- sr-users - kamailio.org

[Users] VoN Berlin & OpenSER Summit
by Daniel-Constantin Mierla 05 Nov '06

05 Nov '06

Hello everybody, I will see many of you at Berlin, next week. The summit has a BoF session on Tuesday, 15:45-17:00 and continues next days with morning and afternoon sessions: see http://www.openser.org for more. Anyone who wants to get in contact before or to have a private meeting, the can find us at booth 321. All participants registered before November 1 got full VON conference pass (valid for the three days, November 6-8, for exhibition and conference, including special events, like party, etc..). Cheers, Daniel

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[Serusers] LCS+SER+ASTERISK
by Felipe Louback 04 Nov '06

04 Nov '06

Howdy all! I am trying to setup the LCS communicating with SER and ASTERISK in the company I work for. I read the following tutorial: http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Int… But there the author points the necessity of 2 LCSs, but I just couldnt grasp why the need of a second LCS. Could anyone who has already setup that give me a hint of why it is necessary a second LCS? I thought I could route LCS to communicated directly with SER. I tried to use only one LCS(one that uses TLS) but it didnt communicate with SER. Thanks in advance, Felipe Louback -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "Come to me, all you who are weary and burdened, and I will give you rest." Matthews 11:28

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RE: [Serusers] how to search in usrloc?
by SER LIST 04 Nov '06

04 Nov '06

You do not need to check usrloc t to implement what you want. Define a timer and when you forward the INVITE to the PBX, you will automatically jump to the failure_route[] if you do not receive a response from the PBX (PBX down!) and you can put your logic there. >From: "sandrinho1976(a)libero.it" <sandrinho1976(a)libero.it> >To: "serusers" <serusers(a)lists.iptel.org> >Subject: [Serusers] how to search in usrloc? >Date: Fri, 3 Nov 2006 01:13:24 +0100 > >Hi everyone. > >I'm facing this problem when configuring a Ser as a proxy and a registrar. > >I have the Ser and a IP pbx: the pbx registers itself on the proxy as >pbx@<pbx_ip> (Contact header). >I had to manage INVITEs messages for incoming call that are going >towards the Ser, sent by UACs in the network. > >INVITEs are coming in the form <telephone_number>@<ser_ip> (Req-URI): >Ser had to decide if <telephone_number> is managed by the IP pbx (it has >this information) and eventually, do a rewrite uri to ><telephone_number>@<pbx_ip> and then forward the message to pbx. > >So far, it's all ok. > >Unfortunately, i have to handle also this situation: the ip pbx is down. >At the present, the INVITE is forwarded towards the pbx, the ip pbx does >not respond and after 10 seconds the requesting UACs sends a CANCEL. >This latency is not acceptable. >I'm looking for a way to implement this: >after the rewrite uri to <telephone_number>@<pbx_ip>, look if I have a >contact in usrloc with <pbx_ip> (i.e. the pbx is still registered): if >not, send immediately a response 4xx to UAC. >The problem is that i'm not able to access the usrloc db from the >routing script... any hint? > >Thanks a lot... > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Not only does Windows Live OneCare provide all-in-one PC care to keep your computer protected and well-maintained, but it also makes creating backup files a breeze. Try it today! http://ideas.live.com/programpage.aspx?versionid=b2456790-90e6-4d28-9219-5d…

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Re: [serusers]: trusted table(permissions module)
by Andrey Kuprianov 04 Nov '06

04 Nov '06

The one you attached, didnt have a semicolon. Anyway, try using a fresh copy of a ready made ser.cfg. Bests, Andrey. On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > Yup I always do the same, Might be copy paste error. Its having semicolon & working fine here. > > Regards > Kamal Mann > > -----Original Message----- > From: Andrey Kuprianov [mailto:andrey.kouprianov@gmail.com] > Sent: Thursday, November 02, 2006 10:34 AM > To: serusers(a)iptel.org > Subject: Re: [serusers]: trusted table(permissions module) > > Hi Kamal, > > Whenever you modified your ser.cfg did you restart SER after? Did you > check your ser.cfg for errors using "ser -c" command? I found some > errors in your ser.cfg code. For instance, inside the main route > block, your "if", which processes INVITEs, is missing a semicolon at > the end of the block. > > if (method=="INVITE") > { > # sl_send_reply("404", "INVITE "); > route(3); > break; > } <----- missing semicolon here > > > If you are comfortable with your current ser.cfg, that's ok, but I'd > suggest you start with a fresh one, make your trusted table work, and > then add a "REFER" processing, like in your currect ser.cfg. Just > download one fresh ser.cfg from iptel.org site and add your > allow_trusted() "if" clause there. > > Regards, > > Andrey. > > > On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > Hi > > Please find ser.cfg enclosed. > > > > Trusted table entry: > > +---------------+-------+--------------+ > > | src_ip | proto | from_pattern | > > +---------------+-------+--------------+ > > | 10.25.119.156 | any | ^sip:.*$ | > > +---------------+-------+--------------+ > > 1 row in set (0.00 sec) > > > > Thanks in anticipation > > Kamal Mann > > -----Original Message----- > > From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kuprianov > > Sent: Wednesday, November 01, 2006 7:03 PM > > To: serusers(a)iptel.org > > Subject: Re: [serusers]: trusted table(permissions module) > > > > Hi Kamal, > > > > I understand your setup. Nevertheless, it will be difficult to figure > > out your problem, w/o your ser.cfg file at hand. Also, send a trusted > > table query result (i.e. that row which contains entry for your > > SIP-AS). > > > > Regards, > > > > Andrey. > > > > On 11/1/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > > Hi All > > > In my scenario I need ser to communicate with my SIP-AS. This SIP-AS is having an application xyz running on it. This XYZ sends an invite to URI "A(a)xcv.de" which is a subscriber in SER and this is in TO header of Invite msg & FROM header contains another URI test(a)xcv.de but this URI (test(a)xcv.de) isn't registered / subscribed to SER. I need SER trust all packets from SIP-AS ip. BUT SER is sending 407 reply of INVITE to XYZ application! IP in trusted table is of SIP-AS on top of which XYZ application is running. > > > > > > Thanks in anticipation > > > Kamal Mann > > > > > > -----Original Message----- > > > From: Maciej Żwirski [mailto:mzwirek@poczta.fm] > > > Sent: Wednesday, November 01, 2006 3:20 PM > > > To: serusers(a)lists.iptel.org > > > Subject: Re: [serusers]: trusted table(permissions module) > > > > > > Kamal.Mann(a)t-systems.com wrote: > > > > > > > Hi All > > > > > > > > I made an entry /"10.25.119.156, any, ^sip:.*$"/ into trusted table so > > > > that any packet sent from 10.25.119.156 would be trusted by SER and > > > > don't ask for its credentials. But SER is replying *407- proxy* auth > > > > needed!! After this I tried with permissions.allow file: > > > > > > > Hi, > > > I had the same issue while interconnecting SER and Asterisk. It turned > > > out to be an issue on Asterisk side (I had the same user added for SER > > > and Asterisk, so the poor thing couldn't authorize either :)) So you > > > could check if the 407 you're getting is from SER or from the remote host. > > > > > > Regards, > > > Maciej Zwirski > > > > > > ---------------------------------------------------------------------- > > > Jestes kierowca? To poczytaj! >>> http://link.interia.pl/f199e > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > >

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[Users] State of T.38 and MediaProxy/RTPProxy
by Bastian Schern 04 Nov '06

04 Nov '06

Hello, does anybody knows the state of T.38 with MediaProxy or RTPProxy/NATHelper? Is it supported in MediaProxy v1.7.2 or the current RTPProxy(NATHelper release? Regards Bastian

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[Users] Syntax Error
by Brandon Armstead 03 Nov '06

03 Nov '06

Hello guys, I have been trying to get openser implemented with SEMS the past day or so although I keep running into problems, any advice on setting up openser with SEMS? (Sip Express Media Server). Although here is a syntax error straight from the documentation: ... modparam("tm","tw_append","append1:Email=avp[i:12];UA=hdr[User-Agent]") modparam("tm","tw_append","append2:body=msg[body]") ... t_write_fifo("voicemail/append1","/tmp/appx_fifo"); ... t_write_unix("logger/append2","/var/run/logger.sock"); Is the syntax for "t_write_unix". Although in my code I have the following: modparam("tm", "tw_append", "vm_email:P-Email-Address=avp[i:34]") Just like the example..... although I get error: 0(57475) ERROR:tm:parse_tw_append: parse error in <vm_email:P-Email-Address=avp[i:34]> around position 25(a) Any idea on what is going on? Thanks, all help is appreciated! -- Brandon Armstead

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[Users] Parallel calls with multiple contacts for 1 user
by Jean-François SMIGIELSKI 03 Nov '06

03 Nov '06

Hello, list! I currently use the debian package of OpenSER (v1.1.0-5), and I see a behaviour I do not explain with RFC's and OpenSER Documentation. I have registered several User-Agent under the same username/domain couple in an OpenSER instance with the registrar module. When I try to call "username@domain", I expected OpenSER to call the contact with the highest q-value, and keep the other contacts into transaction branches for later use (as said in the usrloc module's documentation, in the "lookup" function description). In spite of this, I observed that OpenSER called each contact in a parallel way, and the first UA that responds a 200 is the one that receive the final ACK, other branches are closed. Is it wanted / intentional / normal ? Is the documentation up to date? Whatever will your answer be, thank's a lot! Jean-François Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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[Users] Syntax Error
by Brandon Armstead 03 Nov '06

03 Nov '06

Hello, I get syntax error on the following two commands, does anyone have any idea, thanks: avp_db_load("$ru", "i:34/$email_scheme") modparam("tm", "tw_append", "vm_email: P-Email-Address=avp[i:34]") -- Brandon Armstead

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[Serusers] Multiple user registrations using same port number
by Alan 03 Nov '06

03 Nov '06

I am currently running load testing on SER and have a question regarding multiple contacts registered with the same IP and port number but different contact name. Will this work? Does each contact ID have to be defined with a different port number to establish a session? For example: Contact : sip:loadtest0100@10.180.0.71:10140 Contact : sip:loadtest0101@10.180.0.71:10140 Contact : sip:loadtest0102@10.180.0.71:10140 Contact : sip:loadtest0103@10.180.0.71:10140

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[Users] About forking and unregister
by Bill Zhang 03 Nov '06

03 Nov '06

Hi, Anyone know how openser works with following: 1. Forking. Often times, I may have one account register with multiple SIP client, does it support that? I don't seem to make it working. 2. Unregister and register. Nowadays, there are a lot of SIP client will send an Unregister message to the SIP server and after received response, it will send the new registration again, this is because when a client behind the same router reboot multiple times, it might have registration in the SIP server and only the latest one reflect the current registration and all previous registration may not be valid as the router may already close the mapping port, so in RFC3261, it defines client can send a registration with contact head contains "*" and expeiration timer is 0 to unregister the previous registration. It does not seem that OPENSER can handle it. I know SER handles it. 3. Can I specify OPENSER to only send the INVITE (or other SIP request) to the latest AOR for a particular account? Thanks in advance. Bill

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[Users] mediaproxy pulse / jitter and dropout
by Mark Price 03 Nov '06

03 Nov '06

I installed the openser.cfg at http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy (modified to authenticate against my domain, and to use exit instead of break). Calls between two phones on the domain work, but there is a lot of jitter. Well, not exacty jitter, but kind of a pulsing sound in the background of the call. Also, will occasionally drop out, and then come back after a little bit.

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[Serusers] how to search in usrloc?
by sandrinho1976＠libero.it 03 Nov '06

03 Nov '06

Hi everyone. I'm facing this problem when configuring a Ser as a proxy and a registrar. I have the Ser and a IP pbx: the pbx registers itself on the proxy as pbx@<pbx_ip> (Contact header). I had to manage INVITEs messages for incoming call that are going towards the Ser, sent by UACs in the network. INVITEs are coming in the form <telephone_number>@<ser_ip> (Req-URI): Ser had to decide if <telephone_number> is managed by the IP pbx (it has this information) and eventually, do a rewrite uri to <telephone_number>@<pbx_ip> and then forward the message to pbx. So far, it's all ok. Unfortunately, i have to handle also this situation: the ip pbx is down. At the present, the INVITE is forwarded towards the pbx, the ip pbx does not respond and after 10 seconds the requesting UACs sends a CANCEL. This latency is not acceptable. I'm looking for a way to implement this: after the rewrite uri to <telephone_number>@<pbx_ip>, look if I have a contact in usrloc with <pbx_ip> (i.e. the pbx is still registered): if not, send immediately a response 4xx to UAC. The problem is that i'm not able to access the usrloc db from the routing script... any hint? Thanks a lot...

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[Users] two invites, different session description => broken call
by Mark Price 02 Nov '06

02 Nov '06

I have openser configured with asterisk as following: If a sip REGISTER packet arrives, fix_nated_register() save("location") exit; If a sip packet arrives at openser with a source ip != asterisk ip, t_relay(to asterisk) If a sip packet arrives from asterisk, lookup("location"); force_rport() fix_nated_contact() t_relay(); The asterisk dial plan says the following: if inbound traffic arrives from openser's ip address, Dial( SIP/dstuser@dstdomain). phones a and b are on separate networks behind a firewall, openser and asterisk are on public ip addresses. The sip traffc seems to work just fine. I.e. all the handshakes seem to be happening as they should. However, rtp traffic does not. Whether audio traffic will transmit or not is a crap shoot. Sometimes, if a calls b, b hears a but not vice versa, while if b calls a, audio is two way. In this case, one audio stream is going through asterisk, the other is being directed to go point to point. Sometimes, a calls b and b hears a, and a hears b for a second but a second INVITE comes to phone B that causes it to redirect rtp to be point to point. Sometimes there is no audio. Sometimes, everything works fine. At one point, rtp from a was going to asterisk, but asterisk was not sending the rtp on to b, and b was trying to send traffic point to point.

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Re: [Users] TLS: what is encrypted in TLS?
by Ferianto siregar 02 Nov '06

02 Nov '06

Dear Witt, Thank you very much for your help before. Witt, may I ask you again,please? I have downloaded ssldump.tar.gz packet and installed it. As the INSTALL note in ssldump, I have installed libcap.tar.gz (from www.tcpdump.org) and openssl. I have installed it by using commands: # tar -zxvf ssldump-0.9b3.tar.gz # cd ssldump-0.9b3 # ./configure --with-openssl #make #make install All installation process can run sucessfully (nothing error message shown). But, why when I try to run ssldump, I got nothing. There is any mesage shown when I tried to run #ssldump -i eth0 What happens? Did I do wrong installation process? or I have missed something? I do hope you can help me.Please help me.. Thank you very much for your help Regards, Ferianto Note: Actually, when I run ./configure --with-pcap, I got error message. The error message said that there is no pcap file. Why? Is lipcap packet that I got from www.tcpdump.org (as I read from ssldump INSTALL), same with pcap file ? or I have to download pcap ? Steffen Witt <witt.steffen(a)googlemail.com> wrote: Hello Ferianto, TLS is used for encrypting your SIP signaling, not the media data. You should install "ssldump" on your server to see each step of the TLS handshake. Maybe you have a closer look to this short SSL tutorial: http://www.eventhelix.com/RealtimeMantra/Networking/SSL.pdf Best regards Steffen 2006/10/30, Ferianto siregar : > Dear all, > > Thank you for this chance. > All, I have tried to learn about how can TLS can secure communication in > VoIP. > I must to understand it in order to responsible my openser server (working) > with TLS to my lecturer. > But, I have still doubt. > Actually, what is encrypted in TLS system so the communication become > secure? Does it encrypt the data (voise packet)? or The traffic (so the > sniffer can not get the data? OR anything else? > > I do hope anybody can help me :) > Thanks, > > > Ferianto > > > Note: I cryptography tutorials, I read that the cryptography system secure > the message (text) by encrypting it using public key and private key. So, > How about TLS in VoIP? > > > ________________________________ > Everyone is raving about the all-new Yahoo! Mail. > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > --------------------------------- Get your email and see which of your friends are online - Right on the new Yahoo.com

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[Users] sarge didnt install avp_radius
by Zoran Milic 02 Nov '06

02 Nov '06

Hi, I changed my linux distro to debian (from slack) so i can install openser from packages, but aptitude failed to install avp_radius. When I install from source, avp_radius.so is created, after tweaking the Makefile. What is the problem, and how do i get avp_radius module when i install from aptitude/apt-get/dpkg ? best regards, Zoran

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[Users] determining originating address
by Mark Price 02 Nov '06

02 Nov '06

What is the best way to detect whether an INVITE message came from a certain IP address? I.e. if a call comes from a certain asterisk box? I have thought of using the textops search(), but this seems a little clumsy. Perhaps there is a way to get directly at the via headers? Or is there a keyword of corresponding functionality to the uri keyword? Thanks, Mark Price

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[Serusers] please help me out
by vijay tiwari 02 Nov '06

02 Nov '06

hi all i have problem in configurations of mediaproxy module i have download mediaproxy in my ser server and i have make one file name(mediaproxy.ini) in mediaproxy dir and do some change and run this command mediaproxy start nothing can heppand please help me out on this because at this time i am in very bad conditions thanks vijay --------------------------------- Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Everyone is raving about the all-new Yahoo! Mail.

