because uri != myself\n"); append_hf("P-hint: outbound alias - uri!=myself\r\n"); route(1); break; }; # # Validate destination calls # if (uri=~"sip:31[0-9]*@.*") { # first digits = "31" following with 0-9 # # Local IP calls # log(1, "---- local IP calls\n"); if (is_user_in("From", "local")) { # source user part of local group? # # src part of local group, relay the message # log(1, "---- src part of local group\n"); if (lookup("location")) { # dest user registered ? # # User registered in location table # log(1, "---- dest user registered\n"); log(1, "---- dest = local ATA\n"); route(1); /* relay the message */ break; } else { log(1, "---- dest user NOT registered \n"); sl_send_reply("403", "User not Found"); break; }; } else { log(1, "---- src NOT part of local group\n"); sl_send_reply("403", "No Permission for local calls"); break; }; }; # # Local PBX calls # if (uri=~"sip:3[0-9]*@.*") { # first digit = "3" following with 0-9 # # Local PBX calls # log(1, "---- local PBX calls\n"); if (is_user_in("From", "local")) { # source user part of local group? # # src part of local group, relay the message # log(1, "---- src part of local group\n"); log(1, "---- forward message to VoIP Gateway\n"); rewritehostport("192.168.20.105:5060"); forward(192.168.20.105, 5060); break; } else { log(1, "---- src NOT part of local group\n"); sl_send_reply("403", "No Permission for local calls"); break; }; }; if (uri=~"sip:9[2-9][0-9]*@.*") { /* first digit = "9" following with 2-9 */ # # Free PSTN calls # log(1, "---- free PSTN calls\n"); if (is_user_in("From", "free-pstn")) { # source user part of free_pstn group? # # src part of free-pstn group, forward the message # log(1, "---- forward message to VoIP Gateway\n"); rewritehostport("192.168.20.105:5060"); forward(192.168.20.105, 5060); break; } else { log(1, "---- src NOT part of free_pstn group\n"); sl_send_reply("403", "No Permission for free PSTN calls"); break; }; }; if (uri=~"sip:91[2-9][0-9]*@.*") { /* first digit = "91" following with 2-9 */ # # Long distance PSTN calls # log(1, "---- long distance calls\n"); if (is_user_in("From", "ld")) { # source user part of long_dist group? # # src part of long_dist group, forward the message # log(1, "---- forward message to VoIP Gateway\n"); rewritehostport("192.168.20.105:5060"); forward(192.168.20.105, 5060); break; } else { log(1, "---- src NOT part of long ddistance (ld) group\n"); sl_send_reply("403", "No Permission for long distance calls"); break; }; }; if (uri=~"sip:9011[0-9]*@.*") { /* first digit = "9011" following with 2-9 */ # # International calls # log(1, "---- international calls\n"); if (is_user_in("From", "int")) { # source user part of int group? # # src part of international group, forward the message # log(1, "---- forward message to VoIP Gateway\n"); rewritehostport("192.168.20.105:5060"); forward(192.168.20.105, 5060); break; } else { log(1, "---- src NOT part of international (int) group\n"); sl_send_reply("403", "No Permission for international calls"); break; }; }; # # Invalid calls # log(1, "---- call NOT authorized\n"); sl_send_reply("403", "Call not Authorized"); }

2 1

[Serusers] create admin account for a second domain with one ser
by Lambert, Micheline (Satnet) 29 Sep '05

29 Sep '05

Hi, I'm new with SER since August 2005. I'm using ser v0.9.3 and mysql v4.1.13. With the ser_mysql.sh create command, we can create the ser database with an admin user with a domain let's say: 192.168.20.55. How can I create a second admin user, let's say admin2 for the second domain 192.168.20.155? Do I have to execute the same command ser_mysql.sh create with the second domain 192.168.20.155? I don't think so. Is there a command I can use, or I have to create the admin2 user directly in the database (via the mysql command)? Thanks. Micheline Lambert

1 0

[Serusers] installation & first run problem (I am new, ...)
by Ronald Wiplinger 29 Sep '05

29 Sep '05

I am new to SER, forgive me if I use "strange terms" or ask stupid or obviously things ;-) I downloaded SER and compiled from source. I tried to use the ser_mysql.sh script to create the database and the permittion. For me unknown reasons, the permissions for all ser(ro) are set to 'N' in the table mysql.user I tried to change it to 'Y' with GRANT ALL PRIVILEGES ON ser.* to 'ser'@'localhost' IDENTIFIED BY 'password' but that does not change anything in the mysql.user Strange is also that the database ser does not show up at all in phpMyAdmin and also the 4 records in mysql.user. Before I changed /usr/local/etc/ser/ser.cfg to reflect the mysql I tried to start ser with /usr/local/sbin/serctl start and I had many ser process in my ps list, but now it says: :/usr/local # /usr/local/sbin/serctl start Starting SER : PID file exists! (/var/run/ser.pid) already running? What have I done wrong? bye Ronald Wiplinger

2 2

[Users] asterisk
by Urban 29 Sep '05

29 Sep '05

Hi, I'm new to the list and didn't find this information in the archives so bare with me if this has already been discussed in the list. I'm trying to setup openser toghether with asterisk on the same machine. I don't get it to work using the basic information I found on the web with rewritehostport("localhost:5090") and t_relay(). openser is running on port 5060 and * on 5090. When using tcpdump on the machine openser is not even contacting asterisk on port 5090 instead it returns "100 trying..." to the phone. openser will just work as a proxy to * and add some extra security but most of the pbx routing logic and voicemail will be handled by *. Anyone having this up and running and willing to share the required configuration? thanks urban

