- sr-users - kamailio.org

RE: [Users] DNS queries and TLS
by Papadopoulos Georgios 15 Dec '06

15 Dec '06

Hi Klaus, Here are the packets. The ngrep of the DNS lookup does not show much. I can only see the client IP (in reverse order). client: 213.5.17.236 Openser: 213.5.43.134, 213.5.8.5 DNS: 213.5.41.8 thank you U 2006/12/14 12:00:09.612841 213.5.17.236:5060 -> 213.5.43.134:5060 INVITE sip:2116872933@i-call.gr;user=phone SIP/2.0. Via: SIP/2.0/UDP 213.5.17.236:5060;branch=z9hG4bK-nykmi2fjal1m;rport. From: "George Papadopoulos" <sip:geop@i-call.gr>;tag=pt3u7ggbqm. To: <sip:2116872933@i-call.gr;user=phone>. Call-ID: 3c28be1c1adb-0vv5406jgtfv@213-5-17-236. CSeq: 2 INVITE. Max-Forwards: 70. Contact: <sip:geop@213.5.17.236:5060;line=9yt55o5b>. P-Key-Flags: keys="3". User-Agent: snom190-3.56w. Accept-Language: en. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer. Supported: timer, 100rel, replaces. Session-Expires: 3600. Content-Type: application/sdp. Content-Length: 268. . v=0. o=root 886017132 886017132 IN IP4 213.5.17.236. s=call. c=IN IP4 213.5.17.236. t=0 0. m=audio 10190 RTP/AVP 0 8 18 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:18 g729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2006/12/14 12:00:09.614000 213.5.8.5:50120 -> 213.5.41.8:53 .............236.17.5.213.in-addr.arpa..... # U 2006/12/14 12:00:09.618660 213.5.41.8:53 -> 213.5.8.5:50120 .............236.17.5.213.in-addr.arpa.............*0.7.ns0.altectelecom s.gr. hostmaster.;w.e...p.... ..:...Q. # U 2006/12/14 12:00:09.619325 213.5.43.134:5060 -> 213.5.17.236:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 213.5.17.236:5060;branch=z9hG4bK-nykmi2fjal1m;rport=5060. From: "George Papadopoulos" <sip:geop@i-call.gr>;tag=pt3u7ggbqm. To: <sip:2116872933@i-call.gr;user=phone>. Call-ID: 3c28be1c1adb-0vv5406jgtfv@213-5-17-236. CSeq: 2 INVITE. Server: OpenSer (1.1.0-notls (x86_64/linux)). Content-Length: 0. . > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Friday, December 15, 2006 12:42 PM > To: Papadopoulos Georgios > Cc: Bogdan-Andrei Iancu; users(a)openser.org > Subject: Re: [Users] DNS queries and TLS > > Can you also post the INVITE which is processd and the ngrep > of the DNS lookup (which domain is lookup up) > > regards > klaus > > Papadopoulos Georgios wrote: > > Hi Bogdan, > > > > I am not using any nat test functions, at least at the part of the > > script that handles INVITEs. I am using allow_trusted() in > caching mode. > > But if allow_trusted was doing the DNS query then I should > also see it > > after the first INVITE and before the "407 Proxy Auth > Reqd". I am also > > using lcr module. So, is it possible that load_gws() or > next_gw() is > > responsible for the DNS query? > > > > Regards, > > > > George > > > > > > > >> -----Original Message----- > >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >> Sent: Thursday, December 14, 2006 7:14 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] DNS queries and TLS > >> > >> Hi George, > >> > >> > >> TLS = Thread Local Storage > >> > >> > >> > >> Papadopoulos Georgios wrote: > >> > >>> Hello, > >>> > >>> I am having some performance issues with Openser and I tend > >> to believe > >>> they are related with DNS. So I am trying to figure out > >> when and why > >>> Openser is doing DNS queries. Doing ngrep on port 53 I > >> realized that > >>> right before sending out the "100 trying -- your call is > >> important to > >>> us" message, Openser is doing a reverse DNS lookup for > the IP where > >>> the INVITE came from. So the sequence is something like: > >>> client Openser DNS > >>> |---INVITE------------------------>| > >>> |<---407 Proxy Auth Reqd---| > >>> |---ACK--------------------------->| > >>> |---INVITE------------------------>| > >>> |---client > IP?-------->| > >>> > >> |<-------------------------| > >>> |<--------------------100 trying---| > >>> > >>> > >>> I am using Openser-1.1.0-notls and in my script I have dns=no > >>> rev_dns=no So first question is whether this DNS query is > necessary > >>> and how I could avoid it. > >> the configuration options are ok (as time you do not use > the command > >> line -r or -R). I would say the rev dns query is not > triggered by the > >> t_relay() (actually the params control what DNS queries should be > >> done when testing the "received" VIA param) - I have > tested and I see > >> no query before 100 trying, so it should be ok. > >> > >> maybe you are using some nat test functions (like > >> client_nat_test) or any other script functions that mask a dns > >> query...can you check on this? > >> > >>> > >>> What is furthermore confusing is that I have a test > system with the > >>> same Openser version and same script, where this DNS query is not > >>> happening. Looking into the production system I found the > following: > >>> ser2:/usr/local/openser-1.1.0-notls/sbin# ldd openser > >>> linux-gate.so.1 => (0xffffe000) > >>> libdl.so.2 => /lib/tls/libdl.so.2 (0x55571000) > >>> libresolv.so.2 => /lib/tls/libresolv.so.2 (0x55574000) > >>> libc.so.6 => /lib/tls/libc.so.6 (0x55587000) > >>> /lib/ld-linux.so.2 => /lib/ld-linux.so.2 > >> (0x55555000) whereas > >>> the test system shows: > >>> sertest:/usr/local/openser-1.1.0-notls/sbin# ldd openser > >>> libdl.so.2 => /lib/libdl.so.2 (0x7002c000) > >>> libresolv.so.2 => /lib/libresolv.so.2 (0x70040000) > >>> libc.so.6 => /lib/libc.so.6 (0x70064000) > >>> /lib/ld-linux.so.2 (0x70000000) So the two > systems link to > >>> different libresolv.so libraries. Is the > tls/libresolve.so that is > >>> responsible for the DNS query? Given that in both cases I > >> am using the > >>> notls version of Openser 1.1, why is there a difference > between the > >>> two? > >> the "tls" frm /lib/tls comes from "Thread Local Storage" and there > >> are libraries implementations for thread env. It has nothing to do > >> with TLS (Transport Layer Security) > >> > >> regards, > >> bogdan > >> > >>> > >>> thank you for any help > >>> > >>> George > >>> > >>> > >>> > >>> Disclaimer > >>> > >>> The information in this e-mail and any attachments is > >> confidential. It > >>> is intended solely for the attention and use of the named > >>> addressee(s). If you are not the intended recipient, or person > >>> responsible for delivering this information to the intended > >> recipient, > >>> please notify the sender immediately. Unless you are the intended > >>> recipient or his/her representative you are not > authorized to, and > >>> must not, read, copy, distribute, use or retain this > message or any > >>> part of it. E-mail transmission cannot be guaranteed to be > >> secure or > >>> error-free as information could be intercepted, corrupted, lost, > >>> destroyed, arrive late or incomplete, or contain viruses. > >>> > >>> ------------------------------------------------------------- > >> ---------- > >>> - > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >> > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > -- > Klaus Darilion > nic.at > >

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RE: [Users] DNS queries and TLS
by Papadopoulos Georgios 15 Dec '06

15 Dec '06

Hi Bogdan, I am not using any nat test functions, at least at the part of the script that handles INVITEs. I am using allow_trusted() in caching mode. But if allow_trusted was doing the DNS query then I should also see it after the first INVITE and before the "407 Proxy Auth Reqd". I am also using lcr module. So, is it possible that load_gws() or next_gw() is responsible for the DNS query? Regards, George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Thursday, December 14, 2006 7:14 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] DNS queries and TLS > > Hi George, > > > TLS = Thread Local Storage > > > > Papadopoulos Georgios wrote: > > > Hello, > > > > I am having some performance issues with Openser and I tend > to believe > > they are related with DNS. So I am trying to figure out > when and why > > Openser is doing DNS queries. Doing ngrep on port 53 I > realized that > > right before sending out the "100 trying -- your call is > important to > > us" message, Openser is doing a reverse DNS lookup for the IP where > > the INVITE came from. So the sequence is something like: > > client Openser DNS > > |---INVITE------------------------>| > > |<---407 Proxy Auth Reqd---| > > |---ACK--------------------------->| > > |---INVITE------------------------>| > > |---client IP?-------->| > > > |<-------------------------| > > |<--------------------100 trying---| > > > > > > I am using Openser-1.1.0-notls and in my script I have dns=no > > rev_dns=no So first question is whether this DNS query is necessary > > and how I could avoid it. > > the configuration options are ok (as time you do not use the > command line -r or -R). I would say the rev dns query is not > triggered by the > t_relay() (actually the params control what DNS queries > should be done when testing the "received" VIA param) - I > have tested and I see no query before 100 trying, so it should be ok. > > maybe you are using some nat test functions (like > client_nat_test) or any other script functions that mask a > dns query...can you check on this? > > > > > What is furthermore confusing is that I have a test system with the > > same Openser version and same script, where this DNS query is not > > happening. Looking into the production system I found the following: > > ser2:/usr/local/openser-1.1.0-notls/sbin# ldd openser > > linux-gate.so.1 => (0xffffe000) > > libdl.so.2 => /lib/tls/libdl.so.2 (0x55571000) > > libresolv.so.2 => /lib/tls/libresolv.so.2 (0x55574000) > > libc.so.6 => /lib/tls/libc.so.6 (0x55587000) > > /lib/ld-linux.so.2 => /lib/ld-linux.so.2 > (0x55555000) whereas > > the test system shows: > > sertest:/usr/local/openser-1.1.0-notls/sbin# ldd openser > > libdl.so.2 => /lib/libdl.so.2 (0x7002c000) > > libresolv.so.2 => /lib/libresolv.so.2 (0x70040000) > > libc.so.6 => /lib/libc.so.6 (0x70064000) > > /lib/ld-linux.so.2 (0x70000000) So the two systems link to > > different libresolv.so libraries. Is the tls/libresolve.so that is > > responsible for the DNS query? Given that in both cases I > am using the > > notls version of Openser 1.1, why is there a difference between the > > two? > > the "tls" frm /lib/tls comes from "Thread Local Storage" and > there are libraries implementations for thread env. It has > nothing to do with TLS (Transport Layer Security) > > regards, > bogdan > > > > > thank you for any help > > > > George > > > > > > > > Disclaimer > > > > The information in this e-mail and any attachments is > confidential. It > > is intended solely for the attention and use of the named > > addressee(s). If you are not the intended recipient, or person > > responsible for delivering this information to the intended > recipient, > > please notify the sender immediately. Unless you are the intended > > recipient or his/her representative you are not authorized to, and > > must not, read, copy, distribute, use or retain this message or any > > part of it. E-mail transmission cannot be guaranteed to be > secure or > > error-free as information could be intercepted, corrupted, lost, > > destroyed, arrive late or incomplete, or contain viruses. > > > >------------------------------------------------------------- > ---------- > >- > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > >

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[Users] Sip Message
by Howard Tang 15 Dec '06

15 Dec '06

Hi, Can someone tell me how i can see the SIP message going through in openser please? I want to debug why openser return "Address Incomplete". Regards, Howard

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[Serusers] click2dial issue
by hiSIPatH 15 Dec '06

15 Dec '06

Hi all, i am trying to use click 2 dial in phonebook (serweb). But unfortunately i am not able to get this to work. I always get the following error in the popup window: 407 Proxy Authentication Required The sip trace gives the following ( i have sniffed eth0 and lo) on the eth0 (with public ip@ the dns query and response) No. Time Source Destination Protocol Info 16 3.618415 89.xxx.xxx.218 89.xxx.xxx.129 DNS Standard query SRV _sip._udp.yyyy.zzz No. Time Source Destination Protocol Info 17 3.618868 89.xxx.xxx.129 89.xxx.xxx.218 DNS Standard query response SRV 200 0 5060 proxy.yyyy.zzz on my loopback interface No. Time Source Destination Protocol Info 1 0.000000 89.xxx.xxx.218 89.xxx.xxx.218 SIP/SDP Request: INVITE sip:firstname.lastname@yyyy.zzz, with session description No. Time Source Destination Protocol Info 2 0.000251 89.xxx.xxx.218 89.xxx.xxx.218 SIP Status: 407 Proxy Authentication Required No. Time Source Destination Protocol Info 3 0.000332 89.xxx.xxx.218 89.xxx.xxx.218 SIP Request: ACK sip:firstname.lastname@yyyy.zzz i can see several INVITE-407-ACK but no INVITE with proxy-authorization from controller(a)yyyy.zzz have i forgotten to configure something. thx in advance

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[Serusers] transfer to a speeddial number
by jorain 15 Dec '06

15 Dec '06

Hi all, How to get the referred-by value from sip message header? I need it to transfer the call by dialing speeddial number which was stored in speeddial table. Because it will take the original caller (first caller) as the username instead of the second party who did the transfer. For example, speeddial table: ----------------------------------------------------------- username sd_number new_uri 55555555 123 55555550(a)205.5.5.5 33333333 123 33333330(a)205.5.5.5 When i call from 33333333 to 55555555, then 55555555 transfer the call to 123, by right it will call 55555550(a)205.5.5.5 , but now my system call 33333330(a)205.5.5.5 How can i solve it? Thanks. Regards, jorain