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[Serusers] No Media with Nated Client and rtp proxy
by Ryan Churches 02 Nov '06

02 Nov '06

I just posted about a problem with rtpproxy, but i was able to correct that. now ser is able to connect to rtpproxy, but media is still being lost. someone suggested i put my external ip in the listen= value at the top, but that breaks remote registration. I am following the rtpproxy section of the "Getting Started with Ser Docs" http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html so one would expect i would have ser.cfg set up correctly, but obviously something is wrong. here is some ngrep output. 10.0.0.10 is SER. 10.0.0.110 is a client LOCAL to ser, and 10.0.0.100/67.84.215.54 is a remote client behind yet another nat U 10.0.0.10:6060 -> 10.0.0.110:9267 INVITE sip:1000@10.0.0.110:9267 SIP/2.0. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. To: <sip:1000@bmy.gotdns.org>. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Contact: <sip:1001@67.84.215.54:9170>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 3004w stamp 16863. Content-Length: 285. . v=0. o=- 105355536 105355589 IN IP4 10.0.0.100. s=eyeBeam. c=IN IP4 10.0.0.10. t=0 0. m=audio 20026 RTP/AVP 100 6 0 8 3 18 5 101. a=alt:1 1 : 7D3DDAD1 578F25CA 10.0.0.100 9172. a=fmtp:101 0-15. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=nortpproxy:yes. # U 10.0.0.110:9267 -> 10.0.0.10:6060 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1;received=10.0.0.10. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. . # U 10.0.0.10:6060 -> 67.84.215.54:9170 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. . # U 10.0.0.110:9267 -> 10.0.0.10:6060 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1;received=10.0.0.10. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. . # U 10.0.0.10:6060 -> 67.84.215.54:9170 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. and here is my ser.cfg which ive also attached. # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- 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) listen=10.0.0.10 alias=hostname.dyndns.org port=6060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://user:pw@localhost/ser" # ------------------ 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/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #From Getting Started modparam("auth_db|uri_db|usrloc", "db_url", "mysql://user:pw@localhost/ser") # ----------------- setting module-specific parameters --------------- # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) # -- 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(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; }else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; #======= call type processing section if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" || isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { #------------------------- #REGISTER Message Handler #------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { #---------- #--INVITE Message Handler #----------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { #NAT Traversal if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Contact-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; }

1 0

[Users] 484 UAC handeling
by Helge Waastad 02 Nov '06

02 Nov '06

Hi, I was wondering about the correct handeling of an 484 address incomplete message. I've seen different handeling from UAC to UAC Problem: My grandstreams receives a 484 and returns an ACK. This is how it should.And i hear beep-beep tones. the moment I terminate (on hook), the UAC sends a CANCEL message which generates a baaaad loop in my system.... 1. Is this cancel according to RFC? 2. Is there a way to consume/drop such an cancel? br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 2

[Serusers] Presence ( @msg.supported=~"eventlist")
by SER LIST 02 Nov '06

02 Nov '06

Can someone please explain what the line if ((@msg.supported=~"eventlist")) is testing in the presence example code? This lines always returns false for me and seemingly the XCAP server is not being queried then. Thanks _________________________________________________________________ Ready for the world's first international mobile film festival celebrating the creative potential of today's youth? Check out Mobile Jam Fest for your a chance to WIN $10,000! www.mobilejamfest.com

2 1

[serusers]: trusted table(permissions module)
by Kamal.Mann＠t-systems.com 02 Nov '06

02 Nov '06

Hi All I made an entry "10.25.119.156, any, ^sip:.*$" into trusted table so that any packet sent from 10.25.119.156 would be trusted by SER and don't ask for its credentials. But SER is replying 407- proxy auth needed!! After this I tried with permissions.allow file: "ALL : "sip:mann@imsfw.de$"" but still no progress same response from SER. My ser.cfg entries: modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "source_col", "src_ip") modparam("permissions", "proto_col", "proto") modparam("permissions", "from_col", "from_pattern") modparam("permissions", "default_deny_file","/usr/lib/ser/modules/permissions.deny") modparam("permissions", "default_allow_file","/usr/lib/ser/modules/permissions.allow") Is it correct? Please give your suggestions for the same. Thanks in anticipation Kamal Mann

3 8

[Serusers] WARNING: rtpp_test: can't get version of the RTP proxy
by Ryan Churches 02 Nov '06

02 Nov '06

Im following "Getting Started with SER" and I'm stuck on the chapter about nat traversal Ive tried both methods, currently im working on rtpptoxy. the exact errors I get when i try to make calls from behind NAT are: 0(1731) WARNING: rtpp_test: can't get version of the RTP proxy 0(1731) WARNING: rtpp_test: support for RTP proxy has been disabled 0(1731) ERROR: force_rtp_proxy2: support for RTP proxy is disabled i have compiled and installed rtpproxy, and made sure the permissions were set. bmy bin # ps ax|grep rtp 1714 ? Ss 0:00 ./rtpproxy bmy bin # ls -la /var/run/rtpproxy.* -rw-r--r-- 1 root root 5 Nov 1 19:27 /var/run/rtpproxy.pid srwxr-xr-x 1 root root 0 Nov 1 19:27 /var/run/rtpproxy.sock i have attached my ser.cfg but below are some of the parts of it 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) listen=10.0.0.10 alias=hostname.dyndns.org port=6060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://user:pw@localhost/ser" <skip> modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") <skip> 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(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; }else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; #======= call type processing section if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" || isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { #------------------------- #REGISTER Message Handler #------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { #---------- #--INVITE Message Handler #----------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { #NAT Traversal if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Contact-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; }

1 0

[Serusers] Voice problem
by ram 01 Nov '06

01 Nov '06

Hi all I have installed CDROUTER working fine for 30days now almost all iam able to see 120 users calling at a time no problem but past 7days iam getting lot of voice breaking problems and other side not able to hear my side voice my Server config Xeon with 1 GB Ram 86GB hdd and most of the time looks like below ~# free total used free shared buffers cached Mem: 903484 897328 6156 0 1892 830208 -/+ buffers/cache: 65228 838256 Swap: 0 0 0 when i check top Cpu use is 2% only when i configure directly with provider using PAP2 , voice is clear can some one guide me what is the problem how can i test and how can i resolve this problem help will be appriciate Ram

1 0

[Serusers] Fwd: help in mediaproxy module
by vijay tiwari 01 Nov '06

01 Nov '06

Note: forwarded message attached. Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Check out the New Yahoo! Mail - Fire up a more powerful email and get things done faster.

1 0

[Serusers] my mail are not coming on mail list
by vijay tiwari 01 Nov '06

01 Nov '06

hi all problem my mail are not coming on mail list Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.

1 0

[Users] problem with usrloc db-only mode
by tele＠plexia.com 01 Nov '06

01 Nov '06

Hi all, A simple question about the db-only mode of usrloc (dbmode 3) I've two proxy configured with db_mode 3 the second proxy it's unable to do lookup on the db, there are no error in the debug log but it send reply "404 not found" to the first proxy. I've noticed that when i do lookup("location") on the second proxy it clean the location table and even if the table is filled with the correct contact it reply with 404. sometimes delete all contacts and sometimes contacts remain but failed lookup. what i'm doing wrong? In first proxy i'm running openser-1.1.0 and in the second the latest openser snapshot. the modparam of usrloc on the two proxy are like: modparam("usrloc|domain","db_url","mysql://SQL_USERNAME:SQL_PASSWORD@SQL_HOST/SQL_DB") modparam("usrloc", "db_mode", 3) modparam("usrloc", "matching_mode", 1) thank you, :tele

2 2

Re: [Users] Implementing the CDS(Call Detail Record)
by Bogdan-Andrei Iancu 01 Nov '06

01 Nov '06

1) be sure you are using TM for request forwarding (r_relay) 2) be sure the *db* acc flag is set (and not the one for syslog acc) - see the README 3) look into logs for possible errors. regards, bogdan ShivaSankar Padhy wrote: > > Yah, The same thing now I am looking but also bit confused, I have > used the > > modparam("acc", "log_flag", 2) > and setflag(2) > but I am not able to populate the acc table. > > What is the exact syntax, Can I get it from any where?? > > Thanks a lot for any kind of Help. > Shiva > > On 10/31/06, *Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > you can use mysql - see the docs of the acc module. > > regards > bogdan > > ShivaSankar Padhy wrote: > > > Dear All, > > > > Can I implement the CDR without using the RADIUS Server.??I am using > > openser 1.0.1 notls. Expecting a sooner reply, > > > > Thanks in Advance, > > Shiva > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > >

1 0

[Users] send im in php
by sinclair fils 01 Nov '06

01 Nov '06

when i send im in a script php i have this error 2(3549) ERROR: fifo_server: command must begin with :: MESSAGE 2(3549) **** done consume 2(3549) ERROR: fifo_server: command must have at least 3 chars can help me --------------------------------- Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Profitez des connaissances, des opinions et des expériences des internautes sur Yahoo! Questions/Réponses.

2 1

[Users] New presence server: immediate NOTIFY missing
by Franz Edler 01 Nov '06

01 Nov '06

Hello, many thanks to all developers of the new presence server. Its a great enhancement to OpenSER. As a small contribution from my side I did some tests with Eyebeam clients. Now I found, that the immediate NOTIFY according to RFC3265 § 3.1.6.2 is missing. To document the behaviour I have attached a trace-file from the server. Frame 6 contains a SUBSCRIBE request which is not confirmed with an immediate NOTIFY request. The first NOTIFY is sent only four minutes later after the first presence state change of the presentity occurred (frame 30). Cheers Franz

4 9

[Users] TLS: what is encrypted in TLS?
by Ferianto siregar 01 Nov '06

01 Nov '06

Dear all, Thank you for this chance. All, I have tried to learn about how can TLS can secure communication in VoIP. I must to understand it in order to responsible my openser server (working) with TLS to my lecturer. But, I have still doubt. Actually, what is encrypted in TLS system so the communication become secure? Does it encrypt the data (voise packet)? or The traffic (so the sniffer can not get the data? OR anything else? I do hope anybody can help me :) Thanks, Ferianto Note: I cryptography tutorials, I read that the cryptography system secure the message (text) by encrypting it using public key and private key. So, How about TLS in VoIP? --------------------------------- Everyone is raving about the all-new Yahoo! Mail.

2 1

[Serusers] on hold problem
by Shaun Hofer 01 Nov '06

01 Nov '06

Hi, I've been having a problem, where audio is lost either in one or both directions when conversaion is taken off 'hold'. The parties are both behind NAT and depending UA as whether one or both loose audio. From what I can tell its to do with my loose route and nathelper, and how my ser.cfg deals with the take off hold INVITE from the phones. When the call is taken off hold the rtp streams aren't setup properly again (eg not using mediaproxy correctly). What is the best way to solve this problem? I've seen similarly posts to the mailing list about this problem with no solution posted. http://lists.iptel.org/pipermail/serusers/2006-March/027424.html http://lists.iptel.org/pipermail/serusers/2006-April/027885.html http://lists.iptel.org/pipermail/serusers/2006-May/028407.html Thanks Shaun I have a similarly config to getting started guides ser.cfg # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (!has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if ((method=="INVITE" || method=="REFER") && !has_totag()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; };

3 8

[Users] cpledit
by Doug Meredith 31 Oct '06

31 Oct '06

Hi, I'm trying to find cpledit. It is no longer at iptel.org. Someone on the SER mailing list told me to try here, as he thought Bogdan was behind it. Does anyone know if this still exists and where I can find it? Can anyone suggest an alternative? Thanks for any help. Doug -- Doug Meredith 506-854-7997 ext. 801

2 2

[Serusers] test message
by SER LIST 31 Oct '06

31 Oct '06

pls. ignore _________________________________________________________________ Not only does Windows Live OneCare provide all-in-one PC care to keep your computer protected and well-maintained, but it also makes creating backup files a breeze. Try it today! http://ideas.live.com/programpage.aspx?versionid=b2456790-90e6-4d28-9219-5d…

1 0

[Users] Implementing the CDS(Call Detail Record)
by ShivaSankar Padhy 31 Oct '06

31 Oct '06

Dear All, Can I implement the CDR without using the RADIUS Server.??I am using openser 1.0.1 notls. Expecting a sooner reply, Thanks in Advance, Shiva

2 1

[Serusers] help in mediaproxy module
by vijay tiwari 31 Oct '06

31 Oct '06

hi all i have problem in configurations of mediaproxy module i have download mediaproxy in my ser server and i have make one file name(mediaproxy.ini) in mediaproxy dir and do some change and run this command mediaproxy start nothing can heppand please help me out on this because at this time i am in very bad conditions thanks vijay --------------------------------- Everyone is raving about the all-new Yahoo! Mail.

1 0

[Users] CVS 31.03, TLS and dead tcp child
by Frank DInnocenzo 31 Oct '06

31 Oct '06

Hello, I'm new to openSer and wondered if there was any resolution to this post??? I'm hitting this same crash with openSer (with TLS enabled) using a Snom phone (300 with latest Beta firmware - snom300-6.5-SIP-j.bin). I've gone through the archives looking for help getting this setup working. I've also tried the Snom Softphone with the same result. Thanks Frank >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Hi, I have tried to register to OpenSER CVS 31.03 with my SNOM 360 (5.3.6sw), disabled pretty all modules from OpenSER (radius, xlog, auth, ...) and always the server crashes. Is there something I have missed in configuration ? Thanks a lot for help, -Mika The error message and debug info seems like this: 11(24351) tcpconn_new: new tcp connection to: XXX.XX.XX.XXX 11(24351) tcpconn_new: on port 2171, type 3 11(24351) tls_tcpconn_init: Entered: Creating a whole new ssl connection 11(24351) tls_tcpconn_init: Looking up tls domain [XXX.XX.XX.XXX:5061] 11(24351) tls_tcpconn_init: Using default tls server settings 11(24351) tls_tcpconn_init: Setting in ACCEPT mode (server) 11(24351) tcpconn_add: hashes: 442, 1 11(24351) handle_new_connect: new connection: 0xb608f6a0 24 flags: 0002 11(24351) send2child: to tcp child 0 7(24347), 0xb608f6a0 7(24347) received n=4 con=0xb608f6a0, fd=19 7(24347) DBG: io_watch_add(0x810e580, 19, 2, 0xb608f6a0), fd_no=1 7(24347) tls_update_fd: New fd is 19 11(24351) DBG: handle_tcp_child: dead tcp child 0 (pid 24347, no 7) (shutting down?) 11(24351) DBG: io_watch_del (0x810e420, 17, -1, 0x0) fd_no=16 called 11(24351) ERROR: receive_fd: EOF on 15 11(24351) DBG: handle_ser_child: dead child 7, pid 24347 (shutting down?) 11(24351) DBG: io_watch_del (0x810e420, 15, -1, 0x0) fd_no=15 called 0(24339) child process 24347 exited by a signal 11 0(24339) core was generated 0(24339) INFO: terminating due to SIGCHLD 6(24346) INFO: signal 15 received 6(24346) Memory status (pkg): --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

2 3

[Users] Re: openser
by samuel 31 Oct '06

31 Oct '06

First of all: It's not considered polite to send direct mails for topics which ought to be directed to the mailing list... Try to go to www.onsip.org and download the getting started document, which has a good explanation and several examples for configuring. It is targeted to SER but your openser version should be really close to the config files explained in the getting started document...You might have to adapt some syntax so you would have to check the logs when you start openser. Samuel. 2006/10/31, simo med <simoelmar(a)gmail.com>: > hi Samuel! > i'm asking for an example type of an openser.cfg! if you have any > information or a solution to make my openser work properly, please send it > to me! > i use openser 1.0.1, in linux debian! > thanks in advance > simo

1 0

[Serusers] No query limit ?
by inge 31 Oct '06

31 Oct '06

Hi all, I find a way to show all result for accounting page. I don't want to split it in some pages. So I find in "method.get_acc_entries" the parameter. I write false instead of true. Then I see all entries for the query, but there is always the split in some pages. So for each pages I have all result (lol). Someone have an idea ? Regards.

1 0

[Serusers] SER with a Gateway
by inge 31 Oct '06

31 Oct '06

Hi all, We use SER with a PSTN gateway. My problem is : Although I see my caller ID in result of "show isdn history", my interco provider says that he must put the principal number by default because we don't put the good signalisation for ETSI PRI with the caller ID. That's why on the calling phone, it's not the caller id wich is show on. Someone have an idea ? Is it a problem from SER or Gateway according to you ? (I try to contact Cisco support for this problem too). Regards

1 0

[Serusers] Core dump Presence snapshot 0.10.99-dev35-pa-4.1
by Dave 31 Oct '06

31 Oct '06

The version 0.10.99 of presence snapshot core dumps on a regular basis after running for a certain time. Doing a gdb shows that the core dump occurs at msg_queue_destroy: #0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 147 asm volatile( Doing a backtrace shows: (gdb) backtrace #0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 #1 0x008e0a52 in free_presentity (_p=0xb6293e58) at presentity.c:409 #2 0x008e0b0e in release_presentity (_p=0xb6293e58) at presentity.c:354 #3 0x008de196 in timer_pdomain (_d=0xb626ab78) at pdomain.c:156 #4 0x008d7eb0 in timer_all_pdomains () at dlist.c:234 #5 0x008dd4e7 in timer (ticks=83700431, param=0x0) at pa_mod.c:449 #6 0x080928f1 in compat_old_handler (ti=1339206901, tl=0xb6202738, data=0x1) at timer.c:916 #7 0x08092b6e in slow_timer_main () at timer.c:1026 #8 0x0805e7b5 in main_loop () at main.c:1035 #9 0x0805fda2 in main (argc=1, argv=0xbff59804) at main.c:1627 Any clues wil be appricated.

2 1

[Serusers] serctl domain show ERROR
by lin9k＠yahoo.com 31 Oct '06

31 Oct '06

Hi, When I tried to execute "serctl domain show" but it complained "500 command domain_dump not available". What did I do wrong? Thanks in advance, Ling ____________________________________________________________________________________ Access over 1 million songs - Yahoo! Music Unlimited (http://music.yahoo.com/unlimited)

2 2

[Users] parse_via, bad via?
by Dominik Bay 30 Oct '06

30 Oct '06

Hi, i'm getting sometimes this message: ERROR: parse_via: bad char </> on state 100 ERROR: parse_via on: </UDP ^M ^M > ERROR: parse_via: via parse error ERROR: get_hdr_field: bad via ERROR: bad header field ERROR: parse_msg: message=<SIP/2.0 404 Not Found^M Via: /UDP ^M ^M > ERROR: receive_msg: parse_msg failed it seems that a UA sends ^M which is wrong and openser doesn't support it. Is there any workaround for this? -- Mit freundlichen Gruessen / Kind regards Dominik Bay Cablesurf Technik

1 1

[Serusers] Core dump Presence snapshot 0.10.99-dev35-pa-4.1
by Dave 30 Oct '06

30 Oct '06

The version 0.10.99 of presence snapshot core dumps on a regular basis after running for a certain time. Doing a gdb shows that the core dump occurs at msg_queue_destroy: #0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 147 asm volatile( Doing a backtrace shows: (gdb) backtrace #0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 #1 0x008e0a52 in free_presentity (_p=0xb6293e58) at presentity.c:409 #2 0x008e0b0e in release_presentity (_p=0xb6293e58) at presentity.c:354 #3 0x008de196 in timer_pdomain (_d=0xb626ab78) at pdomain.c:156 #4 0x008d7eb0 in timer_all_pdomains () at dlist.c:234 #5 0x008dd4e7 in timer (ticks=83700431, param=0x0) at pa_mod.c:449 #6 0x080928f1 in compat_old_handler (ti=1339206901, tl=0xb6202738, data=0x1) at timer.c:916 #7 0x08092b6e in slow_timer_main () at timer.c:1026 #8 0x0805e7b5 in main_loop () at main.c:1035 #9 0x0805fda2 in main (argc=1, argv=0xbff59804) at main.c:1627 Any clues wil be appricated.

1 0

[Users] Does anyone use CPL-C module?
by Doug Meredith 30 Oct '06

30 Oct '06

Hi, I'm quite interested in CPL but I see almost no discussion of this on the list. Is anyone actually using this module? Is it worth looking at? Thanks for any info you can give me. Doug -------------------------------------------- Doug Meredith SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com

3 3

[Serusers] BYEs are not written to acc if gateway sends BYEs resulting from RTP timeout
by Ricardo Carvalho 30 Oct '06

30 Oct '06

Hi all, I've implemented RTP timeout in the gateway that I use along with Ser, to be able to overpass problems of wrong accounting resulting from power or network failure of some UA. This is apparently working fine as my gateway sends to Ser BYE messages after the established timeout, although only now I realised that Ser doesn't account those BYE messages!!! Why? Following ser.cfg code, I can see that the execution reads the setflag(1) that I use for accounting, but it seems that Ser doesn't care for these BYEs!! Any help would be much appreciated. Thanks in advance, Ricardo.