2 1

[Users] avp error
by jayesh nambiar 29 Sep '05

29 Sep '05

hi bogdan, Thanks for the help. atleast I got rid of that error but got a new error. I am trying to implement the ACL Bitmap example provided in the avp_docs. The idea is as follows: # 1 - free destination # 2 - local call # 3 - long distance call # 4 - international call # if(uri=~"sip:0900[0-9]+") /* free destinations */ { if (!avp_check("$acl", "and/i:0x01") { sl_send_reply("403","Forbidden - Free Destinations Not Allowed"); return; }; } else if(uri=~"sip:0[1-9][0-9]+") { if (!avp_check("$acl", "and/i:0x04") { sl_send_reply("403","Forbidden - Long Distance Calls Not Allowed"); return; }; } else if(uri=~"sip:00[1-9][0-9]+") { if (!avp_check("$acl", "and/i:0x08") { sl_send_reply("403","Forbidden - International Calls Not Allowed"); return; }; } else { if (!avp_check("$acl", "and/i:0x02") { sl_send_reply("403","Forbidden - Local Calls Not Allowed"); return; }; }; # authorized # rewritehostport("gateway_ip:gateway_port"); rewritehostport("10.10.10.10:5060"); But now I get the following error: ERROR:avpops:parse_check_value: parse error in <and/i:0x01> pos 3 Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: ERROR:avpops:fixup_check_avp: failed to parse checked value Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: ERROR: fix_expr : fix_actions error Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: Binding '2204','sip:2204@192.168.2.20:8484' has expired Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: use_table(): Invalid parameter value Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: db_del_ucontact: Error in use_table Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: wb_timer(): Can't delete contact from the database Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: Binding '2207','sip:2207@192.168.2.18:8467' has expired Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: use_table(): Invalid parameter value Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: db_del_ucontact: Error in use_table Sep 29 15:33:31 localhost /usr/local/sbin/openser[15481]: wb_timer(): Can't delete contact from the database Is there anything wrong with the expression syntax or there could be problem with the entry in the table. Can you pls help me figure it out. thanx in advance. jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

2 3

[Serusers] Mediaproxy works only for the caller
by Felipe Louback 29 Sep '05

29 Sep '05

Hey all, I am using ser-0.9.4 and mediaproxy-1.4.2 but it seems that mediaproxy only works for the caller. During a call, if I use sessions.py, i see: Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------------------------- 150.164.x.x:8000 - 127.0.0.1:35000 - ?.?.?.?:? inactive 0'26" Unknown Audio 0/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 Proxy version: 1.4.2 The status is inactive and after 60 seconds it ends. The called part receives audio but the caller receives nothing. It seems that the called part is not passing through media proxy. Here it is my ser.cfg: route{ if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too many hops"); }; break; }; if (msg:len >= max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; record_route(); if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; t_relay(); break; }; if (method=="REGISTER") { # Mark as NAT'ed if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; if (!www_authorize("mydomain", "subscriber")) { www_challenge("mydomain", "0"); break; } if (!save("location")) { sl_reply_error(); }; break; }; if (method == "BYE" || method == "CANCEL") { end_media_session(); }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } How do I do for the called part to pass through mediaproxy? I've searched about this, but I found nothing. Initially I was using ser0.8.14 and I thought it was a problem with it and I am now using 0.9.0. It seems that the problem wasnt with ser version 0.8.14. Could anyone give me some advice or tell me where I can find more information about it? Thanx, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "For God so loved the world that he gave his one and only Son, that whoever believes in him shall not perish but have eternal life." John 3:16

1 1

[Serusers] Re: RTPproxy and SER
by Alexandre Aractingi 29 Sep '05

29 Sep '05

Hi again, I made the same test with mediaproxy instead of rtpproxy, the problem is exactly the same. In my config there is no NAT at all, private IP addresses are routed via an IP tunnel to my SER proxy, and I'd like to use the media proxy to communicate with an external gateway. The SDP within the 200OK is left intact, so a private IP address is given and the call fails... Any ideas? Thanks! > Le mardi 27 septembre 2005 à 18:30 +0200, Alexandre Aractingi a écrit : > > Hi all, I'm trying to use RTPproxy (CVS) and SER to enable > > locally-routed endpoints to talk to an external public gateway. When a > > call comes from the external gateway, the SDP in the INVITE gets > > rewritten properly with SER's IP address, but then when my phone hangs > > off, the SDP in the 200 OK doesn't get rewritten, so an RFC1918 IP > > address is passed to the external gateway (so I get no audio). > > > > I'm using nathelper with: > > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") > > > > and for my tests I put "force_rtp_proxy()" at the very beginning of the > > routing logic in order to make sure it gets invoked. > > > > But the SDP in the 200 OK is not rewritten no matter what. Is this a > > know issue? Is there a known workaround? > > > > Thanks in advance for any help, > > > Alexandre Aractingi > Net-tone / Active Telecom > Direct IP : +33 1 72 74 70 02 > Standard : +33 1 49 23 76 59 -- Alexandre Aractingi <aaractingi(a)libertysurf.fr>