1 0

[Serusers] SER + Asterisk - Help needed please
by Mr Raams 15 Dec '06

15 Dec '06

Hello All, I am having problems with my SER + Asterisk combination. I have SER and Asterisk working well on their own, however my issue is as follows. I have asterisk running in realtime with a users table of values accountcode & secret for its authentication. I have the SER subscriber table as normal. I want to have the same SIPID in both tables and the secret=password. However when I register to SER as the SIPID and with the password register is ok, local calls are ok, but when I call asterisk it wont go, it times out after some time. -Note: This is with the password set in the asterisk table as secret (remember this works fine if i register direct to asterisk) The funny part is if I remove the secret (password) from the asterisk account, leaving only the accountcode, but leave the username & password in SER the PSTN call works fine. SER passes the call to Asterisk, and Asterisk will terminate the call. I really need SER to pass the correct authentication details on to Asterisk. I have attached the ser.cfg for you to view and comment. Thank you, Send instant messages to your online friends http://au.messenger.yahoo.com

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[Serusers] click to dial
by jorain 15 Dec '06

15 Dec '06

Hi all, Is it possible to enable the conference using click-to-dial script? Thanks Regards, jorain

1 0

[Users] Duplicate INVITEs
by Mahesh Paolini-Subramanya 14 Dec '06

14 Dec '06

We're seeing sporadic situations where fones a) send an INVITE b) ignore the 100-Trying response, and instead c) send another INVITE with the same sequence number. The question is, What should happen here? Currently, we send back a 500 response (server error). Is this correct? (i think not, cause it invariably causes the fone to go fast busy). Is this some other response that should occur? poring over 3261 resulted in a headache, and no additional clarity... -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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[Users] Call based log
by ibrahim tunali 14 Dec '06

14 Dec '06

Hi everone, I want to trace previous or completed SIP dialogs so i need to keep log every sip activate to separate log files which name is <sip_callid>.log . Do you suggest any how to or tool for this issue. Best regards, ibrahim tunali

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[Users] DNS queries and TLS
by Papadopoulos Georgios 14 Dec '06

14 Dec '06

Hello, I am having some performance issues with Openser and I tend to believe they are related with DNS. So I am trying to figure out when and why Openser is doing DNS queries. Doing ngrep on port 53 I realized that right before sending out the "100 trying -- your call is important to us" message, Openser is doing a reverse DNS lookup for the IP where the INVITE came from. So the sequence is something like: client Openser DNS |---INVITE------------------------>| |<---407 Proxy Auth Reqd---| |---ACK--------------------------->| |---INVITE------------------------>| |---client IP?-------->| |<-------------------------| |<--------------------100 trying---| I am using Openser-1.1.0-notls and in my script I have dns=no rev_dns=no So first question is whether this DNS query is necessary and how I could avoid it. What is furthermore confusing is that I have a test system with the same Openser version and same script, where this DNS query is not happening. Looking into the production system I found the following: ser2:/usr/local/openser-1.1.0-notls/sbin# ldd openser linux-gate.so.1 => (0xffffe000) libdl.so.2 => /lib/tls/libdl.so.2 (0x55571000) libresolv.so.2 => /lib/tls/libresolv.so.2 (0x55574000) libc.so.6 => /lib/tls/libc.so.6 (0x55587000) /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x55555000) whereas the test system shows: sertest:/usr/local/openser-1.1.0-notls/sbin# ldd openser libdl.so.2 => /lib/libdl.so.2 (0x7002c000) libresolv.so.2 => /lib/libresolv.so.2 (0x70040000) libc.so.6 => /lib/libc.so.6 (0x70064000) /lib/ld-linux.so.2 (0x70000000) So the two systems link to different libresolv.so libraries. Is the tls/libresolve.so that is responsible for the DNS query? Given that in both cases I am using the notls version of Openser 1.1, why is there a difference between the two? thank you for any help George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[Users] Request for Discussion: MI and FIFO
by Bogdan-Andrei Iancu 14 Dec '06

14 Dec '06

Hi everybody, with the adding of asynchronous reply feature in MI and MI_FIFO implementation, the transition to MI was completed - all former FIFO functions (old implementation) were ported to MI. Also all FIFO capabilities were also ported to the new MI_FIFO module. so, currently there are two FIFO implementations: - the old one, directly in core. - the new one, based on MI interface. the old implementation is obsolete (all external management will be done via MI) , so, at some point, it will be removed from OpenSER in order to maintain consistency and avoid confusion between the two implementations (as they are equivalent from user point of view - at the moment, at least). my question is when to drop the old implementation? my personal preference will be do doing ASAP as : 1) no functionality is lost 2) avoid confusion among users and the explanation overhead 3) it will force the testing of the new FIFO and MI implementation, which will shorter the way to a stable form. I would like to know also some other opinions. regards, bogdan

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[Serusers] send IM before operation in next shell script ends...
by Webnutzer 14 Dec '06

14 Dec '06

Hi ! I have an if block in my ser.cfg. in this block at first I send an IM with an "thanks for input", after that a shell script should start with some operations. SER sends the IM, but after the operations in shell script ends... How can I send the IM , before the shell script with operations finishs or SER start with operations in the shell script ? here is the code: if (method == "Message" && search("[a-zA-Z0-9/.+ ]{1,}-*,[a-zA-Z0-9]{1,}")) { sl_send_reply("200", "OK"); exec_msg("cd /etc/ser/;sh zweiteEingabe"); # send the thanks... log(1, "zweiter Schritt\n"); exec_msg("cd /root/workspace/Check_Name/bin/;sh Check_Name_Start.sh"); # other operations log(1, "Daten uebergabe\n"); break; }; Thanks ___________________________________________________________ Der frühe Vogel fängt den Wurm. Hier gelangen Sie zum neuen Yahoo! Mail: http://mail.yahoo.de

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Re: [Users] full SIP message in perl
by vallimamod abdullah 14 Dec '06

14 Dec '06

Hello, On Thu, 14 Dec 2006 09:31:22 +0100, Bastian Friedrich <bastian.friedrich(a)collax.com> wrote: > Hi > > On Wednesday 13 December 2006 20:12, Greg Fausak wrote: > > I can see where the sip message is exported to perl. > > Can perl modify the message and send it back? > > Yes, modification is possible. I did not implement too many functions > though; you have two means of modification: > * Use the one or two I wrote. The most important probably is > rewrite_ruri, just as you asked for. > * Use the functions that are exported by other modules (esp. > the "textops" module) for message modification > > [...] > A question: is it possible to change the sip message using Perl's regular expression power and send it back to openser ? That would be a killer feature as it will give me the ability to make complex routing without developping a new module in c ! -- Regards, -vma .

2 1

[Users] Pike module
by Stefano Capitanio 14 Dec '06

14 Dec '06

Hi, i want to use pike module to avoid brute-force attack on Register. are the values of the parameters expressed in seconds? it seems that there is a problem because i've set: modparam("pike", "sampling_time_unit", 30) modparam("pike", "reqs_density_per_unit", 10) modparam("pike", "remove_latency", 600) and the routing logic is: if (method=="REGISTER") { # Checks brute-force attacks if (!pike_check_req()) { sl_send_reply("403", "Too many attempts!"); break; }; if (!www_authorize("caspur.it", "subscriber")) { www_challenge("caspur.it", "0"); break; }; save("location"); break; }; but when I reach the limit it blocks the requests only for less than 1 minute... where is the mistake?any idea? thanks, Stefano

2 3

[Users] full SIP message in perl
by Klaus Darilion 14 Dec '06

14 Dec '06

Hi! m->getFullHeader returns only the headers. How can I access the whole message (headers + all bodies)? regards klaus -- Klaus Darilion nic.at

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[Serusers] Daily snapshots
by Jan Janak 13 Dec '06

13 Dec '06

Hello, Daily snapshots of SER, SERctl, SERweb, and RTPproxy are now available on the FTP server, for more information see: http://www.iptel.org/daily_snapshots_of_software Jan.

1 0

[Serusers] Voice Mail
by Jesús Terán 13 Dec '06

13 Dec '06

Hello I am using ser with asterisk as voicemal, i have the solution when the number is busy and when the number is not registered, but my question is how can i do when no body answer after 3 rings, because right now the mailbox starts after one minute that the phone is ringing. Thanks in advance Jesús Terán

2 1

[Users] perl problems
by Klaus Darilion 13 Dec '06

13 Dec '06

Hi! I try the new perl module, but openser does not start: openser[28662]: perl: Setting lib path: '/home/darilion/software/openser-cvs/sip-server/modules/perl/lib/perl/OpenSER.pm' openser[28662]: Error loading perl file "/etc/openser/myperl.pl". openser[28662]: init_mod(): Error while initializing module perl my perl related config: 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/perl.so" ... modparam("perl", "filename", "/etc/openser/myperl.pl") modparam("perl", "modpath", "/home/darilion/software/openser-cvs/sip-server/modules/perl/lib/perl/OpenSER.pm") /etc/openser/myperl.pl is identical to modules/perl/doc/samples/messagedump.pl Any hints what the problem is? thanks klaus -- Klaus Darilion nic.at

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[Serusers] XML-RPC - how to make working on 0.10?
by Libor Chocholaty 13 Dec '06

13 Dec '06

Hi all, I am struggling with xmlrpc module. It is loaded already but ser reports that do not know create_via() during parsing ser.cfg. Could anybody show me the right way to configure xmlrpc.so properly? BR, Libor

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[Serusers] About rewriteuser()
by chungyu 13 Dec '06

13 Dec '06

Hi I have a field called "phone_number" in subscriber,and would append to rewriteuser(). But rewriteuser() type is string. So have a problem. problem: I used AVpops. modparam("avpops","avp_aliases","phone=i:660") modparam("avpops","db_scheme","phone_scheme:table=subscriber;value_col=phone_number;value_type=string") avp_db_load("$ruri","$phone/$phone_scheme"); rewriteuser("$phone"); result is error. because rewriteuser(string). and i want if value of "$phone" is 4761 the result is 4761@domain How should I use rewriteuser() Thank you!! Chungyu 2006/12/13

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[Users] OpenSER.pm perl library installation
by Klaus Darilion 13 Dec '06

13 Dec '06

Hi! I wonder how the OpenSER.pm is handled during installation. Is it copied into a certain directory (e.g. /usr/lib/openser/modules/) or do I have to do it manually? regards klaus -- Klaus Darilion nic.at

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[Users] Re: check if an AVP is blank
by Daniel-Constantin Mierla 13 Dec '06

13 Dec '06

Hello, see is_avp_set() from avpops modules: http://www.openser.org/docs/modules/1.2.x/avpops.html#AEN594 Cheers, Daniel PS. please start a new thread on mailing list if you have questions not related to the subject of the message you reply On 12/13/06 14:29, Fernando Freitas(Brastel.com.br) wrote: > Hi, > > I need to check if an AVP is blank, how can I do it? > > This is what I have in my script: > > avp_write("tU","$avp(tmp)"); > > > I need to check if the content of this tmp AVP is blank, how to check this?? > > What I need is to check if the TO Username from SIP URI is blank if so, I > need to push the To URI into the To URI Username. > > Thanks, > > Fernando > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

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[Users] Tutorial: OpenSER modular installation
by Daniel-Constantin Mierla 13 Dec '06

13 Dec '06

Hello, this might be useful for many people around, and the input from others would improve the content ... On the OpenSER Web Forum was posted a small tutorial about OpenSER modular installation, courtesy of Alistair Doswald: http://www.voipuser.org/forum_topic_8040.html Cheers, Daniel

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[Serusers] Starting SER in fork=no mode.
by Atle Samuelsen 13 Dec '06

13 Dec '06

Hi Guys, I was wondering if there is a technical reason why ser does not startup in fork=no mode if you only listen to TCP. - Atle

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[Users] Question about dispatcher module
by Jose Gil Navarrete 13 Dec '06

13 Dec '06

Hi, I try to configure openser with the dispatcher module like only ponit of contact and load balancer, it is: Register UAC ---Register--> LB ---Register--> proxy UAC <--200 ok----- LB <--200 ok----- proxy INVITE UAC1 ---Invite--> LB ---Invite--> proxy UAC2 <--Invite--- LB <--Invite--- proxy My load balancer works ok, my question is: May I have a problem with the transactions in the load balacer? In my proxy i use openser, then in my openser.cfg I make t_relay("IPLoadBalancer:5060") and the openser.cfg in LB: if ( !is_method("REGISTER") ){ record_route(); }; if (loose_route()) { if (search("Path")) { route(1); }; if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); exit; }; route(1); }; if (!uri==myself) { route(1); } if ( !is_method("REGISTER") ){ if (search("Path")) { route(1); }; if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); }; }; ds_select_domain("2", "0"); t_on_failure("1"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } Thanks in advance. --------------------------------- 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] Resource List Server/XCAP Clients
by Reinhold Buchinger 13 Dec '06

13 Dec '06

Hi, I try to build a presence infrastructure based on OpenSER. Do you know any free RLS and XCAP Clients I can use for this purpose? I hope this is not too much off-topic. Thanks in advance! regards, Reinhold

2 1

[Serusers] IP Based Auth
by Arun Kumar 13 Dec '06

13 Dec '06

HI, I've running ser with mysql and radius. My user are stored in mysql db. my users are able to do the auth againg radius. I'm looking how to setup ip based auth. any kind of doc or link. thanks arun