1 1

Fwd: [Users] Two Invites simultaneously
by ShivaSankar Padhy 30 Oct '06

30 Oct '06

Thank you Vamsi, The problem is for the saving(location) twice, Thanks a lot for cooperating and help me in solving the problem. Thanks, Shiva ---------- Forwarded message ---------- From: Vamsi Pottangi <vamsipottangi(a)gmail.com> Date: Oct 30, 2006 5:32 PM Subject: Re: [Users] Two Invites simultaneously To: ShivaSankar Padhy <shivapd1983(a)gmail.com> Thats good. I feel you should reply to the mailing list saying that problem is solved and it is a mistake in cfg where you were saveing the location twice. This will help people later who would face similar problem. ~Vamsi On 10/30/06, ShivaSankar Padhy <shivapd1983(a)gmail.com> wrote: > > > I am extremely Sorry, Actually, I was having Authentication problem, thats > why I have used the Same(please refere to the commented lines near by > save(location), I have commented when the problem is short out buy > bymistakely, I have not commented the Save(location). Well, Its a my foolish > fault. > > Thanks a Lot, It's worked now.Than ks a Lot. > > Bye, > Shiva > On 10/30/06, Vamsi Pottangi < vamsipottangi(a)gmail.com > wrote: > > > > Remove the first occurence of the save("location") in your script and > > try. Is there any reason why your have two occurences of save("location") ? > > > > On 10/30/06, ShivaSankar Padhy <shivapd1983(a)gmail.com> wrote: > > > > > > Yah, I forgot to Attch the file,Hear is the attachment. Hoping for a > > > quick response. > > > > > > Thanks and regards, > > > Shiva > > > > > > On 10/29/06, Vamsi Pottangi < vamsipottangi(a)gmail.com> wrote: > > > > > > > > > > > > Siva, > > > > > > > > Please send us the openser.cfg ... it is missing in your previous > > > > mail .... > > > > > > > > ~Vamsi > > > > > > > > On 10/29/06, ShivaSankar Padhy < shivapd1983(a)gmail.com> wrote: > > > > > > > > > Hi every one, > > > > > I am a new user of the openser.When I am trying to call to a > > > > > number, the invite messege is going to proxy server but from proxy, its > > > > > sending two invites simultaneously to two phones. > > > > > > > > > > I am attaching my openser.cfg for your kind information.Point to > > > > > be noted that, when the phone resisters, no value is being inserted in to > > > > > the location table of openser Database. > > > > > > > > > > Thanks in Advance, > > > > > Shiva > > > > > > > > > > _______________________________________________ > > > > > Users mailing list > > > > > Users(a)openser.org > > > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > >

1 0

[Users] Openser ALG
by laoli 30 Oct '06

30 Oct '06

Hello, Has anyone configured Openser to as an Application Layer Gateway? Is this configuration possible? Any hints would be greatly appreciated. :-) Thanks!

3 3

[Serusers] cpled
by Doug Meredith 30 Oct '06

30 Oct '06

Hi, I am searching for cpled. I searched iptel.org, google, and the mailing list. The only thing I could find was this page http://old.iptel.org/products/cpled/. The page it links to for download doesn't seem to work anymore. Does this product still exist? Thanks for any help. Doug -- Doug Meredith 506-854-7997 ext. 801

2 1

[Users] contact header radius accounting
by Christian Schlatter 30 Oct '06

30 Oct '06

hi all, I'd like to include caller and callee SIP contact header in radius accounting messages. Our dynamic QoS provisioning framework needs this information to set up QoS per source/destination IP address pair. Caller contact header is easily done using radius_extra and setting an AVP in the INVITE handling route block. But I haven't figured out yet how I could store the callee's contact header in an AVP. The main problem I have is that I can't use any AVP related functions in onreply_route blocks. What I'd like to do is something like modparam("acc", "radius_extra", \ "Originating-Station-IP=$avp(AVP_ORIG_IP); \ Destination-Station-IP=$avp(AVP_DEST_IP)") onreply_route[ONREPLY_DEFAULT_RELAY] { if (t_check_status("2[0-9]{2}")) { avp_write("$ct", "$avp(AVP_DEST_IP)"); } } I've learned from an earlier mail thread (http://www.openser.org/pipermail/users/2005-October/001272.html) that enabling AVP functions in on_reply routes would be rather difficult. Is there still a chance that on_reply AVP handling gets implemented? And is there an alternative method how this could be achieved without patching the acc module? thanks, Christian

2 1

[Users] Redirect with openSER depending on the DNS SRV response
by Oezguer Akkaya 30 Oct '06

30 Oct '06

Dear OpenSER Users or Admins, I would like to setup OpenSER as a SIP Redirect Server in my environment. The point is to redirect depending on the response from DNS or even better DNS SRV. The SIP UA will get than as quick as possible the 3xx Response and resend the request to the redirected address (IP or Domain address) SIP Device. Is there a possible way to do that via config? How many SIP Request (e.g. REGISTER, INVITE) will it be possible to redirect per second? Thanks in Advance for your responses Oetzi

2 1

[Users] Two Invites simultaneously
by ShivaSankar Padhy 30 Oct '06

30 Oct '06

Hi every one, I am a new user of the openser.When I am trying to call to a number, the invite messege is going to proxy server but from proxy, its sending two invites simultaneously to two phones. I am attaching my openser.cfg for your kind information.Point to be noted that, when the phone resisters, no value is being inserted in to the location table of openser Database. Thanks in Advance, Shiva

3 2

[Users] Why is a second RPID header created when I do two subst()?
by Alan Crosswell 30 Oct '06

30 Oct '06

Please ignore previous post which I sent before completing! I have two problems: 1. A 2nd Remote-Party-ID header gets appended when I try to do two subst() on it. 2. Xlog seems to only have a ref to the orignal unedited rpid header. I am attempting to clean up my Remote-Party-ID header so that my Polycom UAs will display the incoming number as a number rather than as a URI. This requires the domain to match that of the UA's server (e.g. siptest.columbia.edu). And, I want to rewrite incoming calling numbers from my PBX to be in 5-digit format and other US 10-digit numbers into proper E.164 format since my PBX and at least one of my ITSPs doesn't provide them as E.164. So, I do a couple of subst() operations on INVITEs. What's wierd is that one Remote-Party-ID header becomes two when it gets to the UA. The call flow is: INVITE r-uri <sip:10508@128.59.59.96:5060> from <sip:9174147124@128.59.59.242> With the original RPID header as: Remote-Party-ID: <sip:9174147124@128.59.59.242> And after route[10] the headers that end up on the UA are: Remote-Party-ID: <sip:9174147124@siptest.columbia.edu> Remote-Party-ID: <sip:+19174147124@128.59.59.242> It looks like the 2nd subst() happened against the original RPID, not the edited one and somehow two RPID headers got created. Here's the code for the route block: route[10] { xlog("L_INFO","route[10] $ci: method $rm r-uri <$ru> from <$fu> rpid <$re>\n"); # first just clean out the domain name subst('/^(Remote-Party-ID:.*<sip:.*)@.*>(.*)$/\1(a)siptest.columbia.edu>\2/'); # now strip our numbers down to 5 digits subst('/^(Remote-Party-ID:.*<sip:)(\+)?(1)?21285([134][0-9]{4})@/\1\2/'); # expand 10-digit NANP numbers to E.164 subst('/^(Remote-Party-ID:.*<sip:)([0-9]{10}.*$)/\1+1\2/'); # ??? xlog doesn't show the result of rewriting it. # xlog("L_INFO","route[10] RPID is now <$re>\n"); xlog("L_INFO","route[10] $ci: RPID was rewritten.\n"); return; } #end of route[10] Any help with this would be greatly appreciated! /a

2 1

[Users] openser.cfg
by simo med 30 Oct '06

30 Oct '06

dear sir or madame; i'm trying to start an openser, but i can't, i need to configure properly my openser.cfg! if you have an example of how can i configure my openser.cfg, to make it work, i will very thanksfull simo

2 1

[Users] How to xlog failure status in failure_route?
by Alan Crosswell 30 Oct '06

30 Oct '06

I tried xlog $rs $rr to log the reason for a failure in failure_route but it just logs NULL. What am I doing wrong? The error status is available to test with t_check_status and is logged by ACC but not xlog. # Error from media gateway failure_route[4] { xlog("L_INFO","failure_route[4] $si: method $rm r-uri <$ru>\n"); if (t_check_status("(5..)|(6..)|(403)")) { # gateway failure/refusal xlog("L_INFO","failure_route[4] gateway returned 5xx or 6xx. Status: $rs $rr\n"); if (!next_gw()) { xlog("L_CRIT","No more gateways for <$tu>\n"); t_reply("503", "Service not available - No more gateways"); return; } else { xlog("L_INFO","failure_route[4] next gw r-uri <$ru> \n"); t_on_failure("4"); t_relay(); return; } } xlog("L_INFO","failure_route[4]: relaying normal error\n"); t_relay(); } Oct 26 18:02:04 jello /usr/sbin/openser[2028]: ACC: call missed: method=INVITE, i-uri=sip:9319174147124@siptest.columbia.edu;user=phone, o-uri=sip:19174147124@67.151.91.22:50 60;transport=udp, call_id=c141e612-75cfa484-cfce783b(a)128.59.37.218, from="Alan Crosswell" <sip:10508@siptest.columbia.edu>;tag=CA5BDB76-6DB44EED, code=403 Forbidden Oct 26 18:02:04 jello /usr/sbin/openser[2028]: failure_route[4] 128.59.37.218: method INVITE r-uri <sip:19174147124@67.151.91.22:5060;transport=udp> Oct 26 18:02:04 jello /usr/sbin/openser[2028]: failure_route[4] gateway returned 5xx or 6xx. Status: <null> <null> Oct 26 18:02:04 jello /usr/sbin/openser[2028]: failure_route[4] next gw r-uri <sip:19174147124@67.151.91.22:5060;transport=udp>

2 1

[Users] Adapting the openserctrl file to a UNIX odbc client...
by Marvin Ottley 30 Oct '06

30 Oct '06

Hello all, I am looking for some information/pointers on how to adapt the openserctl file to use a UNIX ODBC client. Thanks, Marvin

2 1

[Users] Ring group
by Marcos Biscaysaqu 29 Oct '06

29 Oct '06

Hi there. I just getting my Opserser server up and I want to be able to create child accounts from my system to be able to ring multiple extension at the same time, is this possible ? Thanks Marcos Biscayaqu

2 1

[Users] openser and PSTN
by geovani ferronetti 29 Oct '06

29 Oct '06

Hi, I want to connect OpenSER with PSTN, as media gateway I use Cisco AS5300, vfc c542. I am interested about config part from OpenSER. Where I can get some documentation, information? Thx __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i tuoi file e i messaggi http://mail.yahoo.it

2 2

[Users] yate client windows
by sinclair fils 29 Oct '06

29 Oct '06

hello, all someone can tell me how i can configurate yate client to work which openser when i make a config the account cant register please help ___________________________________________________________________________ Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Profitez des connaissances, des opinions et des expériences des internautes sur Yahoo! Questions/Réponses http://fr.answers.yahoo.com

4 3

[Users] Why is a second RPID header created when I do two subst()?
by Alan Crosswell 28 Oct '06

28 Oct '06

I have two problems: 1. I am attempting to clean up my Remote-Party-ID header so that my Polycom UAs will display the incoming number as a number rather than as a URI. This requires the domain to match that of the UA's server (e.g. siptest.columbia.edu). And, I want to rewrite incoming calling numbers from my PBX to be in 5-digit format and other US 10-digit numbers into proper E.164 format since my PBX and at least one of my ITSPs doesn't provide them as E.164. So, I do a couple of subst() operations on INVITEs. What's wierd is that one Remote-Party-ID header becomes two when it gets to the UA. The call flow is: INVITE r-uri <sip:10508@128.59.59.96:5060> from <sip:9174147124@128.59.59.242> rpid <sip:9174147124@128.59.59.242> Here's the code for the route block: route[10] { xlog("L_INFO","route[10] $ci: method $rm r-uri <$ru> from <$fu> rpid <$re>\n"); # first just clean out the domain name subst('/^(Remote-Party-ID:.*<sip:.*)@.*>(.*)$/\1(a)siptest.columbia.edu>\2/'); # now strip our numbers down to 5 digits subst('/^(Remote-Party-ID:.*<sip:)(\+)?(1)?21285([134][0-9]{4})@/\1\2/'); # expand 10-digit NANP numbers to E.164 subst('/^(Remote-Party-ID:.*<sip:)([0-9]{10}.*$)/\1+1\2/'); # XXX dip the calling name database XXX # ??? xlog doesn't show the result of rewriting it. # xlog("L_INFO","route[10] RPID is now <$re>\n"); xlog("L_INFO","route[10] $ci: RPID was rewritten.\n"); return; } #end of route[10]

1 0

[Serusers] Accounting problems after mysql backup restore
by C. Mewes 28 Oct '06

28 Oct '06

After backup and restoring MySQL the accounting did not worked anymore. I could make calls but the amount does not show up. The problem is that o_uri has the format sip:00 + codigo de tarifa pero en la base de datos se esta guardando sin el 00 solamente sip:+codigo de tarifa ( la razon no la se, no he modificado nada de eso ). After changig in accounting $codigo2 = "sip:".$row2->codigo; if ( strncmp ( $codigo2, $row->o_uri, strlen($codigo2) ) == 0) { $tarifa=$row2->tarifa; $nombre=$row2->nombre It works again.What is wrong??? Thanks to everybody who can help. CFM

1 0

[Users] failure route doubts
by Jose Silva 27 Oct '06

27 Oct '06

Hi, I'm having some doubts on how to use/do some things with dispatcher module and failure route. I'll try to explain my problem. On my main route block I select the destination (using ds_select_dst) to where I want to send the message, and indicate that responses should pass through the failure_route, I also add a header with some things that are needed by the next proxy. On the failure route when I get 408 Time Out I mark the previous destination and select the next destination. until here everything goes fine. After this I want to change the message that I'm going to send to next proxy, for this I need to remove the header that I inserted previously on the main route block and access the destination URI that the ds_next_dst updates. My first doubt is that when I print the destination URI it has the previous destination ( the one used in the main route block) but when it relay's the message it goes to the right destination (the one returned from ds_next_dst), Is there any reason for this ? I'm using $du to access it, is this wrong? My second doubt is, what sip message do I have access to in the failure route (the one I received or the one I sent out)? I ask this because I try to remove the header I inserted in the main route block and it does not remove anything but when I relay the message the header I tried to remove goes on the message. Thanks in advance, José Carlos Silva

4 7

[Serusers] some tunning to syslog messages
by Ricardo Carvalho 27 Oct '06

27 Oct '06

Hi all, Along my ser.cfg code I want to print a formatted message using LOG functions like xlog implemented with xlog module of Ser. I already can do it, but now I pretend to write in the same xlog formatted message, some formatted text that can be splitted in multiple lines. I thought that it could be done using \n sintax, like in C, but some testing made me realise that it doesn't work. I also found that this isn't a problem of xlog module of Ser, but a limitation of Syslog service because doing a simple program like this won't result like expected: ////////////////////////////////////////////////////////////////////////////////////////////////////////////// #include <syslog.h> #include <stdio.h> int main(int argc,char ** argv){ char *message = "This is a messagea and this is the rest"; int priority = LOG_INFO | LOG_LOCAL6; syslog(priority,"This is a message %c and this is the rest",13); return 0; } ////////////////////////////////////////////////////////////////////////////////////////////////////////////// The output of this program results in the same line of syslog dump. In ser.cfg if I do something like the following line, the resulting print to syslog is still written in one line: xlog("L_INFO", "===> received [%rm] request\n From URI: [%fu]\n"); Syslog simply ignores the \n parameter. Is there any other way to split xlog formatted messages of ser.cfg to be written to Syslog? Thanks in advance, Ricardo.

1 1

[Serusers] Where to ask questions about CDRTool
by Lokesh Kumar 27 Oct '06

27 Oct '06

Hello List, Can anybody tell me where to ask questions about CDRTool. Thanks very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.14/502 - Release Date: 10/27/2006

1 0

[Serusers] New route graph utility
by Nils Ohlmeier 27 Oct '06

27 Oct '06

Hello, in case you are not following the CVS commit log messages: I just checked in a small python script which should be able to print a dependency graph in ASCII of the route blocks of a SER config file. When you try to understand more complex configuration files this tool might be helpfull. But be warned: this is still alpha code! It will not eat your config file, but the output might not meet your expectations :-) If you run it with the default configuration file the output will look like this: ./route_graph.py ../../etc/ser.cfg Main | \- 1 | \- 1 | \- 1 | \- 1 A more complex scenario (from a config with named routes) could look like this: ./route_graph.py ./ser.cfg Main | \- ROUTE_RPC | | | \- ROUTE_ERROR_REPLY | \- ROUTE_PROXYINIT | | | \- ROUTE_ERROR_REPLY | \- ROUTE_INDIALOG | | | \- ROUTE_ERROR_REPLY | | | \- ROUTE_ST | | | \- ROUTE_FORWARD | | | | | \- ROUTE_LI | | | | | \- ROUTE_RTPPROXY | | | | | \- ROUTE_RR | \- ROUTE_DOMAINPOLICY | | | \- ROUTE_ERROR_REPLY | \- ROUTE_REGISTRAR | | | \- ROUTE_ERROR_REPLY | \- ROUTE_AUTHENTICATION | | [... omitted bigger parts...] Send routes ----------- onsend Failure routes -------------- FAILURE_ROUTE | \- ROUTE_RTPPROXY | \- ROUTE_VOICEMAIL | | | \- ROUTE_SEMS | | | | | \- ROUTE_LI | | | | | \- ROUTE_RR | | | \- ROUTE_ERROR_REPLY | [... omitted bigger parts...] Onreply routes -------------- REPLY_ROUTE Feedback is welcome. Patches even more! :-) Greetings Nils

4 6

[Users] openser domain
by simo med 27 Oct '06

27 Oct '06

dear sir or madame; i enjoy to be one of the members of your great site! i think you for all your pretty informations that you give us. If you want to send me how to create an openser domain! because when i log a user softphone to the openser, i see that it not log to the server! i think that i have problem with domain name. if i must create a domain name to make it work please send me how to do it, and if there is another way send me too, i need your help! for more information i have installed mysql, openser in a debian linux platform, and all is ok. i choose x-lite for a softphone example to log to the openser server. your faithfully simo

2 1

[Serusers] Jslee Arch
by Kamal.Mann＠t-systems.com 27 Oct '06

27 Oct '06

Hi All Do SER architecture based on JSlee architecture? Regards Kamal Mann

3 4

[Serusers] memory allocation error
by Lol Zimmerli 27 Oct '06

27 Oct '06

Hello, In my ser.cfg I would like to add a lot of peers in this form: ... if (uri=~"^sip:\*123[0-9]*@.*"){ xlog("L_INFO", "Forwarding *123 to 'other.sip.com': %fu (%ua - %is) -> %tu, %mf\n"); strip(4); rewritehost("other.sip.com"); } ... If I put four of them, everything is okay, ser starts and the calls are correctly redirected. But if I put 1000 of them: Oct 27 02:55:57 ser ser: ERROR:lex:addstr: memory allocation error Oct 27 02:55:57 ser last message repeated 71 times Oct 27 02:55:57 ser ser: ERROR: mk_action: memory allocation failure Oct 27 02:55:57 ser ser: ERROR: mk_action: memory allocation failure Oct 27 02:55:57 ser ser: ERROR:lex:addstr: memory allocation error Oct 27 02:55:57 ser last message repeated 12 times Oct 27 02:55:57 ser ser: ERROR: mk_action: memory allocation failure Oct 27 02:55:57 ser ser: ERROR: mk_action: memory allocation failure Oct 27 02:55:57 ser ser: ERROR:lex:addstr: memory allocation error Oct 27 02:55:57 ser last message repeated 20 times Oct 27 02:55:57 ser ser: ERROR: mk_elem: memory allocation failure Oct 27 02:55:57 ser ser: ERROR:lex:addstr: memory allocation error I tried to change #define SHM_MEM_SIZE 32 into #define SHM_MEM_SIZE 128 in config.h, but same result. Also, in some old doc, a config item called 'shared_mem_size', but adding shared_mem_size=256 in ser.cfg just leads to a Oct 27 03:04:16 ser ser: parse error (30,1-16): syntax error I am quite sure it's just a number to change, but where? Thanks in advance -- # Lol Zimmerli // S y s C o ® // http://www.sysco.ch/ Keep it right when you make it faster. - The Elements of Programming Style (Kernighan & Plaugher)