4 4

[Serusers] timestamp field in ACC table
by Ryan Pagquil 29 Sep '05

29 Sep '05

Hi guys, I've just found out that the timestamp field in ACC table doesn't contain the real timestamp relative to 1970. It is just like the time field in ACC table but without the ":" and the "-" delimiters. So when I computed for the duration it doesn't give me the correct duration time. Here is the query I used to extract for duration: SELECT t1.from_uri as CLR, t1.to_uri as CLD, (t2.time - t1.time) as Duration, t1.time as Setup_Time, t2.time as Release_Time FROM acc t1, acc t2 WHERE t1.sip_callid = t2.sip_callid AND ((t1.fromtag = t2.fromtag and t1.totag = t2.totag) OR (t1.fromtag = t2.totag and t1.totag = t2.fromtag)) AND t1.sip_method='ACK' AND t2.sip_method='BYE'; attached is a text file that contains one of the record in acc table. Thanks, Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com +-------------------------------------------------------------------------------------+---------------------------------------------------------+------------+------------+-----------------------------------------------------------------------------+------------------------------+----------------------------------+-------------------------------+--------------------------------------------------------+-----------+---------------+----------------------------------+------------+---------------------+----------------+----------------+----------------+ | sip_from | sip_to | sip_status | sip_method | i_uri | o_uri | from_uri | to_uri | sip_callid | username | domain | fromtag | totag | time | timestamp | caller_deleted | callee_deleted | +-------------------------------------------------------------------------------------+---------------------------------------------------------+------------+------------+-----------------------------------------------------------------------------+------------------------------+----------------------------------+-------------------------------+--------------------------------------------------------+-----------+---------------+----------------------------------+------------+---------------------+----------------+----------------+----------------+ | "5997" <sip:5997@202.147.52.22:5060>;tag=5deb812f096b2f10914b000000000000 | "632108" <sip:632108@202.84.24.107:5060>;tag=3332695765 | 200 | INVITE | sip:632108@202.84.24.107:5060;transport=UDP | sip:user1@10.0.1.144:5060 | sip:5997@202.147.52.22:5060 | sip:632108@202.84.24.107:5060 | 000d2858-7ff00284-462b78c8-5c0771e7(a)202.147.52.22 | 5997 | 202.147.52.22 | 5deb812f096b2f10914b000000000000 | 3332695765 | 2005-09-28 06:51:11 | 20050928145111 | 0 | 0 | | "5997" <sip:5997@202.147.52.22:5060>;tag=5deb812f096b2f10914b000000000000 | "632108" <sip:632108@202.84.24.107:5060>;tag=3332695765 | 200 | ACK | sip:202.84.24.107;ftag=5deb812f096b2f10914b000000000000;lr=on | sip:user1@202.84.24.126:1470 | sip:5997@202.147.52.22:5060 | sip:632108@202.84.24.107:5060 | 000d2858-7ff00284-462b78c8-5c0771e7(a)202.147.52.22 | 5997 | 202.147.52.22 | 5deb812f096b2f10914b000000000000 | 3332695765 | 2005-09-28 06:51:11 | 20050928145111 | 0 | 0 | | "5997" <sip:5997@202.147.52.22:5060>;tag=5deb812f096b2f10914b000000000000 | "632108" <sip:632108@202.84.24.107:5060>;tag=3332695765 | 200 | BYE | sip:202.84.24.107;transport=UDP;ftag=5deb812f096b2f10914b000000000000;lr=on | sip:user1@202.84.24.126:1470 | sip:5997@202.147.52.22:5060 | sip:632108@202.84.24.107:5060 | 000d2858-7ff00284-462b78c8-5c0771e7(a)202.147.52.22 | 5997 | 202.147.52.22 | 5deb812f096b2f10914b000000000000 | 3332695765 | 2005-09-28 06:52:13 | 20050928145213 | 0 | 0 |

1 0

[Users] B2bua +Ser +Asterisk
by harry gaillac 29 Sep '05

29 Sep '05

Hello, I wish to set b2bua-1.5.0 (Vovida) between Ser and Asterisk. Ser-------B2bua-------Asterisk---Pstn | Billing How may I set b2bConfig.xml in order to send calls to pstn via b2bua (billing) and asterisk ? Can I receive calls from pstn to ser via b2bua if asterisk is able to send call to b2bua ? Can we use b2bua for both incoming/outgoing calls ? Regards Harry .......... <Proxy_Server> <Address>xxx.xxx.xxx.xxx</Address> <Port>xxxx</Port> </Proxy_Server> .......... ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] voicemail howto
by Taymour A. El Erian 29 Sep '05

29 Sep '05

Hi. Is there any how to on how to configure OpenSER, SEMS for voicemail ?, the one with SEMS does not work -- Taymour A El Erian System Division Manager RHCE, LPIC, CCNA, MCSE, CNA TE Data E-mail: taymour.elerian(a)tedata.net Web: www.tedata.net Tel: +(202)-4166600 Fax: +(202)-4166700 Ext: 1101

3 15

[Users] access control list using avps
by jayesh nambiar 29 Sep '05

29 Sep '05

hi daniel, the $avp is an alias to i:800. My main doubt is where in the table should i mention the values 1,2,3 and 4 for the users who got the appropriate privileges. I am sorry, for i get confused manipulating the values of avp. pls help in this. thanks a lot jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

4 3

[Serusers] about my previous question [please help me about how to config radiusclient!]
by zhangshuai 29 Sep '05

29 Sep '05

Dear all, Sorry to bother you again. After googled the error messages: Sep 29 17:11:06 localhost ser[4408]: rc_ip_hostname: couldn't look up host by addr: DA61FC29 Sep 29 17:11:06 localhost ser[4408]: rc_send_server: no reply from RADIUS server unknown:1812 I found this: http://forums.staros.com/showthread.php?p=21895 "The key thing most people miss is that the IP you enter in the radius server for the remote that will talk to it has to be the IP of the device that the server is listening on. Most people simply use the IP of the gateway device and that is not correct unless that device also has Hotspot or PPPoE or some other service that talks to radius." I still don't know how to check his words are right. What do you think? Shuai http://www.goldentek.biz

1 0

[Serusers] please help me about how to config radiusclient!
by zhangshuai 29 Sep '05