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[Serusers] reinvite problem...
by Natambu Obleton 12 Dec '06

12 Dec '06

Ok. I have SER working with mediaproxy and it rewrites the requests to goto the Mediaproxy server as it should, but after a call has been up for a bit.. it gets another invite from my PSTN gateway and when this invite comes through it doesnt rewrite the media addresses why? My config: http://laplata.fasttrackcomm.net/ser_cfg.txthttp://laplata.fasttrackcomm.net /ser_cfg.txt <http://laplata.fasttrackcomm.net/ser_cfg.txthttp:/laplata.fasttrackcomm.net /ser_cfg.txt> My transaction log: http://laplata.fasttrackcomm.net/translog.txt About ¾s of the way down you will see where 206.123.213.84 sends another invite for the call and SER doesnt change the media information on it. Thanks for your help. Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

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[Users] New module: perl
by Bastian Friedrich 12 Dec '06

12 Dec '06

Hi everybody, since last friday, OpenSER HEAD has a new module, called perl. As the name suggests, it incorporates a Perl interpreter within OpenSER. What can you do with it? ======================== The perl module lets you request information from the SIP message currently active, lets you make decisions based on that information and finally modify the SIP message. In other words, you can do almost everything you could have done in the OpenSER configuration file, plus much, much more. How do I use it? ================ The perl module should be automatically compiled in current CVS checkouts of the HEAD branch, as long as you have a fairly recent Perl version (including possibly necessary development packages) installed on your system. Load it in your OpenSER configuration just like all the other modules. Set the two module parameters "filename" and "modpath"; the first one points to your perl script (samples are included in the doc directory), the second points to the path where the Perl packages reside (they currently are not "installed" anywhere automatically. Feel free to do that yourself). In the configuration file, you may then call the functions defined in your Perl script with the "perl_exec" and "perl_exec_simple" functions. The samples and the README file should tell you more about it. What should the Perl script look like? ====================================== Please take a look at the doc/samples directory located in the perl module directory. They should give you an introduction of how such a script could look like. The full API is described in the README file. The module documentation on the OpenSER website is not 100% apt as of now, but should be updated during the next days. Great! And now? =============== Please feel free to test the perl module. Although it currently runs fine for me, deployment in production environments is not recommended. Tell me about your experiences and any bugs you found. If you write a test script, I'd be thankfull if you send it to me for inclusion as a sample. If you have any questions or remarks, please send a mail to the users or developer mailing lists (depending on the focus), or directly to me. Thx for your attention, Regards, Bastian -- Collax GmbH . Burkheimer Straße 3 . 79111 Freiburg . Germany p: +49 (0) 761-45684-24 f: +49 (0) 761-45684-10 www.collax.com \ Things you never wanted to know: \ In Minnesota it is illegal to cross state lines with a duck on your \ head.

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[Users] Presence Publish/Subscribe Timeout
by christian l 12 Dec '06

12 Dec '06

Hello, we want to use the presence module of openser in combination with eyebeam pa mode. In order to save bandwith, we want the clients to subscribe in an intervall of 1 hour. At the same time, we want to detect almost instantly when I client timeouts, i.e. has a network failure. Is there any way I can detect client timeouts rather fast other than reducing the max_expires value, which also affects the SUBSCRIBE timeout and therefore increasing the network utiliztion again? openser version is 1.2.0-dev12-notls , eyebeam version is 1.5.10.2 build 33793 Thanks in advance for your help Schnell und einfach ohne Anschlusswechsel zur Lycos DSL Flatrate wechseln und 3 Monate kostenlos ab effektiven 5,21 EUR pro Monat im ersten Jahr surfen. http://www.lycos.de/startseite/online/dsl/index.html?prod=DSL&trackingID=em…

1 0

[Users] sent notify after registration
by Pavel A. Siderov 12 Dec '06

12 Dec '06

Hi guys, I'm trying to generate (read data from custom database) and send a notify message to user agent when it registers and there is a voicemail left. I don't want to use an external script due to some reasons. I had a look on the msilo and uac modules but they don't seem to be suitable for the purpose. Is it possible with some of the modules or currently I need to use an external script? Thanks in advance, Pavel

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[Serusers] ser 0.9.6 # mediaproxy # one-way audio
by davor jovanovic 12 Dec '06

12 Dec '06

Hi all, I’m using ser 0.9.6 with mediaproxy and latest version of XLite client. I have a problem with one-way audio when calling client on another domain (not served by our SER). When Xlite uses public address (not behind NAT) I can't receive audio from other side, but when I connect that same XLite client by way of dial-up or ADSL everything is OK. I suppose that in second case (ADSL) mediaproxy is included in the process and that is way call succeeds. I don't know what is a problem when client isn't behind NAT, but it has public IP address. All combinations of calls are working with no problem when all users are using only our SER. I'm using configuration similar to that in SER Getting started (NAT part is same), but if needed I can email it. If anyone had similar problem, please help, a would appreciated it! Thanks /davor

1 1

[Users] Problem with $dlg_count
by Leonid Fainshtein 12 Dec '06

12 Dec '06

Thank you, Daniel. Your recommendation helped. Leonid ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine %tn%

1 0

[Serusers] Idea for concurrent calls statistics?
by Tobias Lindgren 12 Dec '06

12 Dec '06

Hi all, I'd like to somehow count concurrent outgoing calls through my SER from my customers, based on the source ip-address in the SIP-request. I'm running with the TM-module. Does anyone have an idea on how this could be done? Br, Tobias

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[Users] openserctl fifo
by Klaus Darilion 12 Dec '06

12 Dec '06

Hi! I can nowhere found the setting for the fifo use inside openserctl. The default is /tmp/openser_fifo - where can I change this? thanks klaus -- Klaus Darilion nic.at

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Re: [Serusers] Ser accounting question?
by Greger V. Teigre 12 Dec '06

12 Dec '06

Copy the list. Sending private emails just make them add to the end of my big inbox. I don't know where the service type comes from. That's where you need to define it. g-) Lokesh Kumar wrote: > > Where I can define service type? If this not from ser then must be > conflicting from radius dictionary. Can you suggest something.? > > > > Thanks > > > > Lokesh > > > > ------------------------------------------------------------------------ > > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* Thursday, December 07, 2006 10:11 AM > *To:* Lokesh Kumar > *Cc:* serusers(a)iptel.org > *Subject:* Re: [Serusers] Ser accounting question? > > > > The service-type is set by the sender of the accounting packet. I'm > not aware of such a service-type in ser. > g-) > > Lokesh Kumar wrote: > > Hello List, > > > > Below mentioned is my accounting packet, I would like to know what is > IAPP-AP-Check, what this check do , and where its configuration > present. Can I change Service-Type ? > > > > Wed Dec 6 20:08:23 2006 > > Acct-Status-Type = Stop > > Service-Type = IAPP-AP-Check > > Sip-Response-Code = 200 > > Sip-Method = 8 > > User-Name = "69021(a)xxx.yyy.pt" <mailto:69021@xxx.yyy.pt> > > Calling-Station-Id = "sip:69021@xxx.yyy.pt" > <mailto:sip:69021@xxx.yyy.pt> > > Called-Station-Id = "sip:30612@xxx.yyy.pt" > <mailto:sip:30612@xxx.yyy.pt> > > Sip-Translated-Request-URI = "sip:30612@212.13.xx.yy:5060" > <mailto:sip:30612@212.13.xx.yy:5060> > > Acct-Session-Id = > "D4808CE2-1BA9-4572-B82B-5A0BCF6AB7CB(a)192.168.0.47" > <mailto:D4808CE2-1BA9-4572-B82B-5A0BCF6AB7CB@192.168.0.47> > > Sip-To-Tag = "86949D74-2699" > > Sip-From-Tag = "67434843439" > > Sip-Cseq = "3" > > NAS-Port = 5060 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 127.0.0.1 > > Client-IP-Address = 127.0.0.1 > > Acct-Unique-Session-Id = "c6adb5da735b1150" > > Timestamp = 1165435703 > > > > Thanks > > > > Lokesh > > > > -- > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.1.409 / Virus Database: 268.15.6/567 - Release Date: 12/4/2006 > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.1.409 / Virus Database: 268.15.6/567 - Release Date: 12/4/2006 > > > -- > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.1.409 / Virus Database: 268.15.6/567 - Release Date: 12/4/2006 >

3 3

Re: [Users] SIP message being sent twice
by Klaus Darilion 12 Dec '06

12 Dec '06

Fernando Freitas wrote: > Hi all, > > I am having a situation that every SIP message is being sent twice when > reaching openser. It seems that the message when hits proxy it is processed > twice. I double checked my cfg file, but didn't find anything. Please help > me. Maybe you add another branch in your openser config. Are they really the same (also the branch parameter in the Via header)? regards klaus > > Thanks, > > Fernando Freitas. > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- Klaus Darilion nic.at

1 0

Re: Re: [Serusers] SIPSAK - wrong behaviou r??
by tzieleniewski 12 Dec '06

12 Dec '06

yes you are right:) but the stange situation is the following (version from svn): UA1 SIPProxy | | |INVITE 3 | |------------->| | | |407 3 | |<-------------| | | |ACK 3 | |------------->| | | |INVITE 4 | |------------->| | | |TRYING 4 | |<-------------| | | |INVITE 4 | |<-------------| | | |ACK 3 | |------------->| | | |INVITE 4 | |<-------------| | | after the end of dialog related to cseq 3 ua (sipsak) sends an invite (with cseq 4) to sipproxy then there is the provisional response trying and sipproxy sends invite further (actually the same ua) to sipsak and the reaction of sipsak is: ACK with with cseq 3 instead of OK or 4xx, this scenarion is repeated until sipproxy sends the 408 timeout with cseq 4 after wich ua (sipsak) resposes with ACK with cseq 4 what do you think about this?:) bests tomek ---- Wiadomość Oryginalna ---- Od: Michal Matyska <michal(a)iptel.org> Do: serusers(a)lists.iptel.org Data: 11 grudnia 2006 20:59 Temat: Re: [Serusers] SIPSAK - wrong behaviour?? > I was just curious, so I took look on the ngrep.... > > there is short jump in the description between points 2 and 3.... > 2a. sipsak sends ACK upon 407 reply, which is 100% correct > 2b. sipsak resends INVITE with correct credentials > 2c. ser forwards the INVITE to the sipsak @ port 5060 > 3. sipsak sends ACK (with the CSeq: 3, Via: headers look like > combination of original ACK and passed INVITE Vias), which you've found > inappropriate, right? Just to make sure we talk about the same ACK :-D > > Michal > > On Mon, 2006-12-11 at 19:12 +0100, Nils Ohlmeier wrote: > > Hi Tomasz, > > > > On Monday 11 December 2006 17:21, tzieleniewski wrote: > > > Hi ALL! > > > I am trying to run some tests under sipsak but I obtain strange result > > > I invoke sip stack with the following command: > > > sipsak -IU -vv -s sip:sipsak@scscf.ims.touk.pl -a sipsak -l 5061 -H > > > 192.168.0.129 The registration proces runs without any problem, user > > > registers and then sipsak sends invite to ser. And here comes the strange > > > thing > > > 1. sipsak sends invite to ser: > > > 2. Ser sends the proxy authentication required response 407 (from ngrep) > > > 3. sipsak receives the message andresponses with the ACK???? > > > Ive placed the ngrep from the sipsak socket in the attached file. > > > > that looks like a bug. I'll try to look into that later. > > It would be great if you could try if the same behavior is still present in > > the latest version from subversion (let me know if I should send you a > > tarball of that). And the output from sipsak with -vvv would be a great help. > > Just send that directly to me. > > > > Thanks > > Nils > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] SIP message being sent twice
by Fernando Freitas 12 Dec '06

12 Dec '06

Hi all, I am having a situation that every SIP message is being sent twice when reaching openser. It seems that the message when hits proxy it is processed twice. I double checked my cfg file, but didn't find anything. Please help me. Thanks, Fernando Freitas.