3 3

[Users] OpenSER summit - registration will be closed Monday, Nov 30
by Daniel-Constantin Mierla 27 Oct '06

27 Oct '06

Hello everybody, the registration for OpenSER summit will be closed Monday, November 30. Afterwards, you can register via VoN, the access will be free for those having access to VoN conference or exhibition, if there is space in the summit room. There are over 80 registered people, about 20 special guests, any many other executives present at VoN were invited to visit us. The agenda is now in final shape (http://www.openser.org/index.php?option=com_content&task=view&id=60&Itemid=…). This was quite a successful campaign, within a short time, and the event promises a lot. In about 3 weeks we managed to have everything in place. Special thanks to sponsors and those which helped to have it possible. Summit page: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… Cheers, Daniel

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[Users] Problems with Mediaproxy. Bad Port.
by Ricardo Martinez 27 Oct '06

27 Oct '06

Hello list. I'm having a problem with my mediaproxy box. I have a NAT'd client (a softphone) calling to a SIP-H323 conversor through my OpenSER-Mediaproxy box and then to a Cisco Call Manager (it's a very weird configuration). Anyway, the problem presents when the 200 - OK message arrives from my conversor, the SDP part indicate the media address (10.10.148.230) and port (16952) where the remote endpoint will listen RTP, despite of this the mediaproxy open a session for the port 17360. The escenario is something like this. 100.100.148.248 ------- 100.100.148.246 ----- 100.100.154.36 ------ 100.100.154.119 ---- 100.100.148.230 Softphone OpenSer/Mediaproxy Proxy Sip siph323 conversor Call Manager.H323 Please check the log from the mediaproxy and the execution of the "sessions.py" utility, plus the OK message. U 100.100.154.36:5060 -> 100.100.148.246:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.148.246;branch=z9hG4bK67ea.aa12dc97.0. Via: SIP/2.0/UDP 192.168.1.101;received=100.100.148.248;rport=46096;branch=z9hG4bKc0a801650000001d4533a0ee00007c2900000011. Record-Route: <sip:100.100.154.36;ftag=2214787970;lr=on>. Record-Route: <sip:7072408196@100.100.148.246:5060;nat=yes;ftag=2214787970;lr=on>. From: "unknown"<sip:5502408196@sip.desa.mydomain.net>;tag=2214787970. To: <sip:7072408196@sip.desa.mydomain.net>;tag=2260729709. Call-ID: A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101. Cseq: 2 INVITE. Date: Mon, 16 Oct 2006 15:10:37 GMT. Server: siph323. Content-Type: application/sdp. Content-Length: 162. Contact: sip:7072408196@100.100.154.119. . v=0. o=100.100.154.119 1 505533324 IN IP4 100.100.154.119. s=SIP Library call. c=IN IP4 100.100.148.230. t=3370000237 0. m=audio 16952 RTP/AVP 0. a=rtpmap:0 PCMU/8000. Oct 16 12:12:00 sip mediaproxy[18289]: lookup A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101 100.100.148.230:16952:audio 100.100.154.36 sip.desa.mydomain.net local sip.desa.mydomain.net unknown siph323 info=from:5502408196@sip.desa.mydomain.net,to:7072408196@sip.desa.mydomain.net,fromtag:2214787970,totag:2260729709 Oct 16 12:12:00 sip mediaproxy[18289]: execution time: 0.48 ms Oct 16 12:12:00 sip mediaproxy[18289]: session A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101: caller signed in from 100.100.148.248:49154 (RTP) (will return to 100.100.148.248:49154) Oct 16 12:12:00 sip mediaproxy[18289]: session A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101: called signed in from 100.100.148.230:17360 (RTP) (will return to 100.100.148.230:17360) [root@sip ]# /usr/src/mediaproxy-1.7.2/mediaproxy/sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------------------------------------------------------ 100.100.148.248:49154 - 100.100.148.246:60022 - 100.100.148.230:17360 active 0'39" G711u Audio 336.91k/669.53k/332.62k Total traffic: 79.69kbps/78.12kbps/157.81kbps (in1/in2/out) Session count: 1 Proxy version: 1.7.2 Doing more debugs i discover the RTP traffic coming from the 100.100.148.230 it has source port 17360, but the listening port is not the same (i guess is the one indicated in the SDP). Why mediaproxy assumes the 17360 port is for listening too and ommit the SDP 16952 port ? Is this maybe an asymmetric regarding the RTP media?. If so, i added this line into the rtp-asymmetric-clients file: # # Lines starting with a '#' character or empty lines are ignored. # Put a User-Agent name or regular expression per line. # Check is case insensitive. # # This file should only list SIP clients that are asymmetric regarding # the RTP media streams. For clients that are asymmetric regarding the SIP # signaling, use the equivalent SIP version of this file. # Clients that are asymmetric for both SIP signaling as well as RTP media # streams should go in both files. # siph323 But not solve the problem. Can someone help me here? Thanks in advance Regards Ricardo Martinez.- RedVoiss.

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[Users] Q:ABOUT:400 Bad Request RESPONESE
by Alejandro Sanchez 27 Oct '06

27 Oct '06

Hi evreybody My sip server sends this message to the Phone. INVITE sip:192.168.1.10:7191 SIP/2.0 Record-Route: <sip:192.168.1.20;lr=on;ftag=1604fbda> Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bK2f9.039444a7.0 Via: SIP/2.0/UDP 192.168.1.9:7191;branch=z9hG4bK71Fso4gTlg Max-Forwards: 69 Content-Type: application/sdp To: <sip:2109075@192.168.1.20> From: "4422109087" <sip:4422109087@192.168.1.20>;tag=1604fbda Call-ID: 5f1e5301-40b07670(a)192.168.1.9 CSeq: 1596803196 INVITE Allow: INVITE, ACK, CANCEL, MESSAGE, OPTIONS, BYE, UPDATE, REFER, SUBSCRIBE, NOTIFY, SERVICE User-Agent: URMET-UserAgent Contact: "4422109087" <sip:4422109087@192.168.1.9:7191> P-hint: usrloc applied v=0 o=ntzum 16015906 936435728 IN IP4 192.168.1.9 s=- c=IN IP4 192.168.1.9 b=CT:31 t=0 0 m=audio 49648 RTP/AVP 4 96 97 98 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:96 AMR/8000 a=rtpmap:97 X-G723-7200/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-15 m=video 49650 RTP/AVP 99 100 34 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 packetization-mode=1 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 and i have like response a 400 Bad Request. Any hava an idea why? Thank's in Advanced. Alejandro Sanchez ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 1

[Serusers] variables in log messages
by Mark Price 26 Oct '06

26 Oct '06

Is there a method to print variables into log messages? For example, something like this: instead of saying: if(method=="SUBSCRIBE") { log(1,"SUBSCRIBE\n"); } do this log(1,method) which obviously doesn't work. It would be very helpful to be able to print out things like headers. Perhaps the select or more likely AVP functions will work in 0.10? Is there anything available now? Mark

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[Users] uri or to_uri
by Gregoire 26 Oct '06

26 Oct '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi the list! I am just wondering the difference between the variables "uri" and "to_uri". I saw that "to_uri" refers to the value of the field "To", but I can not see at which field refers "uri". Does someone know something about that? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFQHp6I8gmGeMTr0sRAgbuAKCmISnqxdAvCQeWTsJjxtNoqoCY8gCeJA8c kW4+/zwvtIJXofKjTTVwrGA= =j0qD -----END PGP SIGNATURE-----

2 1

[Users] BAD REQUEST
by Jose Gil Navarrete 26 Oct '06

26 Oct '06

Hi. My problem is the next. When my sip server send the message INVITE to the phone, the phone always respond with 400 Bad Request, I think is something wrong in the message, in the UDP message the Checksum always is corrupted by the sip server, is it the problem? or is something else in the sip message? How I can correct this? this is the trace: User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 7191 (7191) Source port: 5060 (5060) Destination port: 7191 (7191) Length: 1057 Checksum: 0x879f [incorrect, should be 0xb0cd] Session Initiation Protocol Request-Line: INVITE sip:192.168.1.8:7191 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 9 Message Header Record-Route: <sip:192.168.1.20;lr=on;ftag=db79a2e2> Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bKdfe7.2eeeaa64.0 Via: SIP/2.0/UDP 192.168.1.7:7191;branch=z9hG4bKN2IVzrueyM Max-Forwards: 69 Content-Type: application/sdp To: <sip:2109075@192.168.1.20> From: "4422109087" <sip:4422109087@192.168.1.20>;tag=db79a2e2 Call-ID: eea4130a-277bd7f8(a)192.168.1.7 CSeq: 1503273304 INVITE Allow: INVITE, ACK, CANCEL, MESSAGE, OPTIONS, BYE, UPDATE, REFER, SUBSCRIBE, NOTIFY, SERVICE User-Agent: PHONE-UserAgent Contact: "4422109087" <sip:4422109087@192.168.1.7:7191> P-hint: usrloc applied Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): ntzum 717537891 1011557901 IN IP4 192.168.1.7 Session Name (s): - Connection Information (c): IN IP4 192.168.1.7 Bandwidth Information (b): CT:29 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 49648 RTP/AVP 4 96 97 98 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:96 AMR/8000 Media Attribute (a): rtpmap:97 X-G723-7200/8000 Media Attribute (a): rtpmap:98 telephone-event/8000 Media Attribute (a): fmtp:98 0-15 Media Description, name and address (m): video 49650 RTP/AVP 99 100 34 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:99 H264/90000 Media Attribute (a): fmtp:99 packetization-mode=1 Media Attribute (a): rtpmap:100 H263-1998/90000 Media Attribute (a): rtpmap:34 H263/90000 Regards! --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 1

[Users] Mysql table version problem in cvs version
by Pletli Antal 26 Oct '06

26 Oct '06

Hi, I have a problem with openser cvs version. Before the installing the cvs version i dropped the previous openser database. I have installed the new version, and after make dbinstall the "version" table contains the following record for version silo: | silo | 5 | but the debug log shows this problem: 0(0) table_version(): Invalid type (2) or nul (0) version columns for silo 0(0) MSILO:mod_init: Wrong version v-1 for table <silo>, need v5 0(0) init_mod(): Error while initializing module msilo What did I wrong? Thanks in advance, Antal

2 1

[Users] pbxnsip and openser
by Nick De Cristofaro 26 Oct '06

26 Oct '06

Anyone here using pbxnsip as their BB user agent with OpenSER. I'm wondering what steps would be required to make it work. Is it a complicated process? Thanks Nick

2 1

[Users] Replication ACK problem
by Dioris Moreno 26 Oct '06

26 Oct '06

Hi everybody I have a working installation of OpenSER (Proxy A). In this installation, the ACK messages are delivery without any problem through loose_route() and t_relay(). Then, I installed a second OpenSER proxy (Proxy B) and configure both servers to replicate REGISTER messages with t_replicate(), and it works fine. I made a lot of captures to check the replication process was working fine and also to check the NAT pings; each proxy takes care of the NATed clients it receives sending OPTIONS message periodically. I'm using mediaproxy to support NATed clients; to avoid each proxy starting a media session when it find a NATed client I used the following rule: a proxy receiving messages from another proxy never starts a media session, the proxy that sends the message takes care of that. Having two registered clients (Client A in Proxy A and Client B in Proxy B), I start making calls from Client A to Client B, but I found something in the captures. All the messages flow correctly, except the final ACK message sent by Client A to Client B. Proxy A receives this message and pass it to Proxy B, but Proxy B never deliver it to Client B. Of course, the media session never starts. I was searching info about this situation and I found an email respond from Bogdan about using t_newtran() to process ACK messages, I tried, but it didn't work. Has anybody ever had the same problem? Is there a special way to deal with ACK messages when we're working with replicated proxies? I'll appreciate any help.

2 1

[Users] Conference is Not Working.... with OpenSER And Asterisk
by sunkara 26 Oct '06

26 Oct '06

Hello Users, Good Morning, I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql modules. And I'm not Using the Zapptel Cards. 9001 ----------> dial 19001(conference Users)-------openSER ---------> Asterisk ------------------------------------------------------------------------ *In Extension.conf * [from-sip] exten => 19001,1,Playback(conf-hasentered) ;exten => 19001,2,Answer exten => 19001,2,Wait(2) exten => 19001,3,CBMysql() exten => 19002,1,Playback(conf-hasentered) ;exten => 19002,,Answer exten => 19002,2,Wait(2) exten => 19002,3,MeetMe(1234|pm) exten => 19003,1,Playback(conf-hasentered) ;exten => 19003,2,Answer exten => 19003,2,Wait(2) exten => 19003,3,MeetMe(1234|pm) ------------------------------------------------------------------------ * In cbmysql.conf* [global] hostname=localhost dbname=conference password= user=root port=3306 sock=/var/lib/mysql/mysql.sock DBOpts=yes ;OptsAdm=asdp ;OptsUsr=sdp ConfApp=MeetMe ConfAppCount=MeetMeCount ; Choose one of the following to modify early join behaviour earlyalert=300 ; Tell the participant if they are too early (seconds) ;fuzzystart= ; Allow participants to join early (seconds) ------------------------------------------------------------------------ * In sip.conf *[19001] type=friend username=9001 secret=august context=from-sip host=192.168.2.75 fromdomain=192.168.2.76 realm=192.168.2.75 ;mailbox=9001@from-sip insecure=very callerid="Ravi" <9001009> disallow=all allow=ulaw allow=gsm nat=yes ------------------------------------------------------------------------ 9001,9002,and 9003 is register from openSER, When They dial 19001 to enter the conference. Following are Showing the Errors and Warning .... *Executing Playback("SIP/9001-08f8d7e0", "conf-hasentered") in new stack -- Playing 'conf-hasentered' (language 'en') -- Executing Wait("SIP/9001-08f8d7e0", "2") in new stack -- Executing CBMySQL("SIP/9001-08f8d7e0", "") in new stack -- Playing 'conf-getconfno' (language 'en') Oct 25 18:15:47 NOTICE[12281]: app_cbmysql.c:373 cb_exec: getConf: 1 -- Playing 'agent-pass' (language 'en') Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags: Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:146 passQuery: CBMySQL: Invalid room or pass Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY: -- Playing 'auth-incorrect' (language 'en') Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags: Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:146 passQuery: CBMySQL: Invalid room or pass Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY: -- Playing 'auth-incorrect' (language 'en') Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags: Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY: Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:376 cb_exec: getPass: 1 == Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:97 checkMax: Currentusers: 0 Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:390 cb_exec: checkMax: 1 Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:276 enterConf: Roomtype: 1234|| == Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 WARNING[12281]: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Oct 25 18:16:13 ERROR[12281]: chan_zap.c:7396 chandup: Unable to dup channel: No such file or directory Oct 25 18:16:13 WARNING[12281]: app_meetme.c:460 build_conf: Unable to open pseudo channel - trying device Oct 25 18:16:13 WARNING[12281]: app_meetme.c:463 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') Oct 25 18:16:17 NOTICE[12281]: app_cbmysql.c:393 cb_exec: enterConf: -1 == Spawn extension (from-sip, 19001, 3) exited non-zero on 'SIP/9001-08f8d7e0' * Please Help me..... -- Thanks & Regards, *Ravi Prakash Sunkara* *M*:+91 9985077535* O*:+91 40 23114549* F*:+91 40 40208727 /*ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com */

2 1

[Users] Changing call ID?
by Olle E Johansson 26 Oct '06

26 Oct '06

I got a bug report in our bug report tracker for Asterisk that claims that SER changes the call ID by adding a :port after the IP address. From reading the RFC, I don't think it's a valid change. For me, it's a new call ID. Questions: - Is changing a Call ID permitted within a dialog? - Does SER/OpenSER actually do this? http://bugs.digium.com/view.php?id=8220 See you at the Summit in Berlin! regards, /Olle

2 1

[Users] Cvs snapshot Invalid table version
by Pletli Antal 26 Oct '06

26 Oct '06

Hi, How can I fix the "version" table to work with openser cvs snapshot? I have problem with grp,silo and location tables. I have installed the DB from the cvs too. +-----------------+---------------+ | table_name | table_version | +-----------------+---------------+ | subscriber | 5 | | missed_calls | 3 | | location | 1003 | | aliases | 1003 | | grp | 2 | | re_grp | 1 | | acc | 3 | | silo | 5 | | domain | 1 | | uri | 1 | | trusted | 3 | | usr_preferences | 2 | | speed_dial | 2 | | dbaliases | 1 | | gw | 3 | | gw_grp | 1 | | lcr | 2 | | sip_trace | 1 | | address | 2 | | domainpolicy | 2 | | presentity | 1 | | presence | 1 | +-----------------+---------------+ Antal

2 1

[Users] Registration of Polycom SoundPointIP phone with OpenSER
by jeevan ravula 26 Oct '06

26 Oct '06

Hi all, I am using Polycom SoundPointIP phone as User Agent.I want to register Polycom phone with OpenSER(with TLS support) server.Can anybody help me out in this regard? I have generated my rootCA and given to polycom phone.The polycom phone does not accept certificate from openser server side.It shows bad certificate. anybody who has used polycom phone earlier can help me out in this matter.Ishall be greatful to them Regards, Jeevan.

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Re: [Serusers] Re: How to change Caller ID in FROM field ?
by lawrence k.y. lin 26 Oct '06

26 Oct '06

Hi there, This is an update for the Caller ID with Telco Carrier. The reason that Caller ID did not work on Verizon's mobile phone is because I only change the Display Name in the From Field. After I changed the From-URI (the field after the Displan Name which is <sip: XXXXXXXXXX@ip>), then Caller ID is displayed on Verizons mobile phone without any problem. Thanks for all that replied to this question. Best Regards, Larry >lawrence k.y. lin wrote: >Steve, > > Thanks for your email and your solutions. > > I tried the first method as follws: > >replace ("^From:(.*)<", "From: \"XXXXXXXXXX\" <") > > before I sent it to another SIP device and PSTN gateway. I did see the >new Caller ID XXXXXXXXXX in another SIP device but the Caller ID to Verison >Mobile phone showed "Unavailable No". I will check with Telco carrier to >see if we did send the Caller ID to them or not. > Also I will try your second method and update to the list later. > > Thanks again. > >Larry > >>From: Steve Blair <blairs(a)isc.upenn.edu> >>To: "lawrence k.y. lin" <lawlin888(a)hotmail.com> >>CC: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Re: How to change Caller ID in FROM field ? >>Date: Tue, 24 Oct 2006 18:57:42 -0400 >> >> >>You can do this several ways. I use to just completely replace the From >>header but this screwed up call accounting (for callers with caller ID >>blocking enabled) which we do in our Cisco gateways. Now I append an >>Remote-Party-ID header leaving the Fromheader intact for accounting >>purposes. Try one of the following: >> >>1) replace("^From:(.*)>" , "From: \"Anonymous\" >><sip:anonymous@invalid.nowhere>"); >> >> Where the second string contains whatever you want the caller ID info >>to be. This example is for caller ID blocking. >> >>2) >> >>if (append_rpid_hf("Anonymous >><sip:","@upenn.edu>;party=calling;id-type=subscriber;screen=yes")) { >>} else { >> log(1, "RPID not found\n"); >>}; >> >>Of course this requires other configuration items. You may also want to >>add the following: >> >>modparam("auth", "rpid_prefix", "<sip:") >>modparam("auth", "rpid_suffix", >>"@upenn.edu;user=phone>;party=calling;screen=no;id-type=subscriber;privacy=off") >> >># >> >>-Steve >> >>lawrence k.y. lin wrote: >> >>>Hi all, >>> >>> Appreciate if someone can give me a suggestion or even an answer. >>> Since the call from SIP device to PSTN phone did not have Caller ID >>>and our Telco carriers request for the Caller ID, we have to set a number >>>before the calls are routed to PSTN gateway. >>> I did it in Asterisk before with the function calls SetCallerID(CLID) >>>or Set(CALLERID(number)=CLID) (depends on the Asterisk version) and these >>>function calls worked. But because of the scalabliity issue, I changed to >>>SER and would like to do the similar thing as in Asterisk. >>> I read the SER document and awared SIP_HF_FROM has the data for the >>>entired FROM field. Is there any other variables for Caller ID only ? >>> I searched the mailing list for SER and did not find any related >>>topic (only found one with title: caller-id with raius using sip-rpid, >>>but it is irrelevant). >>> >>> Thanks in advance. >>> >>>Larry _________________________________________________________________ Try the next generation of search with Windows Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&source=…

1 0

[Users] Re: Conference is Not Working.... with OpenSER And, Asterisk
by Paul Smith 25 Oct '06

25 Oct '06

This is an Asterisk config problem rather than an OpenSer one. The conference applications in Asterisk need a timing source. If you install a Digium or Sangoma card then Asterisk will use it as a timing source. As you are not using such a card you need to compile and modprobe a module called ztdummy to create a pseudo timing device.