29 Sep '05

Hi all, I wish to use softphone X-lite on SER and radius in db_mode. When I installed both freeradius 1.0.4 and radiusclient 0.4.3 on the same server which ser 0.8.14 has installed on, I could log in X-lite through radius authorization. And now I want radiusclient (on server A) to send the request to a remote radius server (on server B) to realize authorization. I have changed those config files: /usr/local/etc/radiusclient/servers, /usr/local/etc/radiusclient/radiusclient.conf and /usr/local/etc/raddb/clients.conf as the document (http://www.iptel.org/ser/doc/ser_radius/ser_radius.html#AEN193) did. Then I test the radius server as the above document said on server A: radclient -f digest server B auth testing123 and I received expected replied message from server B. But when I log in x-lite, there are error messages: Sep 29 17:11:06 localhost ser[4408]: rc_ip_hostname: couldn't look up host by addr: DA61FC29 Sep 29 17:11:06 localhost ser[4408]: rc_send_server: no reply from RADIUS server unknown:1812 How come? Did I mis-config anything? Many thanks for your reply! Shuai http://www.goldentek.biz

1 0

RV: [Users] RV: Nat Help
by Fernando Rodriguez 29 Sep '05

29 Sep '05

Does any one has any idea or can point me to a place to read about this .. Thanks .. _____ De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre de Fernando Rodriguez Enviado el: Miércoles, 28 de Septiembre de 2005 11:18 a.m. Para: users(a)openser.org Asunto: [Users] RV: Nat Help Hello y have 3 interfaces on my box 1 Lan to Internet 2 Lan to 192.168.1.x 3 Lan to 192.168.108.x I only have one domain in my openser I want to know a method to use the mediaproxy only when sending packet tru interfaces for example When making a call from lan 108 to internet or 192.168.1.x Im putting this kind of rules I set different number for number for each lan 1 for eth0 2 for eth1 3 for eth2 so each client has different number regarding the interface he sees .. Is this valid ?? if (from_uri=~"sip:|1[1]|2[0-9]|3[0-9][:@]" && to_uri=~"sip:|1[2]|2[0-9]|3[0-9]|4[0-9][:@]") { log("de 1 a 2\n"); use_media_proxy(); }; if (from_uri=~"sip:|1[2]|2[0-9]|3[0-9]|[0-9][:@]" && to_uri=~"sip:|1[1]|2[0-9]|3[0-9][:@]") { log("de 2 a 1\n"); use_media_proxy(); }; And not to use mediaproxy when calling within the same lan. Thanks for your help Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

2 1

[Users] load_credentials
by Richard Z 29 Sep '05

29 Sep '05

Hi, I am trying to use load_credentials modparam in auth_db to load some colums in the subscriber table. In mysql db table, what type should I use the columns? For example, modparam("auth_db", "load_credentials", "i:15=timeout") If I define timeout as varchar, it gets the value. If as int, it doesn't generate the avp. Is this supposed to be this way? Do I miss anything? Thanks, Richard

3 4

[Users] avp error
by jayesh nambiar 29 Sep '05

29 Sep '05

hi all, can somebody pls tell me what this error is!!! ERROR:avpops_init: "AVP_DB" present but "AVP_TABLE" found empty my db scheme is as follows: modparam("avpops","db_scheme","scheme0:username_col=username;domain_col=domain;value_col=acl;value_type=integer;table=subscriber") I have added a column called acl in the subscriber table and added integer value in it. for eg: for user 201 the acl column value is 4 so that he is privileged to make international calls. Is this correct or some mistake in my understanding. pls let me know.. thanks a lot. jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

2 1

[Serusers] Re: Sems IVR Module
by Nicky 29 Sep '05

29 Sep '05

Please help. ----- Original Message ----- From: "Nicky" <nicky(a)caliber.com.sg> To: <serusers(a)lists.iptel.org>; <sems(a)lists.iptel.org> Sent: Wednesday, September 28, 2005 10:38 PM Subject: Sems IVR Module > Hi all, > > I am testing the sems version from the Head and when i uses a SIP Phone to > call into > the IVR module for recording when I am in the middle of recording, the > program detected DTMF > signal and jump out of the script. Why is it so? as I never pressed on any > key? How can I resolve this > problem? Why is the voice signal become the DTMF signal? Is there any > adjustment that > i can do to solve this problem , as my voice goes louder the more DTMF > signal is being detected. > > Please help. > > regards, > nicky >

1 0

[Serusers] can not open dictionary from freeradiusd!
by Manoranjan Pandey 29 Sep '05