1 0

[Serusers] Monitoring
by Peter Kuebler 11 Dec '06

11 Dec '06

Hello Hendrik, I was reading your post and wondering if you have made any progress on SER monitoring? Especially in regard to enhancing the snmpd to monitor the number of users and transactions? http://lists.iptel.org/pipermail/serusers/2006-January/026771.html Hello Hendrik, What about VRRP for failover ? SER1 (master) sip agents == || === Asterisk Farm==sip/pstn gateway SER2 (slave) Harry --- Hendrik Scholz <hendrik.scholz at freenet-ag.de <http://lists.iptel.org/mailman/listinfo/serusers> > a écrit : > Hi! > > hgaillac-sip at yahoo.fr <http://lists.iptel.org/mailman/listinfo/serusers> wrote: > > Yes I know about snmp module. > > There is no way to monitor SER vi snmp !? > > As others have noted SER does not directly support > SNMP but you can > easily enhance your snmpd to support it. > You add serctl fifo commands using the snmp exec > directive > to monitor the number of users, transactions stuff, > ... > The meminfo patch (now included) is handy and you > can directly pipe > serctl fifo meminfo into SNMP and create nifty > graphs. > > sipsak comes in handy, i.e. sending an OPTIONS > request and measuring > a) if a response comes back and b) the time > processing took. > > Much better would be monitoring INVITE/REGISTER > response behaviour since > that usually requires a whole lot more interaction > (radius, db, ..) > on SER side. > > Hendrik > Peter Kuebler Director Network Reliability tele: 703-636-1114 ext: 112 tele: 703-734-4112(alt) cell: 240-506-5703 SunRocket 8045 Leesburg Pike, Suite 300 Vienna, VA 22182 peter.kuebler(a)sunrocket.com SunRocket NOC 703-637-9970

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[Users] problem with forking...
by Frogger 11 Dec '06

11 Dec '06

I am seeing a problem forking more than 7. Any phones that are beyond the 7th device get 180 ringin but Openser cannot receive the invite to answer the call. Any thoughts on why I cannot increase the children? (BTW, "openserctl ps" shows the proper number of children. For example, when set to 14, "ps" shows 14 children. However any UA beyond 7 cannot answer the forked call.) Thanks, MC ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

2 2

[Users] acc module for syslog
by Daniel Burge 11 Dec '06

11 Dec '06

Hello, Does anyone have a working openser.cfg file that is using the acc module to log transactions to syslog? If so, can you post it here or email it to me? Thanks in advance, Daniel

2 1

[Serusers] TM module error. Please help!!
by Meidiana Ten 11 Dec '06

11 Dec '06

Hi there, I have problem in loading module TM for voicemail setup here. I use SER 0.9.7 as server and the latest SEMS for voicemail usage. I tried to add my ser.cfg with script that can handle the voicemail but I found errors when run SER. Below is my ser.cfg: (Please take a look at line 54 and 56) debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("tm", "fr_timer", 10 ) modparam("tm", "wt_timer", 10 ) line 54: modparam("tm", "pass_provisional_replies", 1) # configure tm to append this when tw_appent voicemail_headers is used line 56: modparam("tm", "tw_append","voicemail_headers:P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[Content-Type];msg[body]") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) # configure avpops db connection modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) modparam( "avpops", "avp_table", "subscriber" ) modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database modparam( "avpops", "db_scheme", "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam( "avpops", "db_scheme", # "language_scheme:table=subscriber;value_col=language;value_type=string") alias="pcr.ac.id" route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; /* if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); break; } */ # make transaction if (!t_newtran()){ log("could not create transaction\n"); sl_send_reply("500","could not create transaction"); break; } # actively absorb ACKs if (method == "ACK") { t_relay(); break; } # pass INFO to SEMS if (method=="INFO") { if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ log("could not contact sems\n"); t_reply("500","could not contact media server"); } } if (uri =~ "sip:101.*@") { if (!t_write_unix("/tmp/sems_sock","myapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:102.*@") { if (!t_write_unix("/tmp/sems_sock","myconfigurableapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:103.*@") { if (!t_write_unix("/tmp/sems_sock","myannounceapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:104.*@") { if (!t_write_unix("/tmp/sems_sock","myjukebox")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:105.*@") { if (!t_write_unix("/tmp/sems_sock","mycc")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:106.*@") { if (!t_write_unix("/tmp/sems_sock","ivr_announce")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } ############ default sems apps if (uri =~ "sip:110.*@") { if (!t_write_unix("/tmp/sems_sock","echo")){ log("could not contact echo\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:111.*@") { if (!t_write_unix("/tmp/sems_sock","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:112.*@") { if (!t_write_unix("/tmp/sems_sock","conference")){ log("could not contact conference\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:113.*@") { if (!t_write_unix("/tmp/sems_sock","mailbox")){ log("could not contact mailbox\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:114.*@") { if (!t_write_unix("/tmp/sems_sock","early_announce")){ log("could not contact early_announce\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("root@localhost","$email"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail/voicemail_headers")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); break; } break; } t_reply("404","Not found"); # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Offline Message Store Section # ----------------------------------------------------------------- if (is_from_local()) { if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); 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("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted() && nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.2:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } -------------------------------------------------------------------------------------------------------------------- If I uncomment line 54 and 56, below is the output when I run SER: 0(4425) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(4425) parse error (54,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm 0(4425) set_mod_param_regex: found <tw_append> in module tm [/usr/local/lib/ser/modules/tm.so] 0(4425) ERROR:tm:parse_tw_append: bad alias spec <$email> 0(4425) parse error (56,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm ----------------------------------------------------------------------------------------------------------------------- If I comment line 54 and 56, below is the output: 0(0) ERROR:tm:fixup_t_write: unknown append name <voicemail_headers> 0(0) ERROR: fix_expr : fix_actions error ERROR: error -6 while trying to fix configuration 0(0) MSILO: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5050) shm_mem_destroy 0(5050) destroying the shared memory lock ----------------------------------------------------------------------------------------------------------- Please tell me what's wrong. And please tell me whether if my configuration on ser.cfg is right or wrong because I doubt it. Thanx before... Regards, Meidiana --------------------------------- Want to start your own business? Learn how on Yahoo! Small Business.

2 2

[Users] Problem with $dlg_count
by Leonid Fainshtein 11 Dec '06

11 Dec '06

Hi, I'm trying to use the exported variable $dlg_count (number of active dialogs) in the configuration script but I get nothing. In the syslog I see the following error message: "xl_get_spec_value: error - null sp->itf". I'm using stable 1.1.0 version. Any suggestions?

2 1

[Serusers] Record a call from SIP Server (SER)
by Matias Rossi 11 Dec '06

11 Dec '06

Hello, my name is Matias Rossi and I want to know if it`s possible to record a call from two "telephones IPs" . So if I have a SER and a call is established I want to record it conversation with the SER, is that possible?. Thanks. Matias. -- Rossi Matias Javier ~ Insfostyle Cel : 15-6397-9708 Nextel: 579*391 Tel./fax : (54) 11 4573-4013 E-mail: matiasrossi(a)infostyle.com.ar Web: www.infostyle.com.ar

3 2

Re: [Serusers] any kind of statistics available
by Jiri Kuthan 11 Dec '06

11 Dec '06

I'm not aware of such publicly available tool. We are using our own to run iptel.org. (snapshot attached). I think the key point is that the management data is available to outside world via xml-rpc and folks can relatively easily bind this data via xml-rpc to their favorite monitoring apps. If it is as easy as availability monitoring, you can use nagios with sipsak as plug-in. -jiri p.s. this is ser mailing list, which is not openser mailing list. openser is quite different piece of software. At 13:28 11/12/2006, Wienand, Joerg wrote: >Hello, > >is there any kind of statistic-support available for OpenSER? >I would like to have statistics about number of calls, time >of calls, active users, ... > >Is there any tool available, which creates statistics. Pictures >and diagrams would be nice to have. > > >best regrads > >Jörg >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] lookup() and forking
by O'Shaughnessy Evans 11 Dec '06

11 Dec '06

Hello. I'm trying to debug a problem I seem to be having with parallel forking and I'd like to check my understanding of ser's behavior on a few points. I have 2 phone numbers registered in the location table for a certain subscriber, each with a q value of 1. When I call the subscriber number, I don't see ser forking and then sending an INVITE to each contact, but only the one with the lower expiration value (the first number that "serctl ul show" lists). I have append_branches set to 1 and desc_time_order set to 0. The way I've understood everything I've read, lookup() should be laying the groundwork for the tm module to fork ser so that when I call t_relay ser will intiate one INVITE to each location with the same q value. Is this not correct? Is there anything special I should be doing to tell it to fork an INVITE to each contact at this point? Thanks, and best regards. -- = o'shaughnessy_evans -=- sys_admin @ (aloha|ilhawaii|myworld|turquoise).net = Pacific LightNet, Inc. "Beware the lollipop of mediocrity. Once you lick it you will suck forever!" -- Capt. Gladys Stotephamphlet

2 1

[Serusers] Subscriber table columns phplib_id, domn & uuid usage
by Kamal.Mann＠t-systems.com 11 Dec '06

11 Dec '06

Hi All I need to know the usage of 'phplib_id, domn & uudi' columns into ser database table subscriber. Please give me your valuable suggestion for the same. Thanks in anticipation Kamal Mann

3 2

[Serusers] chain of rtp proxies ...
by Cesc 11 Dec '06

11 Dec '06

Hi, I am currently testing a system which connects two "islands" ... each island has its own SER (0.9.6) + rtp proxy ... Actually there is NOT going to be NAT or firewall among them, but i still do want to proxy all RTP streams (sorry ... requirements). Probably even within the phones in the same island ... I have a few questions ... - i saw mentioned that chaining is possible ... no problem there right? i need to send an extra parameter to the force_rtp_proxy and that is it? no side-effects if, i.e, call between the phones in the same island (thus, just one rtp proxy)? - What would be the bare minimum config to force rtpproxy, without all the NAT tests? as said, i have no such problem, so i would like a minimal config: fix_nated_sdp and force_rtp_proxy? is that enough? - For testing purposes in my little lab ... can i run two SERs on the same box (that i know i can do :D ) ... which use the same rtpproxy on the box, thus via the same unix socket? just for testing ... they put me in a corner of an office with just a little table :) Regards, Cesc

5 8

[Serusers] Error while loading TM module. Please help!!
by Meidiana Ten 11 Dec '06

11 Dec '06

Hi there, I have problem in loading module TM for voicemail setup here. I use SER 0.9.7 as server and the latest SEMS for voicemail usage. I tried to add my ser.cfg with script that can handle the voicemail but I found errors when run SER. Below is my ser.cfg: (Please take a look at line 54 and 56) debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("tm", "fr_timer", 10 ) modparam("tm", "wt_timer", 10 ) line 54: modparam("tm", "pass_provisional_replies", 1) # configure tm to append this when tw_appent voicemail_headers is used line 56: modparam("tm", "tw_append","voicemail_headers:P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[Content-Type];msg[body]") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) # configure avpops db connection modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) modparam( "avpops", "avp_table", "subscriber" ) modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database modparam( "avpops", "db_scheme", "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam( "avpops", "db_scheme", # "language_scheme:table=subscriber;value_col=language;value_type=string") ----- ----- -------------------------------------------------------------------------------------------------------------------- If I uncomment line 54 and 56, below is the output when I run SER: 0(4425) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(4425) parse error (54,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm 0(4425) set_mod_param_regex: found <tw_append> in module tm [/usr/local/lib/ser/modules/tm.so] 0(4425) ERROR:tm:parse_tw_append: bad alias spec <$email> 0(4425) parse error (56,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm ----------------------------------------------------------------------------------------------------------------------- If I comment line 54 and 56, below is the output: 0(0) ERROR:tm:fixup_t_write: unknown append name <voicemail_headers> 0(0) ERROR: fix_expr : fix_actions error ERROR: error -6 while trying to fix configuration 0(0) MSILO: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5050) shm_mem_destroy 0(5050) destroying the shared memory lock ----------------------------------------------------------------------------------------------------------- Please tell me what's wrong. And please tell me whether if my configuration on ser.cfg is right or wrong because I doubt it. Thanx before... Regards, Meidiana --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.

1 0

[Users] How to communicated Both SIP and IAX2 each other ?
by raviprakash sunkara 11 Dec '06

11 Dec '06

Hello Users.. Is it possible to do. one UA is SIP and other UA is IAX2, UA(sip)----------->OpenSER------> Asterisk------> UA(IAX2) . UA(IAX2) ------- >Asterisk --- > OpenSER ------ > UA (SIP ). other wise we can like that.. UA(SIP ) ----------- > Asterisk--------->UA(IAX2) But SIP message and IAX messages are different , Then How can we communicate the both SIP and IAX2 -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 1

[Users] Mediant 2000 and SER, caller ID issue
by Nick De Cristofaro 11 Dec '06

11 Dec '06

Hello all, Trying to debug a problem in regards to Caller IDs not being showed through our PSTN gateway. Not sure if this is an issue with the ser or with the mediant 2000 gateway or somewhere else but any help to this issue would be great. I am new to this and we are trying to solve this problem, basically one of our boxes isn't providing caller ID to the PSTN network when users are calling. Basically they will show up as "private number" when calls are initiated from inside and terminating on the PSTN networking, when they are terminating inside the ser itself the caller ID shows up with no problem, but not on the outside. We wish to have the ability to show private numbers and to show public numbers, but as far as it goes right now everyone is supposed to have public numbers. Where might the problem be? Any help would be great Thanks Nick

4 3

Re: [Serusers] chain of rtp proxies ...
by Jiri Kuthan 11 Dec '06

11 Dec '06

At 18:46 09/12/2006, Cesc wrote: >A final question ... basically thinking out loud (and writing it down) :) >I read that rtpproxy won't start relaying until it got an rtp packet >from both sides ... is it true? yes >could this not cause problems, >specially with chained rtpproxies, if say, i have one of the phones >not sending rtp packets (say, it starts muted ... muted means no rtp >packets)? indeed but that's it. MS Windows Messenger does that for example with voice inactivity detection. You have to speak to hear. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

2 1

[Users] mon script for openser ?
by G.Jacobsen 10 Dec '06

10 Dec '06

Hello, I intend to test openser health periodically with a (perl) script to see whether it is answering SIP requests - to be used for mon in combination with heartbeat. If anyone has done such thing I would be grateful for any code snippets. I suppose as another possibility one could use one of the Nagios sipsak scripts and modify that for mon ? Gerry

1 0

Re: [Serusers] account forwarded calls
by Mobashir Ahmed 10 Dec '06

10 Dec '06

this has always been a problem with Snom Sip Proxy. we are looking for a solution on this past many months. Please hlep.!! Regards, Mobashir Ahmed. ----- Original Message ---- From: Rosario Pingaro <rpingar(a)nesec.it> To: serusers(a)lists.iptel.org Sent: Sunday, December 10, 2006 2:46:20 PM Subject: [Serusers] account forwarded calls I implememented call forwarding using the exemple found on the onsip.org website. It is working fine but I have an accounting problem. Infact the forwared call is not accounted at all. This means that the accounting modle records the call as it is in the beginning ant not the second leg. Is there a way tha have accounted the forwarded call as originated from the forwader to bill him? Regards Rosario _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com