1 0

[Serusers] How do I allow gateway on ser to not ask for any authentication
by Lokesh Kumar 25 Oct '06

25 Oct '06

Hello , How do I configure my ser and radius so that it won’t ask for credentials from PSTN gateway, Please help Thanks very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 - Release Date: 10/24/2006

2 1

[Serusers] Re: Asterisk Manager
by Maps 25 Oct '06

25 Oct '06

Dear Friends and Supporters! I try to write a php application to monitor the asterisk, but when I open the .php to access to asterisk to retrieve the information about the queues status, sip show peers, realtime mysql status etc... However, It just return to me "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and start asterisk with user is asterisk. HTTPD is run under asterisk user. My manager.conf has an entry. [admin] secret = password deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user However, my php still unable to retrieve the information for asterisk. Did I miss somethings? Your help would be very appreciated! Regards, Lan

2 1

[Serusers] MySQL has gone away
by Mark Price 25 Oct '06

25 Oct '06

Hi, I'm getting the MySQL has gone away error. I have set the following in ser.cfg: modparam("mysql", "auto_reconnect", 1) modparam("mysql", "ping_interval", 60) Restart fixes this, but that is of course not sufficient. I suppose I can detect every time that error happens with swatch or some such thing and restart the service. This seems to be a common error. Is there a common solution? Mark

1 0

[Serusers] MySQL confusion
by Chris Parkinson 25 Oct '06

25 Oct '06

Hello, I'm a complete newbie to SER and I'm having a problem following the MySQL section of the "getting started" guide. Whenever I try to use serctl to add a user I get access denied; LabServ1:/home/chris# export SIP_DOMAIN=mylab.com LabServ1:/home/chris# serctl add test1 test1pass test1(a)mylab.com MySql password: ERROR 1045: Access denied for user: 'serro@localhost' (Using password: YES) error: SER/FIFO not accessible: 2 LabServ1:/home/chris# I think this may be down to the MySQL install. I've installed MySQL and tested it, however I can't find any documentation anywhere to show me what users, tables and databases to setup for SER. I'm using SER on Debian. It's quite possible I'm missing something obvious. I'd appreciate a pointer in the right direction. Chris

3 5

[Users] X-lite is not Working, But SipEyeMedia Softphones is Working Fine.
by sunkara 25 Oct '06

25 Oct '06

Hello Users, I don't Know What Happening in this Issues... Help me in this . When Testing the X-lite Softphones, is not working. When I use the Sip EyeMedia Softphone it Working Fine. Is Problem in X-lite Configuration or Router(Firewall Gateway Keepers ) is blocking it... Help me . -- Thanks & Regards, *Ravi Prakash Sunkara* *M*:+91 9985077535* O*:+91 40 23114549* F*:+91 40 40208727 /*ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com */

2 1

[Serusers] Using different domains
by Nick Hoffman 25 Oct '06

25 Oct '06

Hi guys. I just noticed that when user agents register, the domain in their SIP URI does not have to equal their domain in the subscribers database table. For instance, I have 2 domains in the domain table: sip.altcall.com sip.voxpak.com The username "foobar" in my subscribers table has the domain "sip.altcall.com", but can register and make calls as foobar(a)sip.voxpak.com . Why is this? Cheers, -- Nick E: nick.hoffman(a)voxpak.com P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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[Serusers] Re: SER "presence snapshots": "id" value in <tuple>
by Vaclav Kubart 25 Oct '06

25 Oct '06

Hi, SER always generates new tuple IDs for newly published tuples. This is because it can collect information from more sources - more clients can send PUBLISH with the same tuple IDs, but according to RFC 3903 tuples can rewrite each other only if they were published with the same entity tag (SIP-ETag header field). If you don't want to use tuples created from registration status you can set PA module parameter "use_callbacks" to 0. Or do you need for clients which published something not to use tuples generated from registration status and for those which doesn't publish anything to use such tuples? This is not possible right now, but it is not too hard to implement it... Vaclav On Tue, Oct 24, 2006 at 06:00:03PM +0200, Diego Do?ate wrote: > > Hi, > > I am using the "presence snapshot" version of the SER server, and I want > to use a presence service. My problem is related to the "id" parameter > within the <tuple> (in the NOTIFY xml). The test is like that: > > User A subscribes to the presence of user B : receives a NOTIFY with 1 > <tuple> (id="none", <basic> is closed) > User B registers in SER: user A receives a NOTIFY with 1 <tuple> > (id="string", <basic> is open) > User B publishes a new presence state: user A receives a NOTIFY with 2 > <tuple> (the 2nd is the same as the previous one, and a new one is attached > first with different "id" string value. > > I would like only 1 <tuple>, since the device is the same, is it my > problem (something in the PUBLISH), or is it in the SER? How can I know this > "id"? Whatever I put in the PUBLISH <tuple> "id", the SER puts another > string in the NOTIFY created... > > Thaks in advance > > Diego Do?ate > > > > >

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[Serusers] Re: pa module: presence "substatus"
by Vaclav Kubart 25 Oct '06

25 Oct '06

Hi, doesn't SER send "202 Accepted" instead of "200 OK" before "pending NOTIFY"? You can set PA module parameter "auth" to "none" as described in presence handbook. Registrations are written into database (table "location") only if you set db_mode parameter of userloc module to nonzero value. Vaclav On Tue, Oct 24, 2006 at 09:40:30AM +0200, Diego Do?ate wrote: > > Hi, > > At the end I have running my SER, the "presence snapshots" version... > Thanks very much for your help. > > I have a doubt with the presence service: when user A subscribes to the > presence of user B (sends a SUBSCRIBE request), the SER sends the 200 Ok and > the initial NOTIFY ("pending"). I don't know how user B can authorize the > subscription (SER sends nothing to user B), or if it is possible not to > require authorization, that is, that SER sends automatically the 2nd NOTIFY > (status, not "pending")? > > Related to that, when a user is registered, where can I find/delete the > registration? "Location" table is empty... Maybe the problem with the > presence is related to that, since there is no "subscriber" table, where are > created users? In fact, since I had to uninstall and re-install SER, maybe > the users don't exists yet, is it possible? Although I can register them > (and SER accepts PUBLISH and SUBSCRIBE requests...) > > Thanks a lot in advance > > > > -----Mensaje original----- > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] > Enviado el: lunes, 23 de octubre de 2006 7:51 > Para: Diego Do?ate > CC: serusers(a)iptel.org > Asunto: Re: pa module: presence "substatus" > > Hi, > you have to set LD_LIBRARY_PATH variable before running ser (see > "3.4 Running SER" in presence handbook): > > export LD_LIBRARY_PATH=/base/ser/directory/lib/ser > > /base/ser/directory/sbin/ser -f /base/ser/directory/etc/ser/ser.cfg > > Vaclav > > On Fri, Oct 20, 2006 at 07:06:54PM +0200, Diego Do?ate wrote: > > > > Hi, > > > > I have done followed these steps, but still the same errors. > > Should I unistall something else? > > ... > > 0(13168) qm_malloc(0x8113ea0, 24) returns address 0x8148710 frag. > > 0x81486f8 > > (size=24) on 1 -th hit > > 0(13168) read 2739142786 from /dev/urandom > > 0(13168) seeding PRNG with 3900519771 > > 0(13168) test random number 509866974 > > 0(13168) ERROR: load_module: could not open module > > </usr/local//lib/ser/modules/dialog.so>: lib_ser_cds.so: cannot open > shared > > obj ect file: No such file or directory > > 0(13168) parse error (36,13-14): failed to load module > > 0(13168) ERROR: load_module: could not open module > > </usr/local//lib/ser/modules/rls.so>: lib_ser_xcap.so: cannot open shared > > objec t file: No such file or directory > > 0(13168) parse error (37,13-14): failed to load module > > 0(13168) ERROR: load_module: could not open module > > </usr/local//lib/ser/modules/pa.so>: lib_ser_xcap.so: cannot open shared > > object file: No such file or directory > > 0(13168) parse error (38,13-14): failed to load module > > 0(13168) ERROR: load_module: could not open module > > </usr/local//lib/ser/modules/presence_b2b.so>: lib_ser_xcap.so: cannot > open > > sha red object file: No such file or directory > > 0(13168) parse error (39,13-14): failed to load module > > 0(13168) set_mod_param_regex: No module matching <rls> found > > 0(13168) parse error (76,21-22): Can't set module parameter > > 0(13168) set_mod_param_regex: No module matching <rls> found 0 > > .... > > > > > > Diego > > > > -----Mensaje original----- > > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Enviado el: > > viernes, 20 de octubre de 2006 7:15 > > Para: Diego Do?ate > > CC: serdev-bounces(a)lists.iptel.org > > Asunto: Re: pa module: presence "substatus" > > > > Hi, > > this seems that you are compiling something else than "presence snapshot" > > because "presence snapshot" has modified Makefiles to compile > > libraries automaticaly which was not done in your case. > > > > Try to follow the compilation rules "3.2 Installation from CVS": > > > > cd sip_router/lib > > make -f Makefile.ser install prefix=/base/ser/directory cd .. > > make install include_modules="dialog pa rls presence_b2b xcap mysql > xmlrpc" > > prefix=/base/ser/directory > > > > [in your case will be prefix=/usr/local if you want to install ser > > there] > > > > Vaclav > > > > On Thu, Oct 19, 2006 at 07:57:55PM +0200, Diego Do?ate wrote: > > > > > > Hi, > > > > > > Thanks for the suggestion. I have download the 0.10.99 version > > > ("presence > > > snapshot") but I can not make it work: after the "make install", I > > > go to /usr/local/etc/ser, where is the ser.cfg to run, but with ser > > > -f ser.cfg it gives me this errors: > > > > > > ERROR: bad config file (47 errors) > > > -bash-3.00# ser -f ser.cfg -E > > > 0(15125) ERROR: load_module: could not open module > > > </usr/local/lib/ser/modules/dialog.so>: lib_ser_cds.so: cannot open > > > shared object file: No such file or directory > > > 0(15125) parse error (36,13-14): failed to load module > > > 0(15125) ERROR: load_module: could not open module > > > </usr/local/lib/ser/modules/rls.so>: lib_ser_xcap.so: cannot open > > > shared object file: No such file or directory > > > 0(15125) parse error (37,13-14): failed to load module > > > 0(15125) ERROR: load_module: could not open module > > > </usr/local/lib/ser/modules/pa.so>: lib_ser_xcap.so: cannot open > > > shared object file: No such file or directory > > > 0(15125) parse error (38,13-14): failed to load module > > > 0(15125) ERROR: load_module: could not open module > > > </usr/local/lib/ser/modules/presence_b2b.so>: lib_ser_xcap.so: > > > cannot open shared object file: No such file or directory > > > 0(15125) parse error (39,13-14): failed to load module > > > 0(15125) set_mod_param_regex: No module matching <rls> found > > > > > > > > > And the last line is repeated with the 4 modules it can not find. > > > But the 4 of them exists in /usr/local/lib/ser/modules, where it is > > > looking > > for... > > > > > > I thought the installation was going to be easy, as read in > > > http://ftp.iptel.org/pub/ser/presence/presence-handbook/ar01s03.html > (3.3. > > > Installation from presence-snapshot), but there is something I am > > > doing wrong. > > > > > > Any suggestion? I attach the .cfg > > > > > > Thanks a lot > > > > > > Diego Do?ate > > > > > > > > > -----Mensaje original----- > > > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Enviado el: > > > miércoles, 18 de octubre de 2006 11:22 > > > Para: Diego Do?ate > > > CC: serdev-bounces(a)lists.iptel.org > > > Asunto: Re: pa module: presence "substatus" > > > > > > Hi, > > > I'm afraid that there was not good support for presence in 0.9.6. I > > > recommend to use "presence snapshot". > > > Vaclav > > > > > > On Wed, Oct 18, 2006 at 10:37:27AM +0200, Diego Do?ate wrote: > > > > > > > > Hi, > > > > > > > > I am using the last version of the SER server (0.9.6), and > > > > have a problem with the presence management. I want a user (805) > > > > to subscribe to the presence of another user (806), but when 805 > > > > sends a SUBSCRIBE to the SER, it responses a NOTIFY with > > > > "closed-offline", while both users had sent previously a successful > REGISTER to the SER. > > > > > > > > Both users are added (serctl) to the server, and are displayed > > > > in "subscriber" table of the "ser" DB. Once registered, they are > > > > also in the "location" table. But the ser does not add "open" to > > > > the NOTIFY.. Any > > > idea? > > > > > > > > Once the user 805 is successfully subscribed to the presence > > > > of user 806, when 806 modifies its presence (sends a PUBLISH to > > > > the SER, with > > > "open" > > > > and <note>"Away"</note> in the xml, for instance), the SER is > > > > suppposed to send a NOTIFY with the same xml to user 805 > > > > (subscriber), > > > isn't it? > > > > > > > > "pa" module is added, but in this moment I don't need > > > > authentication. U can see the ser.cfg file bellow: > > > > > > > > Thanks in advance > > > > > > > > > > > > // ******************************** # # $Id: ser.cfg,v 1.25.2.1 > > > > 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config > > > > script # > > > > > > > > # ----------- global configuration parameters > > > > ------------------------ > > > > > > > > #debug=3 # debug level (cmd line: -dddddddddd) > > > > #fork=yes > > > > #log_stderror=no # (cmd line: -E) > > > > > > > > /* Uncomment these lines to enter debugging mode 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" > > > > > > > > loadmodule "/usr/local/lib/ser/modules/pa.so" > > > > > > > > # Uncomment this if you want digest authentication # mysql.so must > > > > be loaded ! > > > > #loadmodule "/usr/local/lib/ser/modules/auth.so" > > > > #loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > > > > > > > # ----------------- setting module-specific parameters > > > > --------------- > > > > > > > > # -- usrloc params -- > > > > > > > > #modparam("usrloc", "db_mode", 0) > > > > > > > > # Uncomment this if you want to use SQL database # for persistent > > > > storage and comment the previous line modparam("usrloc", > > > > "db_mode", > > > > 2) > > > > modparam("pa","db_url","mysql://ser:heslo@localhost/ser") > > > > > > > > # -- auth params -- > > > > # Uncomment if you are using auth module # #modparam("auth_db", > > > > "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to > > > > yes (which true in this config), # uncomment also the following > > > > parameter) # #modparam("auth_db", "password_column", "password") > > > > > > > > # -- rr params -- > > > > # add value to ;lr param to make some broken UAs happy > > > > #modparam("rr", "enable_full_lr", 1) > > > > > > > > # ------------------------- 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=="PUBLISH"){ > > > > if(t_newtran()){ > > > > handle_publish("registrar"); > > > > break; > > > > } > > > > } > > > > if (method=="SUBSCRIBE") { > > > > if (!t_newtran()) { > > > > log(1, "newtran error\n"); > > > > sl_reply_error(); > > > > }; > > > > handle_subscription("registrar"); > > > > break; > > > > }; > > > > > > > > if (method=="REGISTER") { > > > > > > > > # Uncomment this if you want to use digest authentication > > > > # if (!www_authorize("192.168.1.118", "subscriber")) { > > > > # www_challenge("192.168.1.118", "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(); > > > > }; > > > > } > > > > > > > > > > > > > > > > > > > > > > > > // ******************************** > > > > > > > > -----Mensaje original----- > > > > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Enviado el: > > > > miércoles, 18 de octubre de 2006 6:21 > > > > Para: Diego Do?ate > > > > CC: serusers(a)iptel.org > > > > Asunto: Re: [Serdev] pa module: presence "substatus" > > > > > > > > Hi, > > > > notes are quite often used for this. > > > > > > > > Can you send your config here? What SER version do you use? And if > > > > you try register your users via sending REGISTER messages it is > > > > still the > > > same? > > > > > > > > Vaclav > > > > > > > > P.S. Send these messages to serusers too, might be that somebody > > > > else will have the same problem... > > > > > > > > On Tue, Oct 17, 2006 at 05:18:55PM +0200, Diego Do?ate wrote: > > > > > > > > > > Hi, > > > > > > > > > > Thanks, then I will use a <note> tag for the presence > > > > > substatus(<note>Away<\note>, for instance). > > > > > > > > > > I have a problem with the Subscription. User A subscribes to > > > > > the presence of user B and receives a NOTIFY but with > > > > > "close/offline", and both users are previously registered > > > > > successfully > > in SER. > > > > > > > > > > Both users are added with serctl, but it seems the SER does > > > > > not > > > "save" > > > > > user A registration when user B subscribes... Where may be the > > > > > problem? May it be related to autherization? By now, it is > disabled... > > > > > > > > > > Thanks in advance > > > > > > > > > > > > > > > > > > > > -----Mensaje original----- > > > > > De: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Enviado el: > > > > > lunes, > > > > > 16 de octubre de 2006 18:57 > > > > > Para: Diego Do?ate > > > > > CC: serdev(a)lists.iptel.org > > > > > Asunto: Re: [Serdev] pa module: presence "substatus" > > > > > > > > > > Hi, > > > > > support for PIDF extensions (elements from non-pidf namespace) > > > > > is nearly finished in current CVS version, but it still needs some > work. > > > > > It will be done as soon as possible. > > > > > > > > > > Notes should be possible in last presence snapshot without any > > > > > problems > > > > > - they can be within tuple or presence elements according > > specification. > > > > > > > > > > Vaclav > > > > > > > > > > On Mon, Oct 16, 2006 at 05:52:47PM +0200, Diego Do?ate wrote: > > > > > > > > > > > > Hi, > > > > > > > > > > > > In the SER "pa" module, I would like to use a tag like "note" > > > > > > or "substatus" in the pidf of the NOTIFY when a PUBLISH is > > > > > > received (with a presence change, to "Away", for instance), > > > > > > but SER only modified the <status> tag ("open"/"closed"). > > > > > > > > > > > > How can I implement a more detailed presence status > > > > > > management in the SER? > > > > > > > > > > > > Thanks in advance > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > Serdev mailing list > > > > > > Serdev(a)lists.iptel.org > > > > > > http://lists.iptel.org/mailman/listinfo/serdev > > > > > > > > > > > > > > > >