29 Sep '05

hello, guys: I installed freeradiusd now, i want to connect the two together. the problem is that when i use the radiusd-xyz raddb/dictionary, and run radiusd -X, then the messages came out as follow: ------------------------------------- [root@localhost ~]# radiusd -xyz Starting - reading configuration files ... Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary[14]: Couldn't open dictionary " /usr/local/share/freeradius/dictionary": No such file or directory Errors reading radiusd.conf again i am using this command--- [root@localhost ~]# radiusd -x Starting - reading configuration files ... Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary[14]: Could n't open dictionary " /usr/local/share/freeradius/dictionary": No such file o r directory Errors reading radiusd.conf again i am using .. [root@localhost ~]# radiusd -y Thu Sep 29 11:03:19 2005 : Info: Starting - reading configuration files ... again i am using .. [root@localhost ~]# radiusd Thu Sep 29 10:54:56 2005 : Info: Starting - reading configuration files ... WHEN I AM TRYING WITH RADPWTST ----------------------- [root@localhost ~]# radpwtst Bad format in dictionary '/usr/local/etc/raddb/dictionary' at line 14: $INCLUDE /usr/local/share/freeradius/dictionary Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 2 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Access-Request... No reply Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 44 is not defined in your dictionary Attribute number 40 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 41 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Accounting-Request Start... No reply Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 44 is not defined in your dictionary Attribute number 40 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 41 is not defined in your dictionary Attribute number 46 is not defined in your dictionary Attribute number 42 is not defined in your dictionary Attribute number 43 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Accounting-Request Stop... No reply [root@localhost ~]# radpwtst Bad format in dictionary '/usr/local/etc/raddb/dictionary' at line 14: $INCLUDE /usr/local/share/freeradius/dictionary Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 2 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Access-Request... No reply Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 44 is not defined in your dictionary Attribute number 40 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 41 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Accounting-Request Start... No reply Attribute number 1 is not defined in your dictionary Attribute number 6 is not defined in your dictionary Attribute number 4 is not defined in your dictionary Attribute number 5 is not defined in your dictionary Attribute number 61 is not defined in your dictionary Attribute number 44 is not defined in your dictionary Attribute number 40 is not defined in your dictionary Attribute number 30 is not defined in your dictionary Attribute number 31 is not defined in your dictionary Attribute number 41 is not defined in your dictionary Attribute number 46 is not defined in your dictionary Attribute number 42 is not defined in your dictionary Attribute number 43 is not defined in your dictionary No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown No such attribute Unknown sending Accounting-Request Stop... No reply please try to mail me olution of this problem. thanking you and waiting for ur reply. bye.

1 0

[Serusers] Changing contents of TO address
by Russell Cozens 29 Sep '05

29 Sep '05

Hi, I am trying to setup a PSTN gateway, the documentation is fine for this and I have it working successfully with a gateway. However, the gateway I would like to use is a bit fussy and seems to be looking at the SIP TO address as well as the INVITE line. I am using rewritehost to change the domain in the uri and this changes the INVITE header but not the SIP TO address within the INVITE. Is it possible to change this ? I've even tried various replace commands from textops.so but the "to" field is never changed. Thanks Russell

5 4

[Serusers] Arbitrary queries
by sip 28 Sep '05

28 Sep '05

Is the only way to make arbitrary queries from the DB to use an external script? I'm having a devil of a time creating something like: SELECT from_uri FROM acc WHERE to_uri LIKE '%<username@domain>%' AND sip_method='INVITE' ORDER BY TIME DESC LIMIT 1 Any ideas? Or am I stuck with external scripts? N.

2 1

[Users] access control list using avps
by jayesh nambiar 28 Sep '05

28 Sep '05

hi daniel, thanx a lot for the help. I'll definitely try creating a new table. Yeah the example is exactly what i need to achieve at present. Based on that i am planning to enhance my features. But what would the column 'acl' contain, will it contain the integer value based on the user privileges. I did'nt try yet as I am unable to understand the bitwise operation. Sorry if it seems rather stupid. thanx again jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

1 0

[Serusers] Regular Expression Problems in this if statement?
by Sherwood McGowan 28 Sep '05

28 Sep '05

Hello all, very glad to have been a user on this list, though I have been lurking. I've checked the documentation and have had trouble finding out information on how to translate my standard regexes (perl) into SER's routing logic I'm trying to use the following logic: if (uri =~ "^sip:1[0-9]{10}\@[0-9]{3}\.[0-9]{3}\.1\.foo\.bar\.net") { #perform actions } Now, that SHOULD match the following scenario: uri has 1 + 10 digits + @ + 3digits + . + 3digits + . + 1 + .foor.bar.net Unfortunately, it's not working.Any help translating the regex, or even just a link to the syntax allowed, maybe it's not allowing {n} to match n times? Thanks, Sherwood McGowan

3 3

[Users] RV: Nat Help
by Fernando Rodriguez 28 Sep '05

28 Sep '05

Hello y have 3 interfaces on my box 1 Lan to Internet 2 Lan to 192.168.1.x 3 Lan to 192.168.108.x I only have one domain in my openser I want to know a method to use the mediaproxy only when sending packet tru interfaces for example When making a call from lan 108 to internet or 192.168.1.x Im putting this kind of rules I set different number for number for each lan 1 for eth0 2 for eth1 3 for eth2 so each client has different number regarding the interface he sees .. Is this valid ?? if (from_uri=~"sip:|1[1]|2[0-9]|3[0-9][:@]" && to_uri=~"sip:|1[2]|2[0-9]|3[0-9]|4[0-9][:@]") { log("de 1 a 2\n"); use_media_proxy(); }; if (from_uri=~"sip:|1[2]|2[0-9]|3[0-9]|[0-9][:@]" && to_uri=~"sip:|1[1]|2[0-9]|3[0-9][:@]") { log("de 2 a 1\n"); use_media_proxy(); }; And not to use mediaproxy when calling within the same lan. Thanks for your help Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

1 0

RE: [Serusers] Regular Expression Problems in this if statement?
by Sherwood McGowan 28 Sep '05

28 Sep '05

Thanks guys for your reponses _____ From: Ramin NIkaeen [mailto:Rnikaeen@goldline.net] Sent: Wednesday, September 28, 2005 12:29 PM To: Sherwood McGowan Subject: RE: [Serusers] Regular Expression Problems in this if statement? Try uri =~ "^sip:1[0-9]{10}@[0-9]{3}\.[0-9]{3}\.1\.foo\.bar\.net") ie. Remove "\" before the "@". Let me know if it works. ramin -----Original Message----- From: Sherwood McGowan [mailto:madprofzero@yahoo.com] Sent: Wednesday, September 28, 2005 12:22 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Regular Expression Problems in this if statement? Hello all, very glad to have been a user on this list, though I have been lurking. I've checked the documentation and have had trouble finding out information on how to translate my standard regexes (perl) into SER's routing logic I'm trying to use the following logic: if (uri =~ "^sip:1[0-9]{10}\@[0-9]{3}\.[0-9]{3}\.1\.foo\.bar\.net") { #perform actions } Now, that SHOULD match the following scenario: uri has 1 + 10 digits + @ + 3digits + . + 3digits + . + 1 + .foor.bar.net Unfortunately, it's not working.Any help translating the regex, or even just a link to the syntax allowed, maybe it's not allowing {n} to match n times? Thanks, Sherwood McGowan