1 0

[Serusers] account forwarded calls
by Rosario Pingaro 10 Dec '06

10 Dec '06

I implememented call forwarding using the exemple found on the onsip.org website. It is working fine but I have an accounting problem. Infact the forwared call is not accounted at all. This means that the accounting modle records the call as it is in the beginning ant not the second leg. Is there a way tha have accounted the forwarded call as originated from the forwader to bill him? Regards Rosario

1 0

[Serusers] CDRs and NAT Issues
by Mobashir Ahmed 09 Dec '06

09 Dec '06

Hello All, Since the time i joined, i have been trying all these days to get all mails that i missed. Anyways, i have 2 questions at this point of time, 1. Whats the location where the CDRs are situated for SipX. and how are they saved. 2. Can we use somehow SipX as a Proxy along with the Nat Traversal. what other additions are required for it. Thanks and Regards, Mobashir Ahmed. ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

1 0

Re: [Users] multiple BYEs getting accounted
by Jayesh Nambiar 09 Dec '06

09 Dec '06

Hi Bogdan, Thanks for your response. Actually I have setflag(1) at two places in my script. Its like : if(method=="BYE") { setflag(1); }; if(loose_route()) { ---- ---- }; setflag(1); Actually I added the first condition to make sure that BYEs are not missed. I think that condition is not needed, but then where is the best place to do setflag(1) and ,ake sure that BYEs dont get missed.. Is it before loose_route condition as I have my loose_route condition in the main route. FYI flag 1 is the flag for acc. Waiting for a reply, Thanks, Jayesh. ----- Original Message ---- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Friday, 8 December, 2006 10:56:20 PM Subject: Re: [Users] multiple BYEs getting accounted Hi Jayesh, normally, acc is done on transaction level, so the retransmission are not accounted. Are you forcing acc from script via functions?? or maybe you have 2 BYEs, from both directions? regards, bogdan Jayesh Nambiar wrote: > Hi all, > I am using mysql acc table for accounting. Sometimes due to some NAT > issues, my openser does not acknowledge the BYE from these NATed > clients to proper port. This causes the UA to retransmit BYEs. > Now the problem is that all the BYEs that were received by my openser > are accounted in the acc table, This causes a single INVITE to have > multiple BYEs. I wanted to avoid this. > How could I limit it in the script itself that > "whenever a BYE is received; log it in acc table, but if it is a > retransmited BYE for any reaason do not log it in acc table". > Basically I need only single BYE for an INVITE to get logged in the > acc table. Is this possible? > If someone has any clues over it, please help me. > > Thanks in advance, > > w/regards, > jayesh > > ------------------------------------------------------------------------ > Find out what India is talking about on - Yahoo! Answers India > <http://us.rd.yahoo.com/mail/in/yanswers/*http://in.answers.yahoo.com/> > Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. > Get it NOW > <http://us.rd.yahoo.com/mail/in/messengertagline/*http://in.messenger.yahoo.…> > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

1 0

[Users] How do manipulate the FROM field form a SIP message
by Ricardo Martins 08 Dec '06

08 Dec '06

Folks, I´m trying to manipulate the FROM field from a sip message and could not find any way to do that. Tryied using AVPOPS but the problem is that the "push_to" function only works on URI (destination) fields. Anyone could help me with a trick? Best regards to you all. Ricardo Martins.

2 1

[Users] multiple BYEs getting accounted
by Jayesh Nambiar 08 Dec '06

08 Dec '06

Hi all, I am using mysql acc table for accounting. Sometimes due to some NAT issues, my openser does not acknowledge the BYE from these NATed clients to proper port. This causes the UA to retransmit BYEs. Now the problem is that all the BYEs that were received by my openser are accounted in the acc table, This causes a single INVITE to have multiple BYEs. I wanted to avoid this. How could I limit it in the script itself that "whenever a BYE is received; log it in acc table, but if it is a retransmited BYE for any reaason do not log it in acc table". Basically I need only single BYE for an INVITE to get logged in the acc table. Is this possible? If someone has any clues over it, please help me. Thanks in advance, w/regards, jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

2 1

[Users] Domain: is_from_local() is case sensitive in caching mode.
by Papadopoulos Georgios 08 Dec '06

08 Dec '06

hello, Back in March I posted the following http://openser.org/pipermail/users/2006-March/003426.html The problem was solved with the patch that Klaus sent http://openser.org/pipermail/users/2006-March/003456.html However in Openser 1.1 it seems that there is still a problem. The following code shows how hash("iptel.org") != hash("ipTel.org") Any patches for this? thank you George #include "hash_func.h" int main(int a, char* arg){ char s1[50] = "iptel.org"; char s2[50] = "ipTel.org"; str str1, str2; str1.s = s1; str1.len = strlen(s1); str2.s = s2; str2.len = strlen(s2); printf("s1=%s s2=%s\n", s1, s2); unsigned int r1 = core_case_hash(&str1, 0, 128); unsigned int r2 = core_case_hash(&str2, 0, 128); printf("r1=%u r2=%u\n", r1, r2); return 0; } Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Users] presence module
by henri keski-sikkila 08 Dec '06

08 Dec '06

Hello, I have some problems with presence module. When I use presence module it uses all available memory until there isn't any memory left. If I take ps or top there are many openser prosess but only one of them uses more and more memory all the time. Best Regards, Henri

4 6

[Serusers] Monitoring SER
by Peter Kuebler 08 Dec '06

08 Dec '06

Hi David, Did you ever get any help with this? We use SER here and monitoring is a difficult task. You may want to reference this link for some assistance- http://lists.iptel.org/pipermail/serusers/2006-January/026771.html Your post below: http://lists.iptel.org/pipermail/serusers/2006-May/028625.html Hi, My company has deployed SER on a very large scale across the US and I would like to develop a way to monitor critical parts of SER and the servers that SER runs on. I have several questions regarding monitoring SER. 1. Can SER be SNMP enabled via something like SMUX or AgentX from net-snmp? 2. Does SER have any native SNMP support or other monitoring/logging hooks built in? 3. Do any mibs exist for SER / SIP? 4. I am also still trying to find out if SER has anyway to detect when a user looses registration either via SER internals or external watchdog scripts etc. Thanks David Peter Kuebler Director Network Reliability tele: 703-636-1114 ext: 112 tele: 703-734-4112(alt) cell: 240-506-5703 SunRocket 8045 Leesburg Pike, Suite 300 Vienna, VA 22182 peter.kuebler(a)sunrocket.com SunRocket NOC 703-637-9970

3 2

[Users] MediaProxy / RTPproxy / SER / Asterisk / HA / LVS / DNS [load-balancing and high availability]
by Thomas Deillon 08 Dec '06

08 Dec '06

Hi, I will try to summarise different load balancing solution and I hope that you will correct my mistake to have a good point of view of all solution. Thanks in advance. 1. Solution with DNS SRV allows making Load Balancing (on phone side) but need phones that support this function. 1.1: Explanation In your DNS, you can set DNS SRV entry like this: ------------------------------------------------------------------ |sipserver1.bigu.edu. 43200 IN A 10.0.0.21 |sipserver2.bigu.edu. 43200 IN A 10.0.0.22 |; |_sip._udp.bigu.edu. 43200 IN SRV 0 0 5060 sipserver1.bigu.edu. |_sip._udp.bigu.edu. 43200 IN SRV 0 0 5060 sipserver2.bigu.edu. ------------------------------------------------------------------- In your phone, you set in proxy SIP: "bigu.edu". The DNS will see that it's a request SIP in UDP and will return 1/2 times in this order: - sipserver1.bigu.edu. - sipserver2.bigu.edu. Phone understanding DNS SRV, will use the first line and if it doesn't work, will use the second one. The phone witch not understands the DNS SRV will always use the first line even if it doesn't work. Conclusion (C/c): It's a load balancing on phone side and so, if you can't choose phone model, it's not a solution (like IP centrex). To solve this problem, we have to use a load balancing on network side OR server side. 2. Solution with HA (Heart Beat) it's a solution on server side. 1.1 Explanation This solution is a fail over architecture. You will set a VIP (virtual IP address) for two servers. The server 1 will have this VIP and handle all the traffic. The second server will listen to the "heart" of the first server and it's something going wrong, it will take the VIP. C/c: This solution will not load balance traffic, and half of computer will not be used. 3. [Correct this part please] LVS (linux virtual Server) is a solution to load balance traffic but it's not SIP aware. You can use it for TCP connection but for SIP, it's very hard. However, you can set a load balancing on IP source, and so, each phone will see only one server. More than this, the LVS solution will not try to know if Asterisk OR SER is alive but try to know if the server is alive. (most of the time, only the service going down, not the whole server ...) C/c: The LVS is not a good solution. This can help but the reactivity is very bad. 4. RTPproxy with SER RTPproxy with Ser is use for failover and not load balancing, so, it's the same conclusion as HA. 5. MediaProxy with SERs. I'm really not sure, but I thing that we have to use only SER servers and the loadbalancer have to be a registrar server ?????? C/c: can you conclude... 6. The ultimate solution... I'm looking for a solution with SER (or something else) in loadbalancer and multiple Asterisk server behind it witch will do all SIP function (REGISTRAR, ...). This kind of architecture has to support NATed phones. Thanks to read this and thanks for your help, Thomas Deillon

4 3

[Users] Softphone and OpenSER
by Kc Ayuba 07 Dec '06

07 Dec '06

Hi, I'm new. I was wondering if someone could tell me which SIP softphone I can use with OpenSER that supports presence en IM??? regards, Kc

2 1

[Users] new set of modules: presence user agent client implementations
by Daniel-Constantin Mierla 07 Dec '06

07 Dec '06

Hello everybody, during last week a new set of modules have been introduced to CVS repository. They are prefixed with 'pua' which stands for 'presesece user agent'. There is one module actually having the name 'pua', which implements the common API needed by other modules from this category. At this moment there are two modules which behave as presence user agent for user location records (pua_usrloc) and presence user agent for external applications which can use new management interface (MI - via FIFO file or XMLRPC) to publish details -- this module is named pua_mi. Next step in this direction will be implementation of PUA for xmpp. With pua_usrloc module you can configure OpenSER to publish online/offline status in behalf of SIP devices registered to OpenSER but not supporting presence extensions. The users which have phones supporting presence extension can see if the other users are registered or not. pua_mi module is one very interesting. It enables external applications to publish presence information, through a FIFO file or XMLRPC at this moment. In this way, basically one adds SIMPLE support to whatever they need in a easy and comfortable manner. PUBLISH support is there, SUBSCRIBE is implemented but needs an outward communication channel from OpenSER while MI is inward. Having two new features in place (the MI and SIMPLE/Presence stuff), I though it may be more appealing to start trying and testing if some examples are made available. Therefore I created a dokuwiki page where you can find two shell scripts along with OpenSER configuration file. http://openser.org/dokuwiki/doku.php/presence:pua-modules One of the shell scripts publishes details from operating system: logged in users (U), system CPU usage(C), free memory (M) and load average (L). The other one publishes details gathered from OpenSER statistics: processed SIP requests (R), active transactions (T) and user location records (U). I think they are good examples to start with. I made two screenshots of XLite v3.0 subscribed to such presence information. http://www.openser.org/downloads/presence/xlite-openser-system.jpg http://www.openser.org/downloads/presence/xlite-openser-stats.jpg Readme files for these new modules: http://www.openser.org/docs/modules/1.2.x/pua.html http://www.openser.org/docs/modules/1.2.x/pua_mi.html http://www.openser.org/docs/modules/1.2.x/pua_usrloc.html We have a lot of new code in place, the testing is a very appreciated help you can give. Cheers, Daniel

2 2

[Users] Simplified multilateral peering for a lab environment
by Aaron Daubman 07 Dec '06

07 Dec '06

Greetings, I've found some decent documentation on secure multilateral peering for OpenSER, however I'm looking for something more simple. In a lab environment, I'm looking to set up four simulated geographically diverse networks all peering across a common backbone. Each of the four networks should have its own SIP registrar. What I'm looking for is a quick guide on how to allow each of the four registrars tell the other three what prefixes are registered with what address so that all SIP UAs can dial just an extension to reach UAs in any of the four networks. Right now I'd like to leave things unsecured if possible (no need for TLS). Also, if there are any good pointers for setting up the DNS in each of the four networks (each network will have its own DNS as well as SIP server) those would be useful. Finally, IPv4 references would be helpful, but seeing as this will be a v6-only setup, any reverences to IPv6 configuration guides regarding the above would be especially useful. Thanks again, ~Aaron

4 4

[Users] t_relay handling of r-URI, d-URI and Route URIs
by Klaus Darilion 07 Dec '06

07 Dec '06

Hi! Can someone please tell me how t_relay exactly works? What URI will be used for forwarding if d-URI is set and Route headers are present? Or does loose_route set the d-URI? What if I add a route header with append_hf - will it be used for t_relay? regards klaus -- Klaus Darilion nic.at

2 1

[Users] OpenSer Sending CANCEL after receiving 200 OK for a INVITE
by Md. Samiul Aftad Chowdhury 07 Dec '06