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[Serusers] Newbie
by Eddie Johnson Jr 24 Oct '06

24 Oct '06

Jan 11 16:58:01 NOTICE[3987]: chan_sip.c:5387 sip_reg_timeout: -- Registration for '1000(a)serserv.com' timed out, trying again (Attempt #20000) Jan 11 16:58:01 WARNING[3987]: chan_sip.c:1989 create_addr: No such host: serserv.com Jan 11 16:58:01 WARNING[3987]: chan_sip.c:5470 transmit_register: Probably a DNS error for registration to 1000(a)serserv.com, trying REGISTER again (after 20 sec onds) Looked on google and nothing helps with navigating me to the correct answer. Any suggestion? Ed

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[Users] [ANNOUNCE] new domainpolicy module
by Klaus Darilion 24 Oct '06

24 Oct '06

Hi! In openser-CVS there is a new module called domainpolicy. It implements draft-lendl-domain-policy-ddds-02. This version supports draft-lendl-speermint-federations-02 and draft-lendl-speermint-technical-policy-00. This is about federation based and static peering (for SIP providers which do not offer public SIP peering). For background of this please refer to the SPEERMINT working group: http://www.ietf.org/html.charters/speermint-charter.html Attached are 3 documents: one about building openser+domainpolicy one about configuration and one is a small howto about federations and domainpolicies. You can also find a quick summary of domainpolicy in this mail: http://www1.ietf.org/mail-archive/web/speermint/current/msg01598.html Currently, openser CVS does not support interims solutions for infrastructure ENUM lookups - the patch for this in in the queue. Meanwhile you can download it from enum.at (see above link) too. regards Klaus PS: If you do not want to use openser-CVS you can find the module or complete tar balls for openser 1.1 at http://www.enum.at/index.php?id=dokumente -- Klaus Darilion nic.at This files describes how to install the openser for the Austrian IENUM Trial. openser-ienum changelog: Version 0.4: - based on openser-1.1.0 from 2006-10-02 - domainpolicy module 0.4 + sql patch - aligns with draft-lendl-domain-policy-ddds-02 - a few updates and bug-fixes - supports draft-lendl-speermint-federations-02 and draft-lendl-speermint-technical-policy-00 - handling of non-terminal NAPTR records, as well as the DB schema has changed. Version 0.3: - based on openser-1.1.0 from 2006-07-14 Version 0.2: - based on openser-CVS head from 2006-05-22 - tls patch not needed anymore as applied to CVS head - domainpolicy module 0.2 + sql patch - supports integer name based AVPs in "attr" column of domainpolicy table - supports forcing a send_socket via dp_apply_policy - bug fixed (NAPTR lookup need \0 terminated string) Version 0.1: initial release - based on openser-CVS head from 2006-05-17 - ienum patch - tls patch - domainpolicy module 0.1 + sql patch Compared to openser CVS, this openser has the following differences: - ienum patch to allow ienum queries - domainpolicy modul to allow ad-hoc peering This file decribes how to build your IENUM-openser. 1. You have to get the sources. There a 2 choices. You can either download the patched openser-tarball from www.enum.at or you patch it yourself. 1.1. Download the patched tarball: Go to enum.at, download the patched tarball (currently version 0.4). Untar it using the command "tar -xvjf openser-ienum-0.4.tgz" cd sip-server 1.2 Checkout openser from sourceforge's CVS and apply the patches manually. (Patches were tested to apply to openser-1.1.0 from 2006-10-02). checkout openser-cvs cvs -d:pserver:anonymous@openser.cvs.sourceforge.net:/cvsroot/openser login cvs -z3 -d:pserver:anonymous@openser.cvs.sourceforge.net:/cvsroot/openser co -r rel_1_1_0 -P sip-server download ienum patch wget -O ienum-patch.txt "http://sourceforge.net/tracker/download.php?group_id=139143&atid=743022&fil…" download domainpolicy modul und SQL patch from enum.at homepage openser-domainpolicy_0.4.tgz openser-domainpolicy-sql-patch_0.3.txt apply patches cd sip-server patch -p0 <../openser-domainpolicy-sql-patch_0.3.txt cd modules tar -xvzf ../../domainpolicy_0.4.tgz cd .. patch -p0 <../ienum-patch.txt 2. Building the ienum-openser When building openser, you have to install some development libraries, e.g. libssl-dev for TLS and mysql or postgresql development libraries. Further you have to install the libradius-ng-dev package. This must be built from the sources of radiusclient-ng available from berlios. Depending on your Linux Distribution there are different choices to build: 2.1. Debian The easiest way is build packages: make deb This will build TLS enabled openser packages. As this will build all available modules, you need several development libraries installed on the system. E.g. the libmysql-dev, libpg-dev and the radiusclient-ng devel libraries. As the radiusclient-ng is not part of debian, you have to install it manually as described on the openser wiki: http://www.openser.org/dokuwiki/doku.php?id=howto_make_debian_packages Depending on your setup, and the used DB (mysql or postgresql) install the built debian packages. E.g.: NOTE: this step replaces existing installed openser packages. Take care as openser-1.1 syntax is not 100% compatible with openser-1.0 syntax. cd .. su dpkg -i openser_1.1.0-0_i386.deb openser-mysql-module_1.1.0-0_i386.deb Set up the database tables: NOTE: edit the script /usr/sbin/openser_mysql.sh (or set environment variables, e.g. "export DBNAME=ienum_borderproxy") to reflect your DB environment (e.g. to do not overwrite existing openser databases). openser_mysql.sh create Now you can edit the config in /etc/openser/openser.cfg and use the start/stop script /etc/init.d/openser. For debugging watch syslog logging in /var/log/syslog 2.2. Others You can install openser manually. First you have to modify the Makefile to remove the needed modules from "exclude_modules", e.g. mysql or postgres. Then build the usual way: NOTE: this step replaces existing manually installed openser. Take care as openser-1.1 syntax is not 100% compatible with openser-1.0 syntax. export TLS=1 make all su make install This will install openser in /usr/local/sbin/openser and /usr/local/lib/openser/modules/. The configuration is stored in /usr/local/etc/openser/openser.cfg. There is no startup script installed, thus you have to copy it manually from etc/openser.init to /etc/init.d/openser and adopt it to your paths. 3. Config If you are used to ser or openser-1.0.x, please the migration info on http://openser.org/dokuwiki/doku.php?id=migrating_openser_v1.0.x_to_v1.1.x Configuration for the ienum openser 1. Introduction Using Infrastructure ENUM and SIP VoIP peering requires 2 steps: 1. perform IENUM lookups 2. interconnect with the destination published in the IENUM NAPTRs Typically, a VoIP setup has a central node which performs the routing logic. I will call this the "main proxy". We can add the previously described steps in the main proxy or use dedicated nodes for doing the IENUM lookup and session peering. This has many advantages regarding security, flexibility and reliability. We suggest a setup of a dedicated "border proxy" which performs the IENUM lookup and takes care of peering (e.g. authentication of peering partners). This border proxy is a logical node, that means it can be installed on the same PC as the main proxy. Nevertheless doing the IENUM lookup and the peering in the main proxy. If you want to do this, please use the ideas and config snippets from our suggested scenario (a dedicated border proxy) to adopt your main proxy's routing logic. 2. The Border Proxy The border proxy takes care of the IENUM lookup and the SIP peering. The border proxy will perform the following steps: For outgoing calls: - bring number into E.164 format - perform IENUM lookup - perform ENUM lookup (optional) - if no destination is found in IENUM/ENUM, send an apropriate SIP response back to the main proxy. The main proxy will then perform the default routing (usually sends the call to the PSTN gateway) - perform a domain policy lookup. This step detects if the destination found in IENUM will accept SIP calls from us - apply destination domain's peering policy (e.g. set proper TLS parameters, send to certain port, ...) - add P-Asserted-Idendity: Header and Privacy Header the send proper CLI and CLIP/CLIR - activate accounting - send the call to the destination For incoming calls: - receive call - authenticate sender (e.g. TLS or IP address) - activate accounting - adjust CLI to fit local dialplan (e.g reformat P-Asserted-Idendity header) - forward the call to the main proxy 3. Configuration 3.1. Main Proxy The main proxy should forward the call to the border proxy for IENUM lookup and peering. If the main proxy receives a certain response code, it will failover to the standard route to the PSTN. E.g. if the main proxy is (open)ser based, the respective config snippets will be: route{ # all the authentication and NAT traversal stuff ... # user is authenticated. check if destination is an phone number if (uri =~ "^sip:(\+|\*)?[0-9]+@") { # PSTN number detected if (uri=~"^sip:[1-9]") { xlog("L_WARN", "number normalization: illegal number, 404..."); sl_send_reply("404", "Not found - illegal number"); exit; } if (uri=~"^sip:0[1-9]") { xlog("L_INFO","Austria national number detected, remove 0, prefix +43\n"); strip(1); prefix("+43"); } else if (uri=~"^sip:00[1-9]") { xlog("L_INFO","international number detected, remove 00, prefix +\n"); strip(2); prefix("+"); } else if (uri=~"^sip:000") { xlog("L_INFO", "000 detected, illegal number, 404..."); sl_send_reply("404", "Not found - illegal number"); exit; } } # usually calls sent to the gateway have an Remote-Party-ID header # to signal the CLI to the gateway. We send this RPID header also to the # border proxy # create new AVP with the Remote-Party-ID header avp_printf("$avp(s:temp)","$avp(s:rpid);party=calling;id-type=subscriber;screen=yes"); append_hf("Remote-Party-ID: $avp(s:temp)\r\n"); # now, usually we would send the call to the PSTN gateway. But instead # we activate a failure route and send it to the border proxy t_on_failure("1"); # debpending on your domain setup you may need to rewrite the domain # to address the border proxy # rewritehostport("bp1.istp1.com"); # send the request to the border proxy if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { # handles error on border proxy xlog("L_INFO","failure_route(1): reply status = $rs\n"); # if caller cancels the call we will also enter the failure route if(t_was_cancelled()) { xlog("L_INFO","caller cancelled call ...exit\n"); exit; } # border proxy will return "499" in case of unsuccessful IENUM/domainpolicy lookup if(t_check_status("499")) { xlog("L_INFO","callee neither in I-ENUM nor in U-ENUM...sending to PSTN Gateway..."); rewritehostport("gateway.istp1.com"); t_relay(); exit; } xlog("L_INFO","send down error code from border proxy to user...exit\n"); exit; } If the main proxy is Asterisk based, the respective snippet may look like this (untested): [frompbx] ; border proxy will send "503" if no IENUM result found ; in this example the Asterisk is also the gateway. If not you ; have to adopt the second Dial parameter to send the call to your ; gateway exten => _[1-9].,1,Dial(SIP/+431${EXTEN}(a)bp1.itsp1.com,90) exten => _[1-9].,2,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?103:3) exten => _[1-9].,3,Hangup exten => _[1-9].,103,Dial(ZAP/g1/${EXTEN},90) exten => _0[1-9].,1,Dial(SIP/+43${EXTEN:1}@bp1.itsp1.com,90) exten => _0[1-9].,2,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?103:3) exten => _0[1-9].,3,Hangup exten => _0[1-9].,103,Dial(ZAP/g1/${EXTEN},90) exten => _00[1-9].,1,Dial(SIP/+${EXTEN:2}@bp1.itsp1.com,90) exten => _00[1-9].,2,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?103:3) exten => _00[1-9].,3,Hangup exten => _00[1-9].,103,Dial(ZAP/g1/${EXTEN},90) 3.2 Border Proxy The border proxy performs the IENUM/ENUM lookups and the domainpolicy lookup. Further it may perform authentication for peering purposes. Again some config snippets for openser (you need the patched ienum-openser from www.enum.at) ...config section ...module parameter section loadmodule "/usr/lib/openser/modules/enum.so" loadmodule "/usr/lib/openser/modules/domainpolicy.so" loadmodule "/usr/lib/openser/modules/tlsops.so" # carrier enum modparam("enum", "i_enum_suffix", "e164.arpa.") modparam("enum", "bl_algorithm", "c") # domainpolicy module (this must be named like in the domainpolicy table) modparam("domainpolicy", "port_override_avp", "portoverride") modparam("domainpolicy", "transport_override_avp", "transportoverride") modparam("domainpolicy", "domain_prefix_avp", "domainprefix") modparam("domainpolicy", "domain_suffix_avp", "domainsuffix") modparam("domainpolicy", "send_socket_avp", "sendsocket") route{ ..... # now, we split routing depending on the direction # the protocol can be used to identify the direction # if we require TCP or TLS for peering, but use UDP inside # we can use the protocol to detect the call direction # tls: request from peering partner # udp,tcp: local request if( (proto==tls) or (proto==tcp) ){ route(2); } else { route(3); }; } route[2] { # TLS requests - thus requests received from peering partners. # This requests are already authenticated via TLS certificate # validation, thus no further SIP based authentication needed. # Just forward the requests to the main proxy. # Beforehand, we map from the anonymous IENUM SIP URIs to the # user's AoR using the dbaliases module. Note: this mapping can # also be done in the main proxy. xlog("L_INFO","[$Tf] $rm $ru (From: $fu -> To: $tu) entering route(2): TLS requests\n"); if (alias_db_lookup("dbaliases")) { xlog("L_INFO","dbaliases lookup successful, new URI = $ru\n"); avp_write("s:outbound.itsp1.ienum.labs.nic.at","$avp(i:679)"); avp_pushto("$du","$avp(i:679)"); # route(5); # optional process received P-Asserted_Identiy header exit; } xlog("L_INFO","dbaliases lookup failed ... 404\n"); sl_send_reply("404","Unknown user"); exit; } route[3] { # UDP/TCP - thus requests received from the local main proxy. # This requests are already authenticated by the main proxy, # thus no further SIP based authentication needed. # # Perform the following steps: # 1. Try I-ENUM lookup # 2. Try U-ENUM lookup # 3. reply with 499 xlog("L_INFO","[$Tf] $rm $ru (From: $fu -> To: $tu) entering route(3): UDP/TCP requests\n"); if ( i_enum_query() ) { xlog("L_INFO","I-ENUM lookup successful, new URI = $ru, check if useful...\n"); if (uri == myself) { xlog("L_INFO","I-ENUM SIP URI belongs to myself, relay back to main proxy...\n"); sl_send_reply("100","I-ENUM succeeded, number belongs to us, sending back to main proxy..."); route(2); exit; } if (dp_can_connect()) { # check the domainpolicy of the destination xlog("L_INFO","dp_can_connect succeeded:\n"); xlog("L_INFO"," port_override_avp = '$avp(s:portoverride)'\n"); xlog("L_INFO"," transport_override_avp = '$avp(s:transportoverride)'\n"); xlog("L_INFO"," domain_prefix_avp = '$avp(s:domainprefix)'\n"); xlog("L_INFO"," domain_suffix_avp = '$avp(s:domainsuffix)'\n"); xlog("L_INFO"," send_socket_avp = '$avp(s:sendsocket)'\n"); xlog("L_INFO"," tls_client_domain_avp = '$avp(i:400)'\n"); # apply domain policy if (dp_apply_policy()) { xlog("L_INFO","I-ENUM SIP URI is a known peering partner, use this uri and relay request...\n"); xlog("L_INFO"," R-URI = $ru\n"); xlog("L_INFO"," d-URI = $du\n"); sl_send_reply("100","I-ENUM und domainpolicy succeeded, sending to other domain..."); route(4); exit; } xlog("L_INFO","dp_apply_policy failed\n"); } else { xlog("L_INFO","dp_apply_policy failed\n"); } } else { xlog("L_INFO","I-ENUM lookup failed, fallback to U-ENUM ...\n"); } revert_uri(); if ( enum_query() ) { xlog("L_INFO","U-ENUM lookup successful, new URI = $ru, relay request...\n"); sl_send_reply("100","U-ENUM succeeded, sending to other domain..."); route(4); exit; } xlog("L_INFO","I-ENUM lookup failed, send 499 back to main proxy ...\n"); sl_send_reply("499","Number not in ENUM or domainpolicy failed"); exit; } route[4]{ # This route block reformats the rpid from the internal number format # to the number format used in the I-ENUM trial. # Thus, the proxy will take the existing rpid information (Remote-Party-ID header # or P-Asserted_Identity header) and build an P-Asserted-Identity header with # a tel URI. xlog("L_INFO","[$Tf] $rm $ru (From: $fu -> To: $tu) entering route(4): add P-Asserted-Identity header\n"); # write the complete content of the Remote-Party-ID header into an AVP append_hf("Remote-Party-ID: $avp(s:rpidheader)\r\n"); # replace the sip uri with a tel uri # \1 \2 \3 \4 avp_subst("$avp(s:rpidheader)", "/^(.*)<sip:([^@]*)@[a-zA-Z0-9.]+(.*)>(.*)/tel:\2/gi"); # remove the Remote-Party-ID header and create the P-Asserted-Identity header remove_hf("Remote-Party-ID"); append_hf("P-Asserted-Identity: $avp(s:rpidheader)\r\n"); route(1); exit; } 4. Appendix 4.1. Full Config of a Main Proxy # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # Austrian I-ENUM Trial # configuration of main proxy # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 fifo="/tmp/openser_fifo" # syslog logging, configure in /etc/syslog log_facility=LOG_LOCAL0 # /var/log/openser.log server_header ="server_header: itsp1, main proxy" user_agent_header="user_agent_header: itsp1, main proxy" listen=udp:10.10.0.41:5060 listen=tcp:10.10.0.41:5060 listen=tls:10.10.0.41:5061 # our domain alias=itsp1.ienum.labs.nic.at # for addressing via outbound proxy and pre loaded route set # (otherwise will is_myself in loose_route not match) alias=outbound.itsp1.ienum.labs.nic.at # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 1 #tls_method = TLSv1 #tls_certificate = "/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/avpops.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" loadmodule "/usr/lib/openser/modules/enum.so" loadmodule "/usr/lib/openser/modules/domain.so" # ----------------- setting module-specific parameters --------------- # read and write DB access modparam("acc" , "db_url", "mysql://openser:openserrw@127.0.0.1/openser") # accounting parameters modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 0) modparam("acc", "multi_leg_enabled", 0) modparam("acc", "failed_transaction_flag", 4) # radius, syslog and DB modparam("acc", "log_flag", 1) # syslog modparam("acc", "db_flag", 2) # DB modparam("acc", "db_missed_flag", 3) # DB missed calles table # the following modules also supports multidomain, but are not used # by us: group, group_radius, speeddial, uri_db modparam("alias_db|usrloc|registrar|avpops", "use_domain", 1) # -- usrloc params -- # modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") modparam("auth_db|usrloc|registrar", "use_domain", 1) # -- 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{ xlog("L_ERR","[$Tf] [$si:$sp] $rm $ru (From: $fu -> To: $tu) request received....\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # mark transaction for accounting setflag(1); setflag(2); setflag(4); if ( is_method("CANCEL") && !t_check_trans() ) { # CANCEL without matching INVITE transaction, ignore # may happen if the INVITE is slower than the CANCEL # ignore the CANCEL, as the client will retransmit it, and maybe next time # the INVITE transaction is already created xlog("L_WARN","CANCEL without matching transaction ... ignore and discard.\n"); exit; } if ( is_method("CANCEL") ) { # CANCEL with matching INVITE transaction, just # t_relay xlog("L_INFO","CANCEL with matching transaction ... t_relay.\n"); t_relay(); exit; } # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (!has_totag()) { xlog("L_WARN","loose_route request without to-tag, 403...\n"); sl_send_reply("403", "out-of-dialog loose_route not allowed"); exit; } xlog("L_WARN","request loose_route processed...\n"); route(1); }; if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 xlog("L_INFO","local end-to-end ACK for an existent INVITE transaction detected ...t_relay()\n"); t_relay(); } else { xlog("L_WARN","ACK without matching transaction ... ignore and discard.\n"); } exit; } if ( has_totag() ) { # in-dialog requests should be handled by loose_route # this should be fixed in the client or in openser xlog("L_ERR","in-dialog request was not catched by loose_route block, 403... \n"); xlog("L_ERR","this is the complete message:\n"); xlog("L_ERR","$mb\n"); sl_send_reply("403","in-dialog request without loose_route is not allowed, this is a bug in the client or in this proxy\n"); exit; } # 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 will be done after loose_route to do not change the # ftag parameter in the Route header (eyebeam) if (!method=="REGISTER") record_route(); if (!uri==myself) { xlog("L_WARN","outoing call ...\n"); route(1); }; # if the request is for our 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") { if (t_newtran()) { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; save("location"); exit; } else sl_reply_error(); t_release(); }; if (is_method("INVITE")) { # Uncomment this if you want to use digest authentication if (from_uri == myself) { xlog("L_INFO","INVITE from local user, check auth\n"); if (!proxy_authorize("", "subscriber")) { xlog("L_INFO","auth failed...challenge\n"); proxy_challenge("", "0"); exit; }; consume_credentials(); # if avp rpid available add rpid header remove_hf("Remote-Party-ID"); remove_hf("P-Asserted-Identity"); if (is_avp_set("$avp(s:rpid)")) { xlog("L_INFO","RPID defined: rpid=$avp(s:rpid)\n"); #add Remote-Party-ID header # create new AVP with the Remote-Party-ID header avp_printf("$avp(s:temp)","$avp(s:rpid);party=calling;id-type=subscriber;screen=yes"); xlog("L_INFO", "Remote-Party-ID header: $avp(s:temp)"); append_hf("Remote-Party-ID: $avp(s:temp)\r\n"); # #add P-Asserted-Identity header # # create new AVP with the P-Asserted-Identity header # avp_printf("$avp(s:temp)","$avp(s:rpid)"); # xlog("L_INFO", "P-Asserted-Identity header: $avp(s:temp)"); # append_hf("$avp(s:temp)","P-Asserted-Identity"); } else { xlog("L_INFO","RPID not defined...\n"); } if (uri =~ "^sip:(\+|\*)?[0-9]+@") { # PSTN number detected if (uri=~"^sip:[1-9]") { xlog("L_WARN", "number normalization: illegal number, 404..."); sl_send_reply("404", "Not found - illegal number"); exit; } if (uri=~"^sip:0[1-9]") { xlog("L_INFO","Austria national number detected, remove 0, prefix +43\n"); strip(1); prefix("+43"); } else if (uri=~"^sip:00[1-9]") { xlog("L_INFO","international number detected, remove 00, prefix +\n"); strip(2); prefix("+"); } else if (uri=~"^sip:000") { xlog("L_INFO", "000 detected, illegal number, 404..."); sl_send_reply("404", "Not found - illegal number"); exit; } t_on_failure("1"); # rewritehostport("bp1.itsp1.ienum.labs.nic.at"); route(1); } if (!uri==myself) { append_hf("P-hint: Proxy1 outbound alias\r\n"); route(1); }; } else { xlog("L_INFO","incoming call...\n"); } } # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found in location table"); exit; }; xlog("L_INFO","user found, route(1)...\n"); 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 xlog("L_INFO","t_relay....\n"); if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { # handles error on border proxy xlog("L_INFO","failure_route(1): reply status = $rs\n"); if(t_was_cancelled()) { xlog("L_INFO","caller cancelled call ...exit\n"); exit; } if(t_check_status("499")) { xlog("L_INFO","callee neither in I-ENUM nor in U-ENUM...sending to PSTN Gateway...404"); t_reply("404","no PSTN gateway configured"); exit; } xlog("L_INFO","send down error code from upstream proxy...exit\n"); exit; } route[5] { # dummy route for aliases lookup lookup("aliases"); } 4.2. Full Config of a Border Proxy # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # Austrian I-ENUM Trial # configuration of main proxy # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes #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) children=4 fifo="/tmp/openserbp_fifo" # syslog logging, configure in /etc/syslog log_facility=LOG_LOCAL1 # /var/log/openserbp.log server_header ="server_header: itsp1, border proxy" user_agent_header="user_agent_header: itsp1, border proxy" listen=udp:10.10.0.41:5062 listen=udp:10.10.0.46:5062 listen=tcp:10.10.0.41:5062 listen=tcp:10.10.0.41:5080 listen=tls:10.10.0.41:5063 listen=tls:10.10.0.41:5065 alias=itsp1.ienum.labs.nic.at # # uncomment the following lines for TLS support # # default fed = fedA disable_tls = 0 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/etc/certs/fedA/itsp1/cert.pem" tls_private_key = "/etc/certs/fedA/itsp1/privkey.pem" tls_ca_list = "/etc/certs/fedA/demoCA/cacert.pem" #tls_ciphers_list= "NULL" #tls core supports only integer client domain AVPs tls_client_domain_avp = 400 tls_client_domain["v2"] { #specify parameters for a domain in particular, otherwise, #it will use the default. These are the possible parameters to #change for each domain tls_certificate = "/etc/certs/fedA/itsp1/cert.pem" tls_private_key = "/etc/certs/fedA/itsp1/privkey.pem" tls_ca_list = "/etc/certs/fedA/demoCA/cacert.pem" tls_method=TLSv1 # tls_ciphers_list= "NULL" tls_verify_server = 1 } tls_client_domain["fedc"] { #specify parameters for a domain in particular, otherwise, #it will use the default. These are the possible parameters to #change for each domain tls_certificate = "/etc/certs/fedC/itsp1/cert.pem" tls_private_key = "/etc/certs/fedC/itsp1/privkey.pem" tls_ca_list = "/etc/certs/fedC/demoCA/cacert.pem" tls_method=TLSv1 tls_verify_server = 1 } tls_server_domain[10.10.0.41:5065] { #specify parameters for a domain in particular, otherwise, #it will use the default. These are the possible parameters to #change for each domain tls_certificate = "/etc/certs/fedA/itsp1/cert.pem" tls_private_key = "/etc/certs/fedA/itsp1/privkey.pem" tls_ca_list = "/etc/certs/fedA/demoCA/cacert.pem" tls_method=TLSv1 tls_verify_client = 1 tls_require_client_certificate = 1 } tls_server_domain[10.10.0.41:5067] { #specify parameters for a domain in particular, otherwise, #it will use the default. These are the possible parameters to #change for each domain tls_certificate = "/etc/certs/fedC/itsp1/cert.pem" tls_private_key = "/etc/certs/fedC/itsp1/privkey.pem" tls_ca_list = "/etc/certs/fedC/demoCA/cacert.pem" tls_method=TLSv1 tls_verify_client = 0 tls_require_client_certificate = 1 } # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/avpops.so" loadmodule "/usr/lib/openser/modules/options.