1 0

[Users] access control list using avps
by jayesh nambiar 28 Sep '05

28 Sep '05

hi all, I tried going through the avp documents but I have a few doubts related to avp_db_load and avp_check functions. I wanted to implement the access control list. the access control values may be jus like: 1 => local calls 2 => long distance calls 3 => international calls. I am using the default table usr_preferences. when I do the avp_check as: if(!avp_check("$acl", "and/0x01")) I get an error that it cannot be parsed and fix the avp_check. And can someone pls explain how the bitwise check "and", "or" etc works.. thanks, jayesh --------------------------------- Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort.

2 1

[Users] business mailing list
by Daniel-Constantin Mierla 28 Sep '05

28 Sep '05

Hello everybody, a new mailing list was created to provide a forum for parties who wish to discuss commercial or business related activities, opportunities, etc. related to OpenSER, VoIP, SIP ... The email address of the mailing list is "business(a)openser.org". To subscribe/unsubscribe, follow the link: http://openser.org/cgi-bin/mailman/listinfo/business It is list where everybody can subscribe but no archiving is done. This should force anyone interested in some business to get up-to-date information, avoiding questions related to past mails found on web archive. Also, it should prevent some spaming from users trying to increase search engine's ranking via links and message posts. The usual mails sent to this list should be in the form of "I need _this_, who does it?" or "I do _this_, who needs it?", but should not be limited to such pattern. Basic rules for posting on the mailing list: - keep a fair conduit of messages, no bad remarks about others or counter-advertising - do not send images, flash or other graphics - post links instead - it is recommended to reply directly to the sender when it is a one-to-one business, the others may not be interested, otherwise they should post a message - if nobody answers, then nobody needs/provides what you offer/need, it is no reason to send the message hourly, wait for some days or weeks Hopefully, it will help everybody for a mutual benefit. Cheers, Daniel

1 0

[Serusers] How to terminate a session?
by Federico Giannici 28 Sep '05

28 Sep '05

Suppose that we know which calls are open, based on the accounting data, so we have all dialog parameters (Call-ID, From, To, C-Seqs, etc...). How can I terminate a session that is lasting too long? If I suppose that a session is actually closed (for example because one UA has crashed), how can I be sure that it is really closed (eventually actually closing it)? I'm thinking to send two BYE messages to both UAs (impersonating the other UA), so to make them close the session (if it is still open). Is this tecnically possible? Are there any contraindications? Is there a better way to obtain it? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

1 0

[Serusers] Aliases and Location Problem... please help
by Ryan Pagquil 28 Sep '05

28 Sep '05

Hi SER users, Why does my aliases and locations are not persistent? Considering that they appear in my database (aliases and location), when I restarted SER all aliases and locations are not being read by SER. What will I do is to restart my X-lite softphone for me to re-register. And for my aliases I will run again serctl alias add <alias> sip:<uri> for the aliases to work. It seems like SER is not reading my database and all contact informations are only stored on its memory. But all these contacts appears in my location and aliases tables of the ser database. Is there any known bugs about this? Please help. I'm using SER-0.9.3 in SPARC64 Debian linux with mediaproxy as nat traversal solution. Any help is greately appreciated... Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

1 0

[Serusers] 483 Too Many Hops problem ;;;;
by 송 미진 27 Sep '05

27 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 dear, I'm working on a project and i'm using ser as a sip server. actually i have this problem: every time i send an instant message to a subscriber i get this error message: 483 too many hops i tried to fix it but i couldn't and i don't know what to do ! (My computer connect many computers. Is it reason?!) if any one can help me i'll be very grateful i'm using ser 0.9.3 When I use sipsak, I received following messages. root]# sipsak -T -s sip:user@mydomain.com warning: IP extract from warning activated to be more informational 0: 192.168.178.194 (0.148 ms) SIP/2.0 483 Too Many Hops 1: 192.168.178.194 (1.915 ms) SIP/2.0 483 Too Many Hops 2: 192.168.178.194 (25.673 ms) SIP/2.0 483 Too Many Hops 3: 192.168.178.194 (4.855 ms) SIP/2.0 483 Too Many Hops 4: 192.168.178.194 (9.286 ms) SIP/2.0 483 Too Many Hops 5: 192.168.178.194 (20.285 ms) SIP/2.0 483 Too Many Hops 6: 192.168.178.194 (17.068 ms) SIP/2.0 483 Too Many Hops 7: 192.168.178.194 (15.038 ms) SIP/2.0 483 Too Many Hops 8: 192.168.178.194 (16.379 ms) SIP/2.0 483 Too Many Hops 9: timeout after 500 ms 9: timeout after 1000 ms 9: timeout after 2000 ms 9: 192.168.178.194 (1480.452 ms) SIP/2.0 483 Too Many Hops 10: 192.168.178.194 (20.068 ms) SIP/2.0 483 Too Many Hops 11: 192.168.178.194 (21.916 ms) SIP/2.0 483 Too Many Hops 12: 192.168.178.194 (22.478 ms) SIP/2.0 483 Too Many Hops 13: 192.168.178.194 (22.984 ms) SIP/2.0 513 Message too big without Contact header ====================================== here's my code: # ----------- 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 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://root:vlftmd@localhost/ser" alias=sam.cse # ------------------ 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/textops.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_url","mysql://root:vlftmd@localhost/ser") #modparam("usrloc", "db_mode", 0) modparam("registrar","use_domain",1) # 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","secret","123123123") #modparam("auth_db","db_url","mysql://root:vlftmd@localhost/ser") #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 >= 2048 ) { 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 if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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 if (!www_authorize("sam.cse", "subscriber")) { www_challenge("sam.cse", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } _________________________________________________________________ 싸게 싸게 MSN공동구매 http://www.waawaa.com/cobuy/cobuy_default.asp?siteid=10160