07 Dec '06

Hello Everyone, I am facing a problem when I am trying to send calls from my Openser(1.1.0) to another sip switch "MERA MSIP v.1.0.2." Here is the senario: openser( PublicIP 54.15.32.44) <====> Mera Sip Switch(PublicIP 52.132.36.43) ======================= ========================== INVITE ------------------> <----------------- 100 Trying <----------------- 183 Session Progress <----------------- 200 OK (for invite) CANCEL -------------------> <----------------- 200 OK (for cancel) <----------------- 487 Request Terminated ACK -------------------> This seems quite normal senario but prolem is that this "CANCEL" is not generated by the end point (in mycase IP Phone in public IP). This "CANCEL" generated from openser just after receiving the 200 OK of INVITE request. Why openser generateing a CANCEL while it already received 200 OK for that INVITE?? Here is my openser configuration script: openser.cfg ===================Script Begin=========================== # USE THIS VERSION IS REGISTER # ----------- 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 */ log_facility=LOG_LOCAL0 listen=54.15.32.44 alias="sip.myfone.com" check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- mpath="/usr/local/openser/lib/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "exec.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "acc.so" loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #modparam("rr", "enable_double_rr", 1) # For fail over # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others modparam("tm", "fr_inv_timer", 60 ) modparam("tm", "fr_timer", 30 ) #for Radius Accounting modparam("acc", "radius_flag", 1 ) modparam("acc", "radius_missed_flag", 3 ) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT ## Just for Fun,It shows color Log modparam("xlog", "force_color", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ 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; }; route(2); if (!method=="REGISTER") record_route(); if(method=="INVITE" || method=="BYE") setflag(1); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; #This is the best place for rewrite host port if (!method=="REGISTER") { rewritehostport("52.132.36.43:5060"); }; if (uri==myself) { if (method=="REGISTER") { if (!radius_www_authorize("54.15.32.44")) { www_challenge("54.15.32.44", "0"); exit; }; if (isflagset(5)) { setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; if (method=="INVITE") { xlog("L_INFO", "Invite -> [CallerID=$fu] and [DialedNo=$tU]\n"); if(uri=~"sip:[0-9]{20,}@.+") { sl_send_reply("513", "Message too big"); exit; }; if (!radius_www_authorize("54.15.32.44")) { www_challenge("54.15.32.44", "0"); exit; }; consume_credentials(); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ # xlog("L_INFO","Invite -> Force Rtp Proxy"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes;lr=on'); t_on_reply("1"); } failure_route[1] { if(!t_was_cancelled()) { rewritehostport("52.132.36.43:5060"); xlog("L_INFO","Invite -> Fail Over Route 1 at Time [$Tf]"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("2"); t_relay(); } } failure_route[2] { if(!t_was_cancelled()) { rewritehostport("52.132.36.43:5060"); xlog("L_INFO","Invite -> Fail Over Route 2 at Time [$Tf]"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("3"); t_relay(); } } failure_route[3] { if(!t_was_cancelled()) { rewritehostport("52.132.36.43:5060"); xlog("L_INFO","Invite -> Fail Over Route 3 at Time [$Tf]"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("4"); t_relay(); } } failure_route[4] { if(!t_was_cancelled()) { rewritehostport("52.132.36.43:5060"); xlog("L_INFO","Invite -> Fail Over Route 4 at Time [$Tf]"); if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } t_on_failure("5"); t_relay(); } } failure_route[5] { if (isflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy("r"); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isflagset(6)) { fix_nated_contact(); } exit; } ==========================Script End====================== Here is the sip trace (from ngrep): U IP Phone -> Openser INVITE sip:9128222570841@54.15.32.44 SIP/2.0. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport. Max-Forwards: 70. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. Authorization: Digest username="561705239298", realm="54.15.32.44", nonce="4577de9a6e32a8fafc108687cbc9b520cfd2b1ea", uri=" sip:9128222570841@54.15.32.44", response="c18eeb314b2c5524ab3d7fe50b52851c", algorithm=MD5. CSeq: 2 INVITE. Supported: replaces. Content-Type: application/sdp. Content-Length: 134. . v=0. o=- 12369099 22311551 IN IP4 58.88.13.13. s=SIP CALL. c=IN IP4 58.88.13.13. t=0 0. m=audio 8704 RTP/AVP 4. a=rtpmap:4 G723/8000. # U Openser -> IP PHONE SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: openser (1.1.0 (i386/linux)). Content-Length: 0. Warning: 392 54.15.32.44:5060 "Noisy feedback tells: pid=27317 req_src_ip= 58.88.13.13 req_src_port=5060 in_uri=sip:9128222570841@54.15.32.44 out_uri= sip:9128222570841@52.132.36.43:5060 via_cnt==1". . # U Openser -> Mera INVITE sip:9128222570841@52.132.36.43:5060 SIP/2.0. Record-Route: <sip:54.15.32.44;lr=on;ftag=1pRGOP6lOmlBFIxy>. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.0. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Max-Forwards: 69. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. CSeq: 2 INVITE. Supported: replaces. Content-Type: application/sdp. Content-Length: 152. . v=0. o=- 12369099 22311551 IN IP4 58.88.13.13. s=SIP CALL. c=IN IP4 54.15.32.44. t=0 0. m=audio 39780 RTP/AVP 4. a=rtpmap:4 G723/8000. a=nortpproxy:yes. # U Openser -> Mera INVITE sip:9128222570841@52.132.36.43:5060 SIP/2.0. Record-Route: <sip:54.15.32.44;lr=on;ftag=1pRGOP6lOmlBFIxy>. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Max-Forwards: 69. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. CSeq: 2 INVITE. Supported: replaces. Content-Type: application/sdp. Content-Length: 152. . v=0. o=- 12369099 22311551 IN IP4 58.88.13.13. s=SIP CALL. c=IN IP4 54.15.32.44. t=0 0. m=audio 39780 RTP/AVP 4. a=rtpmap:4 G723/8000. a=nortpproxy:yes. # U Mera -> Openser SIP/2.0 100 Trying. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.0,SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Content-Length: 0. . # U Mera -> Openser SIP/2.0 100 Trying. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1,SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44>. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Content-Length: 0. . # U Mera -> Openser SIP/2.0 183 Session Progress. Record-Route: <sip:52.132.36.43:5060;lr>,<sip:54.15.32.44 ;lr=on;ftag=1pRGOP6lOmlBFIxy>. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: MERA MSIP v.1.0.2. Contact: <sip:9128222570841@52.132.36.43:5060>. Content-Type: application/sdp. Via: SIP/2.0/UDP 54.15.32.44:5060;branch=z9hG4bKa124.220e0023.0,SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Content-Length: 136. . v=0. o=- 1165483497 1165483497 IN IP4 52.132.36.43. s=-. c=IN IP4 52.132.36.43. t=0 0. m=audio 37548 RTP/AVP 4. a=rtpmap:4 G723/8000. . # U Openser -> IP PHONE SIP/2.0 183 Session Progress. Record-Route: <sip:52.132.36.43:5060;lr>,<sip:54.15.32.44 ;lr=on;ftag=1pRGOP6lOmlBFIxy>. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: MERA MSIP v.1.0.2. Contact: <sip:9128222570841@52.132.36.43:5060;nat=yes>. Content-Type: application/sdp. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Content-Length: 153. . v=0. o=- 1165483497 1165483497 IN IP4 52.132.36.43. s=-. c=IN IP4 54.15.32.44. t=0 0. m=audio 39784 RTP/AVP 4. a=rtpmap:4 G723/8000. . a=nortpproxy:yes. # U Mera -> Openser SIP/2.0 200 OK. Record-Route: <sip:52.132.36.43:5060;lr>,<sip:54.15.32.44 ;lr=on;ftag=1pRGOP6lOmlBFIxy>. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: MERA MSIP v.1.0.2. Contact: <sip:9128222570841@52.132.36.43:5060>. Content-Type: application/sdp. Via: SIP/2.0/UDP 54.15.32.44:5060;branch=z9hG4bKa124.220e0023.0,SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Content-Length: 136. . v=0. o=- 1165483501 1165483501 IN IP4 52.132.36.43. s=-. c=IN IP4 52.132.36.43. t=0 0. m=audio 37548 RTP/AVP 4. a=rtpmap:4 G723/8000. . # U Openser -> IP Phone SIP/2.0 200 OK. Record-Route: <sip:52.132.36.43:5060;lr>,<sip:54.15.32.44 ;lr=on;ftag=1pRGOP6lOmlBFIxy>. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. CSeq: 2 INVITE. Server: MERA MSIP v.1.0.2. Contact: <sip:9128222570841@52.132.36.43:5060;nat=yes>. Content-Type: application/sdp. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKI7LP7qhgGKFHt904;rport=5060. Content-Length: 153. . v=0. o=- 1165483501 1165483501 IN IP4 52.132.36.43. s=-. c=IN IP4 54.15.32.44. t=0 0. m=audio 39784 RTP/AVP 4. a=rtpmap:4 G723/8000. . a=nortpproxy:yes. # U Openser -> Mera CANCEL sip:9128222570841@52.132.36.43:5060 SIP/2.0. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. To: "9128222570841" <sip:9128222570841@54.15.32.44>. CSeq: 2 CANCEL. User-Agent: openser (1.1.0 (i386/linux)). Content-Length: 0. . # U Mera -> Openser SIP/2.0 200 OK. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. To: "9128222570841" <sip:9128222570841@54.15.32.44>. CSeq: 2 CANCEL. Content-Length: 0. . # U Mera -> Openser SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. To: "9128222570841" <sip:9128222570841@54.15.32.44>. CSeq: 2 INVITE. Content-Length: 0. . # U Openser -> Mera ACK sip:9128222570841@52.132.36.43:5060 SIP/2.0. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.1. From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. To: "9128222570841" <sip:9128222570841@54.15.32.44>. CSeq: 2 ACK. User-Agent: openser (1.1.0 (i386/linux)). Content-Length: 0. . # U IP Phone -> Openser ACK sip:9128222570841@52.132.36.43:5060 SIP/2.0. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKTpMoqwSSNb4LvZrv;rport. Route: <sip:54.15.32.44;lr=on;ftag=1pRGOP6lOmlBFIxy>, <sip:52.132.36.43:5060 ;lr>. Max-Forwards: 70. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. Authorization: Digest username="561705239298", realm="54.15.32.44", nonce="4577de9a6e32a8fafc108687cbc9b520cfd2b1ea", uri=" sip:9128222570841@52.132.36.43:5060", response="7b312414aa7de3f999f8fd4a2a81c8a5", algorithm=MD5. CSeq: 2 ACK. Content-Length: 0. . # U Openser -> Mera ACK sip:9128222570841@52.132.36.43:5060 SIP/2.0. Record-Route: <sip:54.15.32.44;lr=on;ftag=1pRGOP6lOmlBFIxy>. Via: SIP/2.0/UDP 54.15.32.44;branch=z9hG4bKa124.220e0023.3. Via: SIP/2.0/UDP 58.88.13.13:5060;branch=z9hG4bKTpMoqwSSNb4LvZrv;rport=5060. Route: <sip:52.132.36.43:5060;lr>. Max-Forwards: 69. User-Agent: PA168S V1.55.013 CFG0 . From: "561705239298" <sip:561705239298@54.15.32.44>;tag=1pRGOP6lOmlBFIxy. To: "9128222570841" <sip:9128222570841@54.15.32.44 >;tag=ffff1200ffff7710ff00000f1f6fff3b. Call-ID: BCmB3zph5YmuC8oz(a)58.88.13.13. Contact: <sip:561705239298@58.88.13.13:5060>. Authorization: Digest username="561705239298", realm="54.15.32.44", nonce="4577de9a6e32a8fafc108687cbc9b520cfd2b1ea", uri=" sip:9128222570841@52.132.36.43:5060", response="7b312414aa7de3f999f8fd4a2a81c8a5", algorithm=MD5. CSeq: 2 ACK. Content-Length: 0. P-hint: rr-enforced. .