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/alias_db.so" loadmodule "/usr/lib/openser/modules/enum.so" loadmodule "/usr/lib/openser/modules/domain.so" loadmodule "/usr/lib/openser/modules/domainpolicy.so" loadmodule "/usr/lib/openser/modules/tlsops.so" # ----------------- setting module-specific parameters --------------- # -- database configuration -- # read only DB access # modparam("domain", "db_url", "postgres://openserro:openserro@localhost/openser") # XXX user at43 db for production use modparam("domain|uri_db|alias_db|lcr", "db_url", "mysql://openserro:openserro@127.0.0.1/openserbp") #modparam("avpops", "avp_url", "postgres://serro:rororo@83.136.32.160/at43") # read and write DB access modparam("acc" , "db_url", "mysql://openser:openserrw@127.0.0.1/openserbp") # the following modules also supports multidomain, but are not used # by us: group, group_radius, speeddial, uri_db modparam("alias_db|usrloc|registrar|avpops", "use_domain", 1) modparam("domain", "db_mode", 1) # Use caching in domain module # accounting parameters modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 0) modparam("acc", "multi_leg_enabled", 0) modparam("acc", "failed_transaction_flag", 4) # radius, syslog and DB modparam("acc", "log_flag", 1) # syslog modparam("acc", "db_flag", 2) # DB modparam("acc", "db_missed_flag", 3) # DB missed calles table #modparam("acc", "swap_direction", 0) #modparam("acc", "detect_direction", 1) # extra accounting of TLS paramters and sockets modparam("acc", "db_extra", "source_ip=$si;source_port=$sp;received_ip=$Ri;received_port=$Rp;tls_peer_subject=$tls_peer_subject;tls_peer_issuer=$tls_peer_issuer;tls_my_subject=$tls_my_subject;tls_my_issuer=$tls_my_issuer") # carrier enum modparam("enum", "branchlabel", "i") modparam("enum", "bl_algorithm", "cc") # we do never read from the default AVP table, thus we define a dummy # table which will never be used #modparam("avpops", "avp_table", "dummy") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("domainpolicy", "port_override_avp", "portoverride") modparam("domainpolicy", "transport_override_avp", "transportoverride") modparam("domainpolicy", "domain_prefix_avp", "domainprefix") modparam("domainpolicy", "domain_suffix_avp", "domainsuffix") modparam("domainpolicy", "send_socket_avp", "sendsocket") # ------------------------- request routing logic ------------------- # main routing logic route{ remove_hf("Supported"); route(21); # 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; setflag(1); setflag(2); setflag(3); setflag(4); # we do not allow REGISTER messages, they are # not needed for VoIP peering if (is_method("REGISTER")) { xlog("L_INFO","[$Tf] [$si:$sp] $rm $ru (From: $fu -> To: $tu) reject with 405\n"); sl_send_reply("405","REGISTER is not allowed"); exit; } # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (!has_totag()) { xlog("L_WARN","loose_route request without to-tag, 403...\n"); sl_send_reply("403", "out-of-dialog loose_route not allowed"); exit; } xlog("L_WARN","request loose_route processed...\n"); route(1); }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 xlog("L_INFO","local end-to-end ACK for an existent INVITE transaction detected ...t_relay()\n"); t_relay(); } else { xlog("L_WARN","ACK without matching transaction ... ignore and discard.\n"); } exit; } if ( has_totag() ) { # in-dialog requests should be handled by loose_route # this should be fixed in the client or in openser xlog("L_ERR","in-dialog request was not catched by loose_route block, 403... \n"); xlog("L_ERR","this is the complete message:\n"); xlog("L_ERR","$mb\n"); sl_send_reply("403","in-dialog request without loose_route is not allowed, this is a bug in the client or in this proxy\n"); exit; } # # We only allow requests which are directly addressed to us. # This is a security feature to avoid bypassing of the # main proxy routing logic if (!uri==myself) { xlog("L_INFO","[$Tf] [$si:$sp] $rm $ru (From: $fu -> To: $tu) non local domain, reject with 403\n"); sl_send_reply("403","please address requests directly to me"); exit; }; # allow OPTIONS requests for keep alive monitoring if ( is_method("OPTIONS") && !(uri =~ "sip:.*@.*") ) { # OPTIONS request for us options_reply(); exit; } # # at this time, we only have out-of-dialog requests # now, we split routing depending on the direction # the protocol can be used to identify the direction # tls: request from peering partner # udp,tcp: local request if(proto==tls) { route(2); } else { route(3); }; exit; } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { # TLS requests - thus requests received from peering partners. # This requests are already authenticated via TLS certificate # validation, thus no further SIP based authentication needed. # Just forward the requests to the main proxy. # Beforehand, we map from the anonymous IENUM SIP URIs to the # user's AoR using the dbaliases module. Note: this mapping can # also be done in the main proxy. xlog("L_INFO","[$Tf] $rm $ru (From: $fu -> To: $tu) entering route(2): TLS requests\n"); if (alias_db_lookup("dbaliases")) { xlog("L_INFO","dbaliases lookup successful, new URI = $ru\n"); avp_write("s:outbound.itsp1.ienum.labs.nic.at","$avp(i:679)"); avp_pushto("$du","$avp(i:679)"); route(5); exit; } xlog("L_INFO","dbaliases lookup failed ... 404\n"); sl_send_reply("404","Unknown user"); exit; } route[3] { # UDP/TCP - thus requests received from the local main proxy. # This requests are already authenticated by the main proxy, # thus no further SIP based authentication needed. # # Perform the following steps: # 1. Try I-ENUM lookup # 2. Try U-ENUM lookup # 3. reply with 499 xlog("L_INFO","[$Tf] $rm $ru (From: $fu -> To: $tu) entering route(3): UDP/TCP requests\n"); if ( i_enum_query("ienum.labs.nic.at") ) { xlog("L_INFO","I-ENUM lookup successful, new URI = $ru, check if useful...\n"); if (uri == myself) { xlog("L_INFO","I-ENUM SIP URI belongs to myself, relay back to main proxy...\n"); sl_send_reply("100","I-ENUM succeeded, number belongs to us, sending back to main proxy..."); route(2); exit; } if (dp_can_connect()) { # check the domainpolicy of the destination xlog("L_INFO","dp_can_connect succeeded:\n"); xlog("L_INFO"," port_override_avp = '$avp(s:portoverride)'\n"); xlog("L_INFO"," transport_override_avp = '$avp(s:transportoverride)'\n"); xlog("L_INFO"," domain_prefix_avp = '$avp(s:domainprefix)'\n"); xlog("L_INFO"," domain_suffix_avp = '$avp(s:domainsuffix)'\n"); xlog("L_INFO"," send_socket_avp = '$avp(s:sendsocket)'\n"); xlog("L_INFO"," tls_client_domain_avp = '$avp(i:400)'\n"); # apply domain policy if (dp_apply_policy()) { xlog("L_INFO","I-ENUM SIP URI is a known peering partner, use this uri and relay request...\n"); xlog("L_INFO"," R-URI = $ru\n"); xlog("L_INFO"," d-URI = $du\n"); sl_send_reply("100","I-ENUM und domainpolicy succeeded, sending to other domain..."); route(4); exit; } xlog("L_INFO","dp_apply_policy failed\n"); } else { xlog("L_INFO","dp_apply_policy failed\n"); } } else { xlog("L_INFO","I-ENUM lookup failed, fallback to U-ENUM ...\n"); } revert_uri(); if ( enum_query("ienum.labs.nic.at") ) { xlog("L_INFO","U-ENUM lookup successful, new URI = $ru, relay request...\n"); sl_send_reply("100","U-ENUM succeeded, sending to other domain..."); route(4); exit; } xlog("L_INFO","I-ENUM lookup failed, send 499 back to main proxy ...\n"); sl_send_reply("499","Number not in ENUM or domainpolicy failed"); exit; } route[4]{ # This route block reformats the rpid from the internal number format # to the number format used in the I-ENUM trial. # Thus, the proxy will take the existing rpid information (Remote-Party-ID header # or P-Asserted_Identity header) and build an P-Asserted-Identity header with # a tel URI. xlog("L_INFO","[$Tf] $rm $ru (From: $fu -> To: $tu) entering route(4): add P-Asserted-Identity header\n"); # uncomment the next few lines if the local rpid format is # Remote-Party-ID: <sip:+43101@proxy1.ienum.labs.nic.at>;... # first the dirty hack version: just use a regexp over the whole message # this will fail with unusual formated SIP messages #if (subst('/^Remote-Party-ID:(.*)sip:([^@]*)@[a-zA-Z0-9.]+(.*)$/P-Asserted-Identity: tel:\2\r/ig')) { # xlog("L_INFO","successfuly replaced Remote-Party-ID with /P-Asserted-Identity...\n"); #} else { # xlog("L_INFO","failure replacing Remote-Party-ID with P-Asserted-Identity...500\n"); # sl_send_reply("500","failure replacing Remote-Party-ID with P-Asserted-Identity"); # exit; #} # # now the more complicated version using AVPs, should be more robust: # write the complete content of the Remote-Party-ID header into an AVP append_hf("Remote-Party-ID: $avp(s:rpidheader)\r\n"); # replace the sip uri with a tel uri # \1 \2 \3 \4 avp_subst("$avp(s:rpidheader)", "/^(.*)<sip:([^@]*)@[a-zA-Z0-9.]+(.*)>(.*)/tel:\2/gi"); # remove the Remote-Party-ID header and create the P-Asserted-Identity header remove_hf("Remote-Party-ID"); append_hf("P-Asserted-Identity: $avp(s:rpidheader)\r\n"); route(1); exit; } route[5]{ # This route block reformats the rpid received in the P-Asserted-Identity header # (the number format used in the I-ENUM trial) into the local used format. # The local used format may be for example a Remote-Party-ID header, a # P-Asserted_Identity header or a rewritten From header. xlog("L_INFO","[$Tf] $rm $ru (From: $fu -> To: $tu) entering route(5): process P-Asserted-Identity header\n"); route(1); exit; } onreply_route { # default onreply_route route(22); } route[21] { xlog("L_INFO","$ci [$Tf]: -------------------- new SIP request received --------------------\n"); xlog("L_INFO","$ci [$si:$sp] $rm $ru (From: $fu -> To: $tu)\n"); xlog("L_INFO","[$Tf] $$tls_peer_subject = '$tls_peer_subject'\n"); xlog("L_INFO","[$Tf] $$tls_peer_subject_cn = '$tls_peer_subject_cn'\n"); xlog("L_DBG", "message buffer:\n"); xlog("L_DBG", "$mb\n"); } route[22] { xlog("L_INFO","$ci [$Tf]: -------------------- new SIP response received --------------------\n"); xlog("L_INFO","$ci [$si:$sp] $rs ($cs) $rr (From: $fu <- To: $tu)\n"); xlog("L_INFO","[$Tf] $$tls_peer_subject = '$tls_peer_subject'\n"); xlog("L_INFO","[$Tf] $$tls_peer_subject_cn = '$tls_peer_subject_cn'\n"); xlog("L_DBG", "message buffer:\n"); xlog("L_DBG", "$mb\n"); } 1. Why do we need SIP peering? When SIP was developed, there was no need for SIP peering. SIP was considered to be as open as email. Like everybody can send emails to everyone, every SIP user should be able to contact every other SIP user via SIP. Therefore, two requirements are necessary: - The SIP provider of the caller must allow calling SIP URIs which are outside of the local SIP domain. - The SIP provider of the callee has to allow incoming SIP calls from callers outside of the local domain. For further discussion, a SIP services which fulfill these two requirements will be referred as "open SIP service". Currently there are lots of open SIP services, e.g. iptel.org or fwd.pulver.com. But there are also other SIP services, which are "closed" or "restricted". Closed means, that the SIP service provider allows only internal SIP calls (when calling other users of this SIP service provider). All calls to other users will be routed via the PSTN. Restricted means, that the SIP service provider allows SIP calls from/to certain domains, but not all domains. This of course requires authentication of incoming calls. (examples of closed service providers are yahoo broadband, inode, vonage, IMS/NGN) There are mainly 3 reasons why a service provider does not offer an open service, but only closed or restricted: - business model: The service provider need the PSTN fees for its business model. - security: The service provider is afraid of possible security impacts - spit: The service provider is afraid of SPIT (VoIP SPAM) which will probably occur sooner or later (like email spam) Having "open" SIP providers while other SIP providers are "closed" leads to several interconnection problems. Following are a few examples: - A user of an open SIP service calls a user of a closed SIP service using the callees SIP URI. The callees SIP provider ignores the incoming SIP messages, causing failed call setups (without any information why the call failed). - A user of a closed SIP service calls a user of an open SIP service via its E.164 phone number. Although the callee can be called directly via ENUM+SIP, the call will be routed via the PSTN causing higher call setup times, call costs and probably worse quality due to multiple transcoding and packetization. - Two users want to have a SIP video conversation amongst them. Both SIP providers support video sessions, but as the interconnect between both providers will be done via the PSTN, there is no possibility to have multimedia session (video, IM, presence). To resolve these issues, a dedicated peering mode is necessary. Probably lots of people will argue that peering is not necessary and SIP providers should just open their SIP services. But this will lead to the same problems we have with email. Further, if there are service providers which do not want open there service for everybody, their position should be accepted. Even if a service provider opens its SIP service, it might want to apply some restrictions on how the interconnect will be done. Thus, to encourage SIP providers to open their SIP service, we need to offer them mechanisms to authenticate other service providers. There are several ways to authenticate SIP users or other SIP service providers, e.g: SIP over TLS (certificate based authentication), IP address based (TCP or TLS as UDP can be spoofed easily), SIP identity-draft, SIP+S/MIME, SIP+domainkeys, Layer2 VPNs, IPsec, ... Regardless of the technology the service provider requires for authenticating calls, the technology must be somehow announced to the other service providers. This is the reason why we need - draft-lendl-domain-policy-ddds-02.txt and - draft-lendl-speermint-federations-02.txt These drafts define federations and how a service prodiver can publish its peering policies (= federation memberships). 2. What is a federation A federation is like a club. All the members share a common interest. In case of VoIP peering, the members share the common wish to interconnect directly via SIP instead of the old style TDM/SS7 interconnect. A federation needs at least 2 members (this is like a bilateral peering agreement), but usualy may have several members which treat each other as identical. There will be lots of different federations with different goals an dinterconnect policies. E.g. one federation may have to policy to have settlement free peering (sender keeps all) and everbody may participate as long they authenticate via TLS. Other federations may still stay with current settlement policies (termination fees ...) and members most fulfill certain requirements (e.g. must be a mobile operator ...). Thus, it may often happen that a VoIP service provider is a member of several federations. The next sections shows how service providers may participate in several federations and how the domainpolicy technology can be used to achieve "plug-and-play" interworking in several federations. 3. Use Cases - The Federations In the following example we show the use case of a VoIP service provider. This virtual VoIP service provicer is located in Austria and offers VoIP services via cable technology. It is an early adopter of the "triple play" offering TV, Internet and Voice to its customers. Let's call this VoIP service provider "CableItsp1". There are also other cable technology based VoIP service providers in Austria. As they mostly have their local regions without overlapping, these cable guys are not competitors. Their competitors are traditional voice service providers (VSPs). These cable guides set up a federation called "cable-guys-austria" to fight jointly against their competitors. Further there business model is not based on selling voice mintues, but an a monthly subscription fee with flat rate. Thus, having settlement free voice interconnect between each others will be a benefit as they can simplify their billing systems. Further, there is a worldwide federation called "voip-providers-international" which allow every VoIP service provider to join and allow settlement free worldwide interconnect - as long as the members pay they monthly federation subscription. Thus this federation may be interesting for VoIP service providers with a lot of international connections. To have one more example, there is an Austrian federation called "voip-peering-austria" which still insists of interconnect fees. Thus, the simple but very reasonable goal is to keep VoIP calls on IP and avoid transcoding and gatewaying for TDM based interconnect. The settlement must be still agreed bilatereal between the VSPs. All of these federations have different goals and different interconnect policies: cable-guys-austria ================== subscription fee: no settlement: no fees technology: SIP via UDP via IPsec (commom shared secret) via well known IP addresses URI (identifier): http://www.cableguysaustria.at/peering-v1 domain-suffix: cableguysaustria.at voip-providers-international ============================ subscription fee: yes, monthly settlement: no fees technology: SIP via TCP with IP access lists SIP-port: 5080 URI (identifier): http://www.voip-providers-international.com/peeringv1 voip-peering-austria ==================== subscription fee: no settlement: fees must be agreed bilateral technology: SIP via TLS, certificate signed by voip-peering-austria TLS-port: 6666 URI (identifier): http://www.voip-peering-austria.at/TlsPeeringV1 Note: Why some use a dedicated port instead of the SIP default port will be described later. 4. Finding and Peering 4.1. Is the callee reachable via SIP? Before any interconnect policies can be applied, the originating service provider has to find out to which VSP a phone number belongs. These can be done via lookup tables. Probably the most open and commen lookup mechanism is ENUM, especially infrastructure ENUM. The VSP registers the ENUM domains for its numbers and announces the SIP URI for inbound routing in the NAPTR. Thus, the originating service provider performs an IENUM lookup for the called phone number. If there is no ENUM entry for a called number, the originating VSP will send the call to its PSTN gateway as usual. If there is a SIP URI in ENUM, the next step is: 4.2. Does the callee's VSP accept VoIP interconnect from the originator? The SIP URI returned from the IENUM lookup points to the terminating service provider. Thus, the domain in the SIP URI is the "key" to find out the peering policies of the terminating service provider. For example, the IENUM lookup for the called number returns sip:+4312345@sip.itsp66.com. Then, the domain "sip.itsp66.com" will be used as the input data for the domainpolicy algorithm described in draft-lendl-domain-policy-ddds. The originating service provider will perform NAPTR DNS lookups for the domain "sip.itsp66.com" with service types "D2P+...". In this example, there will be 3 NAPTRs returned: $ORIGIN sip.itsp66.com. IN NAPTR 10 50 "U" "D2P+SIP:fed" "!^.*$!http://www.cableguysaustria.at/peering-v1!" . IN NAPTR 20 50 "U" "D2P+SIP:fed" "!^.*$!http://www.voip-peering-austria.at/TlsPeeringV1!" . IN NAPTR 30 50 "U" "D2P+SIP:fed" "!^.*$!http://sipxyconnect.example.org/!" . Now the originator verifies the published federation identifiers agains a local table, in which the originating VSP has configured its federation membership. If there is a matching federation, the next step will applies. 4.3. Apply the federation policies Different federations will use different techniques to interconnect and authenticate each other. Thus, before sending the call to the terminating VSP, the federatations policy must be applied. E.g. if a federation authenticate each other using TLS with certificates signed by the federation's CA, the border proxy must switch to TLS and present the proper certificate to the ingress point of the terminating VSP. Some of these "attributes" must be applied by the border proxy (e.g. tranport protocol, dedicated SIP signaling port, TLS certificate ...) whereas others might be application agnostic (e.g. IPsec encryption, VPNs ...). 5. ENUM routing and federation policies When a VSP publishes its ingress proxy for VoIP peering in ENUM, it publishes only a single NAPTR. Although then ENUM standard allows publishing of several "SIP NAPTRs" for a certain number there is no semantics defined how to deal with multiple received SIP NAPTRs (serial forking? parallel forking? which one to choose?). Thus it is suggested to publish only a single SIP NAPTR in ENUM. Using plain ENUM and plain SIP, all incoming requests for a certain phone number will be received on the same ingress proxy. This is fine in a "open SIP service" environment, but not in a dedicated peering scenario where traffic from certain peering partners should be received on certain ingress points. A common use case is the use of TLS. As TLS does not allow "virtual hosting", authentication with different certificates for different peering partners requires a dedicated socket for each "TLS domain". Another use case is peering in a dedicated VLAN with public IP addresses. If a VSP has an ingress proxy in the public Internet and an ingress proxy in a private LAN (for dedicated peering partners), there must be an algorithm which allows to originating VSP to resolve from the common SIP URI to the respective IP address (public or private). This can be done by applying federation policies to the SIP URI. We will call these "override attributes" as they override the default SIP routing behavior described in RFC 3263. Each federation defines its override attributes which allow to "manipulate" the default RFC3263 call routing according to the federation's policy. Currently the openser domainpolicy module supports the following override attributes: - port_override: If this attribute is set, the port in the destination URI is set to this port. Setting an override port disables NAPTR and SRV lookups according to RFC 3263. - transport_override: If this attribute is set, the transport parameter in the destination URI is set to the specified transport ("udp", "tcp", "tls"). Setting an override transport also disables NAPTR lookups, but retains an SRV lookup according to RFC 3263. - domain_prefix: If this attribute is set, the domain in the destination URI will be prefixed with this "subdomain". E.g. if the domain in the request URI is "example.com" and the domain_prefix contains "inbound", the domain in the destinaton URI is set to "inbound.example.com". - domain_suffix: If this attribute is set, the domain in the destination URI will have the content of the attribute appended to it. E.g. if the domain in the request URI is "example.com" and the domain_suffix contains "myroot.com", the domain in the destination URI is set to "example.com.myroot.com". Thus, using override attributes manipulates the destination URI - the socket to which the message will be sent. Nevertheless, the request URI (the SIP URI in the first line of the SIP request) is untouched as will always be the value of the ENUM lookup. 6. Configuring inbound routing This section shows how the previously presented VSP "CableItsp1" configures it inbound routing. As desrcribed earlier, CableItsp1 is member of 3 different federations, each having its own interconnect policies. 6.1. ENUM CableItsp1 chooses a simple format for its NAPTRs: the user part is the E.164 phone number and the domain part is its domain cableitsp1.at, for example: $ORIGIN 5.4.3.2.1.3.4.e164.arpa. IN NAPTR 10 50 "U" "E2U+sip" "!^.*$!sip:+4312345@cableitsp1.at!" . 6.2. Announcing federation memberships CableItsp1 announces its federation memberships (= its interconnect policies) in the DNS: $ORIGIN cableitsp1.at. IN NAPTR 10 50 "U" "D2P+SIP:fed" "!^.*$!http://www.cableguysaustria.at/peering-v1!" . IN NAPTR 20 50 "U" "D2P+SIP:fed" "!^.*$!http://www.voip-peering-austria.at/TlsPeeringV1!" . IN NAPTR 30 50 "U" "D2P+SIP:fed" "!^.*$!http://www.voip-providers-international.com/peeringv1!" . 6.3. Federation ingress configuration 6.3.1 Federation cable-guys-austria As this federation uses IPsec for authentication, CableItsp1 decided to use a dedicated IP address 11.22.33.44 on its border proxy for dealing with the IPsec stuff. Thus, the domain cableitsp1.at.cableguysaustria.at (SIP URI domain + domain-suffix according to the federation policies) will resolve to 11.22.33.44. CableItsp1 configues the IPsec to use the shared secret of the federation and forbidds any SIP signalling from/to 11.22.33.44 without IPsec. The list of IP addresses of interconnect partners (members of this federation) will be downloaded once a day from the federations web server and the IPsec configuration will be updated accordingly. In the openser call routing, inbound calls from this federation can be detected by: if(dst_ip==11.22.33.44) { log("message received via IPsec\n"); ... }; 6.3.2 Federation voip-providers-international This federation uses SIP over TCP over port 5080. Thus, CableItsp1 configures its border proxy to also listen on port 5080 for TCP, and adds the IP addresses of the peers to the IP access list on its firewall. In the openser call routing, inbound calls from this federation can be detected by: if( (dst_port==5080) && (proto==TCP) ) { log("message received via TCP port 5080\n"); ... }; 6.3.3 Federation voip-peering-austria This federation uses TLS for authentication. CableItsp1 generated a certificate signing request (CSR) using the openssl tools. This CSR was signed by the voip-peering-austria CA. Then the border proxy was configured to listen on port 6666 for TLS and use the respective certificate and CA list: listen=tls:10.10.0.41:6666 tls_server_domain[10.10.0.41:6666] { tls_certificate = "/etc/certs/voip-peering-austria/cert.pem" tls_private_key = "/etc/certs/voip-peering-austria/privkey.pem" tls_ca_list = "/etc/certs/voip-peering-austria/cacert.pem" tls_method=TLSv1 tls_verify_client = 1 tls_require_client_certificate = 1 } In the openser call routing, inbound calls from this federation can be detected by: if( (dst_port==6666) && (proto==TLS) ) { log("message received via TLS on port 6666\n"); ... }; 7. Configuring outbound routing The call routing configuration for domainpolicy based routing is rather simple: if (dp_can_connect()) { # check the domainpolicy of the destination # apply domain policy if (dp_apply_policy()) { t_relay(); exit; } ... } The configuration for the domainpolicy module is mainly entering the federation policy in the domainpolicy table. The domainpolicy table consists of the following columns: id: unique id rule: Name of column containing the domain policy rule name which is equal to the URI as published in the domain policy NAPTRs (the federation identifier). att: The override attribute. Name of column containing the AVP's name. If the rule stored in this row triggers, than dp_can_connect() will add an AVP with that name. val: Value of the override attribute. Name of column containing the value for AVPs created by dp_can_connect(). comment: a description of these table entry The id and comment table are not used by openser. The following table shows the required domainpolicy table for the previously defined federations: id rule type att val comment 1 http://www.voip-providers-international.com/peeringv1 fed portoverride 5080 2 http://www.voip-providers-international.com/peeringv1 fed transportoverride tcp 3 http://www.voip-peering-austria.at/TlsPeeringV1 fed tls_client_domain voip-peering-austria 4 http://www.voip-peering-austria.at/TlsPeeringV1 fed portoverride 6666 5 http://www.voip-peering-austria.at/TlsPeeringV1 fed transportoverride tls 6 http://www.cableguysaustria.at/peering-v1 fed transportoverride udp 7 http://www.cableguysaustria.at/peering-v1 fed sendsocket 11.22.33.44 Further, it requires the configuration of a TLS client domain and the AVPs for the domainpolicy module: tls_client_domain["voip-peering-austria"] { tls_certificate = "/etc/certs/voip-peering-austria/cert.pem" tls_private_key = "/etc/certs/voip-peering-austria/privkey.pem" tls_ca_list = "/etc/certs/voip-peering-austria/cacert.pem" tls_method=TLSv1 tls_require_server_certificate = 1 } modparam("domainpolicy", "port_override_avp", "portoverride") modparam("domainpolicy", "transport_override_avp", "transportoverride") modparam("domainpolicy", "domain_prefix_avp", "domainprefix") modparam("domainpolicy", "domain_suffix_avp", "domainsuffix") modparam("domainpolicy", "sendsocket_avp", "sendsocket")