2 2

[Users] Nat Help
by Fernando Rodriguez 27 Sep '05

27 Sep '05

Hello y have 3 interfaces on my box 1 Lan to Internet 2 Lan to 192.168.1.x 3 Lan to 192.168.108.x I only have one domain in my openser I want to know a method to use the mediaproxy only when sending packet tru interfaces for example When making a call from lan 108 to internet o 192.168.1.x And not to use mediaproxy when calling within the same lan. Thanks for your help Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

2 1

[Serusers] problem detectec: Not found in usrloc
by Ángel Gómez 27 Sep '05

27 Sep '05

Hi, I have found the problem, but i don't know ho to solve it. To search the problem I started SER with a debug level 9, redirectin STDERROR to a file ( if some one want to do it is: /etc/init.d/ser start 2>>/ser.log1 ) and when I was reading the file I found it: 2(1926) lookup(): 'ninguna' Not found in usrloc This is what the server say when i send a MESSAGE to ninguna(a)10.22.1.249, but then I used: serctl ul show, and got this: ...Record(0x4045d1b8)... domain: 'location' aor : 'ninguna' ~~~Contact(0x4045d218)~~~ domain : 'location' aor : 'ninguna' Contact : 'sip:ninguna@10.22.1.249' Expires : 3583 q : Call-ID : '1549255136(a)10.22.1.5' CSeq : 1 User-Agent: 'sipsak 0.8.9_pre' received : '' State : CS_SYNC Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... can any one give me a solution? BEFORE of sending the message I send a REFISTER for the user ninguna.

2 1

[Serusers] Adding Aliases for inbound PSTN
by Ashutosh kumar 27 Sep '05

27 Sep '05

Hi, I want to configure my ser.cfg so as to bind a user john to a DID 1234 from my pstn switch. John is a user in my ser DB and i want to assign him the phone number 1234, so that whenever my pstn GW forwards this call for 1234 to SER, SER should forward this call to user john.Which command do i use ni this case. serctl alias add john 1234, OR serctl alias add 1234 john, beacuse in my case, none of these is working, although i can see the switch forwarding the call to SER. How can i use save("aliases") in this scenario? I have used lookup ("alaises") at appropriate places. Thanks Ashutosh

3 4

[Users] mail to sobomax
by harry gaillac 27 Sep '05

27 Sep '05

(open)ser users, Forget this post I just try to get a reply from Mr Maxim Sobolev. Dear Maxim, Please to reply to my mails I've been sending since several months. http://mail.iptel.org/pipermail/serusers/2005-August/022745.html Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] RTPproxy and SER
by Alexandre Aractingi 27 Sep '05

27 Sep '05

Hi all, I'm trying to use RTPproxy (CVS) and SER to enable locally-routed endpoints to talk to an external public gateway. When a call comes from the external gateway, the SDP in the INVITE gets rewritten properly with SER's IP address, but then when my phone hangs off, the SDP in the 200 OK doesn't get rewritten, so an RFC1918 IP address is passed to the external gateway (so I get no audio). I'm using nathelper with: modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") and for my tests I put "force_rtp_proxy()" at the very beginning of the routing logic in order to make sure it gets invoked. But the SDP in the 200 OK is not rewritten no matter what. Is this a know issue? Is there a known workaround? Thanks in advance for any help, -- Alexandre Aractingi <aaractingi(a)libertysurf.fr>

1 0

[Serusers] cisco 7905G
by Alberto 27 Sep '05

27 Sep '05

Hi, I have the next scheme: 7905G ------- NAT ------- Internet ------ SER+mediaproxy When the 7905G sends a INVITE the ser responds: Status: 407 Proxy Authentication Required. This is right, but the destination port this packet not is right because the ser sends the packet with destination port 5060 (private port) and not the (public port). If I open this port all it is right, I do not want to open no port. This single problem happens to me with 7905G (no linksys, no X-Ten, ...). I do not understand what it is happening. Thanks for your aid. -- Alberto

3 5

RE: [Serusers] Restriction on port number??
by pankaj.munjal＠wipro.com 27 Sep '05

27 Sep '05

Hi, Yes, I've set the SIP proxy address as <proxy-IP>:7065 - Pankaj. -----Original Message----- From: sip [mailto:sip@arcdiv.com] Sent: Tuesday, September 27, 2005 8:32 PM To: Pankaj Munjal (WT01 - Voice & Next Generation Networks); serusers(a)lists.iptel.org Subject: Re: [Serusers] Restriction on port number?? Is x-lite set to use your proxy:7065 ? There's no restriction as far as I know. N. On Tue, 27 Sep 2005 20:17:58 +0530, pankaj.munjal wrote > Hi everyone, > > I had a basic question about SER. > Is there a restriction that we have to bring up SER on port number 5060 only? > > When I tried changing the port number in the ser config file, ser comes up normally, but it doesn't answer the REGISTER message from the UA. > I am using XLite as a UA and bringing up the ser on port 7065. > > Any pointers as to what can be reason for this? > > Cheers, > > Pankaj Munjal > Automation - Nortel Networks > Wipro Technologies, Bangalore. > ESN : 6-877-5666 > PBX: 080- 28520408, Extn:- 83265 > > > Confidentiality Notice > > The information contained in this electronic message and any attachments to this message are intended > for the exclusive use of the addressee(s) and may contain confidential or privileged information. If > you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately > and destroy all copies of this message and any attachments. > Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