2 2

[Users] Broken link for PUA_USRLOC
by Nicolas Palix 07 Dec '06

07 Dec '06

Hi, In the page http://www.openser.org/docs/modules/1.2.x/ PUA_USRLOC module refers to http://www.openser.org/docs/modules/devel/pua.html instead of http://www.openser.org/docs/modules/devel/pua_usrloc.html Regards, -- Nicolas Palix http://phoenix.labri.fr/people/palix/

2 1

RE: [Users] MediaProxy SER with AsteriskS
by Thomas Deillon 07 Dec '06

07 Dec '06

Hi all, I try a configuration with a SER server with mediaproxy that load balance SIP/RTP two 2 asterisks server (REGISTRAR, ... ) SER: 192.168.1.81 and 10.10.10.1 Asterisk1: 10.10.10.12 with default route 10.10.10.1 Asterisk2: 10.10.10.13 with default route 10.10.10.1 Mediaproxy.ini: [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock group = root defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = root proxyIP = 192.168.1.81 portRange = 60000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 openser.cfg ... 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/dispatcher.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/mediaproxy.so" loadmodule "/usr/lib/openser/modules/domain.so" loadmodule "/usr/lib/openser/modules/mysql.so" modparam("domain", "db_url", "mysql://ser:ser@localhost/ser") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "natping_interval", 20) modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") route{ ... if(method=="REGISTER"){ # For the moment I just do like this to register # my phone on one server forward("10.10.10.12"); } else { if (method==INVITE) { use_media_proxy(); } ds_select_dst("1", "4"); t_relay(); } } Dispatcher.list: 1 sip:10.10.10.12:5060 1 sip:10.10.10.13:5060 Thanks a lot for your help, Thomas

1 0

[Serusers] pstn gateway
by jorain 07 Dec '06

07 Dec '06

Hi all, I have a PSTN gateway (IP:123.456.78.90), when I route to it, there is no audio between PSTN and UA. I found that the PSTN gateway IP is different from the media IP, which is specified inside the SDP Connection Information (c) : IN IP4 xxx.xxx.xxx.xxx My RTPProxy is working and tested before. Please help. Thanks Regards, jorain

2 1

[Serusers] Ser accounting question?
by Lokesh Kumar 07 Dec '06

07 Dec '06

Hello List, Below mentioned is my accounting packet, I would like to know what is IAPP-AP-Check, what this check do , and where its configuration present. Can I change Service-Type ? Wed Dec 6 20:08:23 2006 Acct-Status-Type = Stop Service-Type = IAPP-AP-Check Sip-Response-Code = 200 Sip-Method = 8 User-Name = "69021(a)xxx.yyy.pt" Calling-Station-Id = "sip:69021@xxx.yyy.pt" Called-Station-Id = "sip:30612@xxx.yyy.pt" Sip-Translated-Request-URI = "sip:30612@212.13.xx.yy:5060" Acct-Session-Id = "D4808CE2-1BA9-4572-B82B-5A0BCF6AB7CB(a)192.168.0.47" Sip-To-Tag = "86949D74-2699" Sip-From-Tag = "67434843439" Sip-Cseq = "3" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "c6adb5da735b1150" Timestamp = 1165435703 Thanks Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.15.6/567 - Release Date: 12/4/2006

2 1

[Serusers] load balancing amongst servers
by Zeeshan Sajid 07 Dec '06

07 Dec '06

First of all thanks for the response and support ... really appreciate that ... This time the problem is alittle more complex now coming to the point I really need to know how can i do load balancing for a situation like [SER1] DIALER ----------------> [SER2] --------> PSTN [SER3] I have 3 SER servers and i want some kind of load balancing amongst them. I have been trying to make SER work behind a load balancer but that seriously involves mangling with some headers. So, is there any clean way to get this done ... Please help me in this as i am pretty struck at this point... regards, Zeeshan. --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.

2 1

[Users] openser.org web site: new look
by Daniel-Constantin Mierla 07 Dec '06

07 Dec '06

Hello, just to let you know that a new template which is more in harmony with the OpenSER logo has been deployed on the web site. Anyone who considers that can improve the design and make user's experience better, please send an email to devel(a)openser.org or team(a)openser.org. The site uses the open source cms mambo/joomla engine. The OpenSER logo should have replaced old variants, and has been posted in relevant new places (e.g., dokuwiki). See: http://www.openser.org If you want to get the OpenSER logo, there are two variants available on the site, with white background and transparent: http://www.openser.org/images/openser-logo.jpg http://www.openser.org/images/openser-logo-transparent.png Cheers, Daniel

1 0

[Users] avpops multi db query
by vallimamod abdullah 07 Dec '06

07 Dec '06

Hello list, For my call routing algorithm, I need to query different databases to retrieve the destination of an outgoing call. From what I have understood, I can specify only one db url in the avpops module to use with avp_db_query(). Is there a (simple) way to make queries in different databases with openser ? Thanks. -- Regards, -vma .

2 1

[Users] ACK and re-INVITE ordering
by Jerome Martin 07 Dec '06

07 Dec '06

Hello everyone, I found many references in serdev, openser users, etc. MLs to the issue at hand, but only came up with diverging opinions and no solution whatsoever. The issue appears in case of a UA sending a re-INVITE inside of an established dialog. Depending on your configuration file, version, network conditions, there is NO guarantee that this re-INVITE will be sent to (I'm not even talking about reaching) the next hop (typically a PSTN gateway) AFTER the ACK to the first INVITE of the dialog. I perfectly understand why (openser being transaction and not dialog statefull being the only reason that really matters, given the fact that INVITE and ACK are separate transactions, often handled by separate openser processes) it is this way. However, it seems that several gateways are not handling this well at all, and just drop the transaction typically sending a "500 Server internal error". I've read reports about cisco AS5300, Asterisk, and I am myself experiencing this with an Andiocodes Mediant 2000 gateway. In my case, the issue is there 99% of the time. I know I could profile a bit my config file to ensure ACKs are processed faster than INVITEs (which propably is already the case), but this is hardly a workaround. I got tempted to even call an external "timer" via exec_msg when detecting a re-INVITE (if it is an INVITE, has a to_tag and is loose routed), but come on, I'm sure we can do better than this ! So no matter what the RFCs say or mean to say, no matter how long we argue about this, the facts are : - gateways often don't stand this ordering issue - many iPBXs are using reINVITEs for several call features - the only way to solve this is to be dialog statefull, at least for ACKs and INVITES So my conclusion is : I'm going to code an ugly little hack - using an external database and avpops - so that ACKs are logged per Cseq+callid, and when reinvites are detected, relay will be delayed until the last ACK is logged as sent. What do you think ? Is there a way to use the dialog module to optimize this (flagging the SIP headers themselves instead of an external DB) ? Best Regards, Jerome Martin

6 20

Re: [Serusers] SER Oracle Support
by Dmitry Semyonov 07 Dec '06

07 Dec '06

Abdul, On Tue, 5 Dec 2006 at 18:22 -0800, Abdul Qadir wrote: > I have attached my oralce include file. You should not send big attachments to the list. I unpacked your archive to /usr/local/oracle9, and changed DEFS definition inside modules/oracle/Makefile to DEFS += -I/usr/local/oracle9/demo -I/usr/local/oracle9/public After that I was able to build the module without errors with "make modules=modules/oracle modules" command. Please, double-check your path to the includes, and verify it is properly passed to the compiler. Sample output for dbase.c compilation is as follows (note the include paths at the end): gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -ftree-vectorize -mtune=athlon64 -DNAME='"ser"' -DVERSION='"0.9.7-pre7"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DLINUX -D_GNU_SOURCE -D_REENTRANT -I/usr/local/oracle9/demo -I/usr/local/oracle9/public -c dbase.c -o dbase.o > Dmitry Semyonov <dsemyonov(a)dins.ru> wrote: Abdul, > > On Mon, 4 Dec 2006 at 23:44 -0800, Abdul Qadir wrote: > > > Thats why I did modified the make file to reflect my path. But > > still getting the same error. Do u think I still need to install > > Instant Client, although I have full client installed. > > I don't think so. -- ...Bye..Dmitry.

2 3

Re: [Serusers] Call transfer
by jorain 07 Dec '06

07 Dec '06

Hi all, Is it deal with the ctd.sh script? Configure inside loose_route()? I can get the REFER message but no responses (202 Accepted and NOTIFY) to it. So I send reply 202 inside loose_route if method=REFER. After that, inside ethereal, I can see the code 202 response to REFER. However, It send REFER again and after a while 408 TIMEOUT. This is what i configured: if(loose_route()){ if(method=="BYE"){ setflag(1); setflag(2); }; if(method=="REFER"){ sl_send_reply("202","Accepted"); }; t_relay(); break; }; Please help. Thanks in advance Regards, jorain ----- Original Message ----- From: Greger V. Teigre To: jorain Sent: Tuesday, December 05, 2006 9:30 PM Subject: Re: [Serusers] Call transfer cc serusers jorain wrote: Thanks Greger, Could you give me some clues how to do that or some references about it? I really have no idea. Thanks again. Regards, jorain ----- Original Message ----- From: Greger V. Teigre To: jorain Cc: SER+RTPproxy Sent: Monday, December 04, 2006 7:48 PM Subject: Re: [Serusers] Call transfer Well, yes, you have to route REFER, and you should allow NOTIFYs. g-) jorain wrote: Hi all, I have some question about call transfer. When I press transfer key follow by the new destination and "send", I can see the REFER message inside ethereal but after that request timeout error(408). Inside REFER message, i can see the refer-to and referred-by values are correct. Do I need to add something inside ser.cfg to enable call transfering? Because I cannot find any sample codings regarding ser call transfer. Thanks for any help. Regards, jorain ------------------------------------------------------------------------ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 2

Re: [Users] using the acc module
by Daniel Burge 06 Dec '06

06 Dec '06

Hi Nick, Here is how the top of my cfg looks now: debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL6 I also added this line to syslog.conf: # Save openser accounting logs to openseracc.log local6.* -/var/log/openseracc.log Previously there was information written to /var/log/openser.log which contained mostly things that happened on start up. It seems that openser logs to that file by default even if there is no log_facility set in the cfg file. Now all those messages are instead being put in /var/log/openseracc.log. However, still there are no accounting messages logged. Any ideas? Thanks again, Daniel nick wrote: > Daniel Burge wrote: >> Hello, >> >> I'm trying to configure the accounting module to log messages to >> syslog by following the instructions here: >> >> http://www.openser.org/docs/modules/1.1.x/acc.html >> >> I've included these lines where appropriate according to the example >> on that page. The meaningful parts of the code are as follows: >> >> loadmodule "/usr/local/lib64/openser/modules/acc.so" >> modparam("acc", "log_level", 1) >> modparam("acc", "log_flag", 1) >> >> route >> { >> if (method=="INVITE" || method=="BYE" || method=="CANCEL") >> { >> setflag(1); /* set for accounting (the same value as in log_flag!) >> } } >> >> However this is not causing any information to be logged to syslog. >> >> Any idea? >> >> Thanks in advance, >> >> Daniel >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > > > Make sure you put a line at the top of your config like: > > log_facility=LOG_LOCAL6 > > The number at the end can vary, it depends on what Log facilities you > have "free" on your system. > >

1 0

[Users] using the acc module
by Daniel Burge 06 Dec '06

06 Dec '06

Hello, I'm trying to configure the accounting module to log messages to syslog by following the instructions here: http://www.openser.org/docs/modules/1.1.x/acc.html I've included these lines where appropriate according to the example on that page. The meaningful parts of the code are as follows: loadmodule "/usr/local/lib64/openser/modules/acc.so" modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) route { if (method=="INVITE" || method=="BYE" || method=="CANCEL") { setflag(1); /* set for accounting (the same value as in log_flag!) } } However this is not causing any information to be logged to syslog. Any idea? Thanks in advance, Daniel

2 2

[Users] NEW Pseudo-Variable: $T_reply_code
by Bogdan-Andrei Iancu 06 Dec '06

06 Dec '06

Hi everybody, following some time ago discussion, a new Pseudo-Variable (PV) is exported by the TM module "- $T_reply_code" - this PV will offer access the the reply code from all different kind of routes. See: http://www.openser.org/docs/modules/1.2.x/tm.html#AEN562 This will allow mainly access (printing, comparing, pushing into AVPs, logging, etc) to the reply code of the winning reply in failure_route. regards, bogdan

1 0

[Users] UAC module use and REGISTER problem
by Christophe Irles 06 Dec '06

06 Dec '06

Hi, I'm using UAC module in order to centralized authentication when I need to use an external proxy server to call PSTN numbers. To use it, I use a unique tuple login/pwd/domain The environment is: My openser handles the domain: "myopenserdomain.com" For each INVITE or REGISTER of SIP client of this domain, I force the authentication The external proxy handles the domain "externaldomain.com" Now I have two questions: 1) before to send an INVITE to the external proxy, must I change the contact header with the correct information of the tuple login/pwd/domain ? Or I must wait the 401 or 407 message before ? In fact I tests several external proxy. Some are waiting a contact header with the information of the tuple and some don't. What is the correct way to do this ? 2) Some of this external proxy accept only my INVITE (with or without contact header modified) if I registered first on it ! I receive 403 message in this situation. But I don't know how to REGISTER first since I don't find any openser module to that . How can I handle this situation ? Thanks, Regards, Christophe

2 1

[Users] NEW FEATURES: t_replay() enhancements
by Bogdan-Andrei Iancu 06 Dec '06

06 Dec '06

Hi everybody, following several discussion threads from some time ago, I added a support for controlling the behaviour of t_relay() functions. This addresses: 1) to ability to stop t_relay() from automatically sending the 100 trying provisional reply before relaying the request. This is useful in scenarios where you already send a stateless 100 trying from the script. 2) ability to do serial forking in failover scenarios - as original t_relay() automatically sends a negative reply in case of a failure (unresolvable URI, TCP connect refused, etc), it was impossible to continue with the next destination. Now, the auto-reply can be disabled. backward compatibility is kept. For more details (how to do it, options, etc), please see the online docs: http://www.openser.org/docs/modules/1.2.x/tm.html#AEN340 regards, bogdan

1 0

RE: [Users] Load Balancing
by Thomas Deillon 06 Dec '06

06 Dec '06

Hi all, Someone using LVS can send his configuration to help us? Thanks, Thomas -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Klaus Darilion Envoyé : mercredi, 6. décembre 2006 09:58 À : Watkins, Bradley Cc : users(a)openser.org Objet : Re: [Users] Load Balancing Hi Bradley! Watkins, Bradley wrote: > > I use UltraMonkey (www.ultramonkey.org), but it's basically a packaging > of Linux-HA and LVS. > > LVS supports connection persistence, and that works very well. The > level of granularity of load balancing is on a per-IP basis, not > per-transaction (LVS only does L3 load-balancing, not L4-7). But > assuming a reasonable value for the persistence timeout, there should be > no problems with SIP transactions (and I haven't experienced anything > problematic). Are you using TCP or UDP (or both) for SIP? Reading LVS doc I see there are several methods for packet forwarding: - Direct Routing - IP-IP Encapsulation - NAT Which one do you use? regards Klaus > > Ldirectord also supports SIP health checks (they're pretty basic OPTIONS > requests, but it's something), and will fail over to working SIP > proxies. Of course, in-progress transactions will fail as the OpenSER > process on one box doesn't have the stateful knowledge of the other. > > Regards, > - Brad > The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- Klaus Darilion nic.at _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Users] NEW: XMLRPC implementation for MI
by Bogdan-Andrei Iancu 06 Dec '06