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[Users] dialog expired accounting
by tele 24 Oct '06

24 Oct '06

Hi all, I've read about the dialog module and the new "default_timeout" or "timeout_avp" features. What i need to know it's if in case of default_timeout or timeout by avp the calls are anyway accounted. How are accounted the calls if dialog timeout expired? thanks :tele

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[Users] OpenSER Dokuwiki updated
by Daniel-Constantin Mierla 24 Oct '06

24 Oct '06

Hello, for those which haven't spotted yet, OpenSER dokuwiki site has been updated. Lot of useful documentation was brought to light, the new index should ease the browsing and let find quickly the relevant pages. Also, the information is grouped now on namespaces, based on topics. We would like to thank to all that contributed with examples, code snippets or they shared the VoIP experiences there. http://www.openser.org/dokuwiki/ Cheers, Daniel

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[Users] OpenSER Summit - Preview (I)
by Daniel-Constantin Mierla 24 Oct '06

24 Oct '06

Hello, the agenda is getting close to final version. I just want to preview the content of the event, now with two weeks before the event. I believe we manged to bring there in discussion most of the domains VoIP could touch and hot topics nowadays. We cover from development to business cases, from research to IMS, 3/4G. Next is a briefing of agenda, as of today, although it may change a bit in the future. The summit day (November 8) will start with introduction of special guests and history of OpenSER. It will be followed by a set of heavy subjects of VoIP: Application Server based on Java SipServlet technology, geographic redundancy and VoIP on mobile network. VoIPUser open VoIP services will then show their statistics and evolution over the years, and how they become one of the biggest VoIP communities world wide. Noon break will allow to visit the exhibition and group chats for hot discussions. Afternoon session will bring very interesting subjects: safe peering; high availability; OpenSER IMS prototype; developing VoIP applications; a look educational VoIP infrastructures - INRIA & SIP.edu plus a walk through VoIP opportunities. We end the day with what should be very entertaining: open discussion about OpenSER future and roadmap. This is not all, there are other special guests that can give you key notes about other topics and products of VoIP. As usual, details at: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… More will come as the event approaches. Cheers, Daniel

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[Users] X-lite is not Working, But SipEyeMedia Softphones is Working Fine.
by sunkara 24 Oct '06

24 Oct '06

Hello Users, I don't Know What Happening in this Issues... Help me in this . When Testing the X-lite Softphones, is not working. When I use the Sip EyeMedia Softphone it Working Fine. Is Problem in X-lite Configuration or Router(Firewall Gateway Keepers ) is blocking it... Help me . -- Thanks & Regards, *Ravi Prakash Sunkara* *M*:+91 9985077535* O*:+91 40 23114549* F*:+91 40 40208727 /*ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com */

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[Serusers] Dropping Replies
by Sebastian Damm 24 Oct '06

24 Oct '06

Hi, I know that what I want to do is a bit odd but here is why: When using Asterisk as PSTN gateway and using SIP INFO for DTMF digits, there is a problem with devices not understanding SIP INFO. When Asterisk sends out its INFO packet and gets back a negative reply, it stops bridging and throws away the whole call. That is a really bad issue on Asterisk side, but I just wanted to find a workaround. So what I wanted to do is drop negative replies on INFO packets, so Asterisk just retransmits them and does not get any feedback. Those INFO packets come through my loose route and there I put the following little statement in there: if(method=="INFO" && src_ip == 10.0.0.1 ) { xlog("L_NOTICE"," - INFO Request: %mb"); t_on_reply("7"); setflag(3); }; With that answers jump into onreply_route[7]. This part works ok. onreply_route[7] { if (search("CSeq:[0-9 ]+INFO")) { xlog("L_NOTICE"," - INFO Reply: %mb"); if(t_check_status("405")) { xlog("L_NOTICE"," - Dropping INFO Reply"); drop(); } } } This part does not work. I also tried creating a route just for dropping and then jumpinng into this route from my reply_route, but then I also only have the log line from that route, but drop does not work. I tried sl_send_reply, t_release and other commands of which I thought they might be helpful. But the "offending answer" is always transmitted to my gateway. Does anybody have an idea how to just "throw away" reply packets? Thanks in advance, Sebastian

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[Users] generating ACK
by Ramin Dousti 23 Oct '06

23 Oct '06

Hello, How can I generate an ACK locally on openser proxy as a responsetof a failure? And subsequently drop the failed message. In other word I do not want to pass the failed message back to my UA, I just want to process it on openser... Thanks in advance, -- Ramin

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[Users] Changing TM Retransmission Count?
by Tavis P 23 Oct '06

23 Oct '06

Reading through the documentation for the TM module (1.1.0) i've been unable to find anything indicating that i can change the retransmission count, only the retransmission intervals By my count OpenSER is retransmitting an INVITE to an endpoint user 12 times and i would like to reduce this as it is interfering with the call forward no answer logic i've put in place (timer doesn't hit until all 12 retransmissions are sent, which is about 42 seconds using the default timers) Do i have any options other than modifying the tm module code? If so, could someone give me a pointer where i need to look (i've had a quick read and its rather complicated so i might take me some time to find the logic i'm looking for) tavis

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[Users] aliases
by Benko 23 Oct '06

23 Oct '06

Hello! Are entries in the aliases table only read at startup? While a record that was in the table before i started openser is available since the beginning, a entry i inserted a while ago(+1h) but after starting openser is still not working(and also doesn't appear in the memory cache). I'd be glad if someone can give me info on this particular matter, thx! Regards Christian p.s.: just stumbled over dbaliases, guess this will work better...

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