2 1

[Serusers] Restriction on port number??
by pankaj.munjal＠wipro.com 27 Sep '05

27 Sep '05

Hi everyone, I had a basic question about SER. Is there a restriction that we have to bring up SER on port number 5060 only? When I tried changing the port number in the ser config file, ser comes up normally, but it doesn't answer the REGISTER message from the UA. I am using XLite as a UA and bringing up the ser on port 7065. Any pointers as to what can be reason for this? Cheers, Pankaj Munjal Automation - Nortel Networks Wipro Technologies, Bangalore. ESN : 6-877-5666 PBX: 080- 28520408, Extn:- 83265 Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

2 1

[Serusers] Database queries?
by sip 27 Sep '05

27 Sep '05

I'm attempting to figure out how to use AVPops to grab data from the calls_forwarding database table so I can avoid having to write my own module to handle that sort of thing, and I can avoid doing what I'm CURRENTLY doing which is using a pretty hideous exec_msg SQL query hack. I've run into a snag that I can't figure a way around.... In the calls_forwarding table, I have these fields used in this fashion: username == username domain == domain uri_re == username@domain information (for use in a couple of ways) purpose == purpose for this entry (callblock, callfwd, screening, etc) action == action to take (for callblock, could be reply or relay, etc) param1 == option for the action above param2 == additional option So... if I have a call block entry, it would be like this: username = bob domain = mydomain.com uri_re = motherinlaw(a)another.domain.com purpose = callblock action = reply param1 = 486 param2 = "Busy" The logic would be along the lines of, when a call comes in for bob(a)mydomain.com, I check to see if he has any callblock entries. If so, I check those to see if any match the From user. If there's a match, take the action in action (in this case reply with 486 - Busy... could be relay to voicemail, etc). The problem I'm having is in the grabbing of all the data from the DB and comparing it to the right values. For instance, with this avpops query avp_db_load("$ruri/username", "$urire/$urire_scheme"); # Load the urire column for this and check to see if # it matches the $from address if(avp_check("$urire", "eq/$from/I")) { send reply... take action... etc. } I'm calling from the number motherinlaw(a)another.domain.com. If I use the above, the avp_check test fails. If I put if(avp_check("$urire", "eq/motherinlaw(a)another.domain.com/I")) { send reply... take action... etc. } Then the test succeeds and the appropriate action is taken. Any guesses as to what I'm missing here... why the test is failing when I use the $from variable for AVPops, but not when I hard-code the $from address into the test (which implies to me that it's grabbing the right data from the database, but there's an issue in either the $from variable or the comparison statement)? I'd love a little shove in the right direction. I thought about writing a module that returned what I need without the need for complex logical paths, but my C is pretty rusty, and I'm trying to save myself the hair-pulling. :) N.

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[Serusers] new voip service [ot] FYI
by Iqbal 27 Sep '05

27 Sep '05

I know I know not supposed to advertise, but this is not my company its dixons, a 1400 retail unit in the UK, will have vonage running for cover in the UK http://www.theregister.co.uk/2005/09/27/dixons_freetalk/ £79.99 includes hardware and unlimited calls to UK landline for a year..... plusnet £3.99 for 3500 mins per month (but need there bandwidth) and ipclouds.co.uk (£69.99) Iqbal

1 0

[Serusers] SEMS+IVR
by Abdul Lateef 27 Sep '05

27 Sep '05

Hi all, I wanted configure IVR with SEMS. I have a samall question. Is SEMS need SER, or SEMS can work with another SIP GK. i have already SIP GK and i wanted to use SEMS priticular for IVR. Please let me know if SEMS can work withou SER? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

[Serusers] Seperate SER and Oubound Proxy
by Kevin Brennan 27 Sep '05

27 Sep '05

We would like to use STUN as preferred NAT traversal method but for those situations where STUN does not work we would offer an outbound proxy (configured in user agent). We would like to keep the NAT traversal proxy completely separate from SER. I wonder has anybody implemented this ? how ? and any other suggestions/feedback. Br /Kevin Brennan.

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RE: [Serusers] is SER ( 0.9.4) 100% 3GPP SIP call flow compliant ?
by Ashish Gilotra 27 Sep '05

27 Sep '05

Hi Klaus Thanks for answer. What is missing in SER which makes it non-compliant to 3GPP IMS call flow ? Br Ashish -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Tuesday, September 27, 2005 2:51 PM To: ashish gilotra Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] is SER ( 0.9.4) 100% 3GPP SIP call flow compliant ? ashish gilotra wrote: > Hi > > We are evaluating SER for our purpose. > > We need to use 3GPP SIP call flows. > > Is SER compliant ? No. regards klaus _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Users] Problem with ACK's (branch=0)
by Rodrigo P. Telles 27 Sep '05

27 Sep '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've been experiencing some troubles with ACK's with branch=0. I found a thread about it but I didn't find a 'solution' folowing the thread. http://mail.iptel.org/pipermail/serdev/2005-April/004296.html Can some one point me to the correct answer for that question? Thanks in advance. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> TI Manager Devel-IT - http://www.devel.it ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDJbvNiLK8unYgEMQRAoATAJwJSLGiN1/XuAUk2aVm4rm5oGD00ACfYJf0 XbL8Vv4unK6U3j974UOtYU0= =gZzI -----END PGP SIGNATURE-----

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