06 Dec '06

Hi, thanks to Lavinia Andrei, we added today a new transport implementation for the Management Interface (MI). This is XMRPC and it the second after the initial FIFO transport, but not the last :). The module is build by using the xmlrpc library, following all XMLRPC standards. It is in alpha stage, so more testing is needed. NOTE that only commands ported to the new MI interface (and not all former FIFO cmds) are available via XMLRPC, but this is a temporary problem - there are only 2 module left to update (TM and USRLOC). For more info, please see the online documentation: http://www.openser.org/docs/modules/1.2.x/mi_xmlrpc.html or ask on the mailing list. Any feedback - bugs or improvements - are welcomed. Regards, Bogdan

2 3

[Users] bug in 1.1.x siptrace online docs?
by Klaus Darilion 06 Dec '06

06 Dec '06

Hi! http://www.openser.org/docs/modules/1.1.x/siptrace.html#AEN119 As far as I can see the "trace_local_ip" module parameter is not in 1.1, but in the online docs. regards klaus -- Klaus Darilion nic.at

2 1

[Users] mySQL database documentation
by Gregoire 06 Dec '06

06 Dec '06

Hi all! Does someone know where I can find documentation about the database structure used with OpenSER. I mean the signification of the tables and their fields. Thanks in advance Greg

3 2

[Serusers] OT: Looking for a hardware SIP phone that supports IPv6
by Aaron Daubman 06 Dec '06

06 Dec '06

Greetings all, Sorry for the somewhat off-topic post. I will be using SER on a ipv6-only network and am looking for some hardware SIP phones that support IPv6. Currently, I've found very few (and unfortunately all now defunct) links on iptel's phone list that point to IPv6 capable hardphones, as well as: http://www.moimstone.com/e_product/s01.php which has very little detail other than "Support IPv6 (Optional), IPv4". Does anybody know of other hardware SIP phones that support IPv6? Has anybody actually used hardware IPv6 phones with (Open)SER? Thanks, ~Aaron

1 1

[Serusers] Help! Run SER with MySQL to do authentication
by Tako Chang 06 Dec '06

06 Dec '06

Hi, all First of all, I read [SER - Getting Started<http://siprouter.onsip.org/doc/gettingstarted/>] and downloaded [ SER Source package<ftp://siprouter.onsip.org/pub/gettingstarted/packages/ser-0.9.7-pre1.Gettin…>] after I make the package under debug mode, it runs well. Now, I can receive REGISTER requests and send 200 OK Responses and play a SIP proxy server I want to test "auth_mysql.cfg" in Chapter 7.... so, i remove "exclude_modules?= jabber mysql\" in Makefile but i still can't compile the mysql.so Is there some more detail documents or suggestions for me? Thank!

2 1

[Users] Using Openser's CPL module...
by Marvin Ottley 06 Dec '06

06 Dec '06

Good day all, I'm trying to use the CPL module and I'm having trouble figuring out how to load the cpl xml script into the cpl database. Thanks for your pointers Marvin

4 5

[Serusers] Error while loading module TM. Please solution!!!
by Meidiana Ten 06 Dec '06

06 Dec '06

Hi there, I have problem in loading module TM for voicemail setup here. I use SER 0.9.7 as server and the latest SEMS for voicemail usage. I tried to add my ser.cfg with script that can handle the voicemail but I found errors when run SER. Below is my ser.cfg: (Please take a look at line 54 and 56) debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("tm", "fr_timer", 10 ) modparam("tm", "wt_timer", 10 ) line 54: modparam("tm", "pass_provisional_replies", 1) # configure tm to append this when tw_appent voicemail_headers is used line 56: modparam("tm", "tw_append","voicemail_headers:P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[Content-Type];msg[body]") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) # configure avpops db connection modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) modparam( "avpops", "avp_table", "subscriber" ) modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database modparam( "avpops", "db_scheme", "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam( "avpops", "db_scheme", # "language_scheme:table=subscriber;value_col=language;value_type=string") alias="pcr.ac.id" route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; /* if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); break; } */ # make transaction if (!t_newtran()){ log("could not create transaction\n"); sl_send_reply("500","could not create transaction"); break; } # actively absorb ACKs if (method == "ACK") { t_relay(); break; } # pass INFO to SEMS if (method=="INFO") { if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ log("could not contact sems\n"); t_reply("500","could not contact media server"); } } if (uri =~ "sip:101.*@") { if (!t_write_unix("/tmp/sems_sock","myapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:102.*@") { if (!t_write_unix("/tmp/sems_sock","myconfigurableapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:103.*@") { if (!t_write_unix("/tmp/sems_sock","myannounceapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:104.*@") { if (!t_write_unix("/tmp/sems_sock","myjukebox")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:105.*@") { if (!t_write_unix("/tmp/sems_sock","mycc")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:106.*@") { if (!t_write_unix("/tmp/sems_sock","ivr_announce")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } ############ default sems apps if (uri =~ "sip:110.*@") { if (!t_write_unix("/tmp/sems_sock","echo")){ log("could not contact echo\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:111.*@") { if (!t_write_unix("/tmp/sems_sock","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:112.*@") { if (!t_write_unix("/tmp/sems_sock","conference")){ log("could not contact conference\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:113.*@") { if (!t_write_unix("/tmp/sems_sock","mailbox")){ log("could not contact mailbox\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:114.*@") { if (!t_write_unix("/tmp/sems_sock","early_announce")){ log("could not contact early_announce\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("root@localhost","$email"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail/voicemail_headers")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); break; } break; } t_reply("404","Not found"); # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Offline Message Store Section # ----------------------------------------------------------------- if (is_from_local()) { if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); 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("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted() && nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.2:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } -------------------------------------------------------------------------------------------------------------------- If I uncomment line 54 and 56, below is the output when I run SER: 0(4425) set_mod_param_regex: parameter <pass_provisional_replies> not found in module <tm> 0(4425) parse error (54,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm 0(4425) set_mod_param_regex: found <tw_append> in module tm [/usr/local/lib/ser/modules/tm.so] 0(4425) ERROR:tm:parse_tw_append: bad alias spec <$email> 0(4425) parse error (56,19-20): Can't set module parameter 0(4425) set_mod_param_regex: tm matches module tm ----------------------------------------------------------------------------------------------------------------------- If I comment line 54 and 56, below is the output: 0(0) ERROR:tm:fixup_t_write: unknown append name <voicemail_headers> 0(0) ERROR: fix_expr : fix_actions error ERROR: error -6 while trying to fix configuration 0(0) MSILO: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5050) shm_mem_destroy 0(5050) destroying the shared memory lock ----------------------------------------------------------------------------------------------------------- Please tell me what's wrong. And please tell me whether if my configuration on ser.cfg is right or wrong because I doubt it. Thanx before... Regards, Meidiana --------------------------------- Have a burning question? Go to Yahoo! Answers and get answers from real people who know.

1 0

[Users] migrating DB to OpenSER
by Stefano Capitanio 06 Dec '06

06 Dec '06

Hi, i want to migrate from SER 0.9.4 to OpenSER 1.1.0 and i would like to mantain the MySQL DB actually in use. is it possible? which tables i have to alter and how? thanks, Stefano

1 0

[Users] identifying what an ACK is in response to
by Mark Price 06 Dec '06

06 Dec '06

Hi, How does one determine which part of a SIP conversation an ACK packet is responding to? I would have guessed that it would be in tne CSeq, but that field just says, for example, CSeq: 3 ACK The only method I have come up with so far is looking at the source and destination ports + the destination of the Request-LIne and try to determine the most recent message it might be responding to. Thanks, Mark Price

2 1

[Users] bug in create DB scripts?
by Klaus Darilion 06 Dec '06

06 Dec '06

Hi! I wonder if the creation of the serweb default admin user works. Because the function credentials() is never called inside postgresqldb.sh and mysql.... regards klaus -- Klaus Darilion nic.at

2 4

RE: [Users] Load Balancing
by Watkins, Bradley 06 Dec '06

06 Dec '06

> -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Andreas Granig > Sent: Tuesday, December 05, 2006 8:20 AM > To: Marcos Hack > Cc: Thomas Deillon; users(a)openser.org > Subject: Re: [Users] Load Balancing > > Marcos Hack wrote: > > I don't know these modules, but here we solve this problem using a > > unique VIP (Virtual IP) for all openSER cluster nodes. > > How do you ensure that all messages belonging to one > transaction are balanced to the same physical server? Is > there now SIP support in LVS? > I use UltraMonkey (www.ultramonkey.org), but it's basically a packaging of Linux-HA and LVS. LVS supports connection persistence, and that works very well. The level of granularity of load balancing is on a per-IP basis, not per-transaction (LVS only does L3 load-balancing, not L4-7). But assuming a reasonable value for the persistence timeout, there should be no problems with SIP transactions (and I haven't experienced anything problematic). Ldirectord also supports SIP health checks (they're pretty basic OPTIONS requests, but it's something), and will fail over to working SIP proxies. Of course, in-progress transactions will fail as the OpenSER process on one box doesn't have the stateful knowledge of the other. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

2 1

[Users] forwarding requests...
by Frogger 06 Dec '06

06 Dec '06

I am trying to forward requests so that the request is completely gone and the UA is forwarded to the final destination with no signaling from the forwarding system. For example: "5551212@" would be forwarded to 1.1.1.2 but 1.1.1.2 would not get a redirect request from 1.1.1.1 rather, directly from the UA. Any thoughts? Thanks! FR ____________________________________________________________________________________ Access over 1 million songs - Yahoo! Music Unlimited (http://music.yahoo.com/unlimited)

2 2

[Users] Jabber (Authentication to the Jabber server failed)
by Vanderson Rocha 05 Dec '06

05 Dec '06

Hi all, I'm trying to configure jabber module in OpenSER and this is what I get when sending a message from Ekiga to a Jabber user: "0(26030) XJAB:xj_jcon_connect: socket [7] 0(26030) XJAB:xj_worker: auth to jabber as: [teste2] / [xxx] 0(26030) XJAB:xj_worker:26154: Authentication to the Jabber server failed ... 0(26030) XJAB:xj_jcon_disconnect: -----START----- 0(26030) XJAB:xj_jcon_disconnect: socket [7] 0(26030) XJAB:xj_jcon_send_presence: -----START----- 0(26030) INFO: signal 13 received 0(26030) XJAB:xj_jcon_send_presence: Error - presence not sent 0(26030) INFO: signal 13 received 0(26030) XJAB:xj_jcon_disconnect: error closing stream 0(26030) XJAB:xj_jcon_disconnect: -----END----- " "jusers" table has only one entry: juid => 1 jab_id => teste2 jab_passwd => teste2 sip_id => sip:teste2@voip.mydomain ( This is the Ekiga user) type => 0 "openser.cfg" modparam("jabber", "db_url","postgres://openseruser:mypass@192.168.95.78/sip_jab") modparam("jabber", "jaddress", "192.168.95.135") modparam("jabber", "jport", 5222) modparam("jabber", "jdomain", "xmpp.mydomain=*") modparam("jabber", "registrar", "jabber_gateway(a)127.0.0.1") modparam("jabber", "workers", 2) modparam("jabber", "max_jobs", 10) modparam("jabber", "cache_time", 600) modparam("jabber", "delay_time", 90) modparam("jabber", "sleep_time", 20) modparam("jabber", "check_time", 20) Any sugestion? Thanks Vanderson.

1 0

[Users] SER/OpenSER + Asterisk + Queue
by lists＠infoway.net 05 Dec '06

05 Dec '06

We are in the process of redesigning our single Asterisk server that handles several queues for our clients. We offer our clients hosted queueing/call center basic services. All the agents are in remote locations behind NATs using either softphones or PAP2-like devices. What we would like to accomplish is setup a SER or OpenSER ("SER") server(s) in front of our Asterisk box such that all incoming and outgoing calls are handled by SER. The basic idea is to get set up for scaleability and redundancy. The goal is to be able to add additional Asterisk servers to spread our queue loads. Nothing fancy, maybe just separate clients on different boxes (not load balancing queues across multiple Asterisk boxes since that a totally different scope of project). We could then add additional SER boxes to protect our inbound and outbound SIP gateways to our SIP providers (all our calls are SIP-based - e.g. no TDM circuits). Lastly, all our agents would register against the SER server(s) instead of directly to the Asterisk boxes. Has anyone done this? Can anyone point me to some tips/documentation? Does anyone care to comment? If agents login using AgentCallBackLogin, will Asterisk know where the agents are and send the calls to them via SER? Thank you so much in advanced. - Daniel

1 0

[Serusers] SER/OpenSER + Asterisk + Queue
by lists＠infoway.net 05 Dec '06

05 Dec '06

We are in the process of redesigning our single Asterisk server that handles several queues for our clients. We offer our clients hosted queueing/call center basic services. All the agents are in remote locations behind NATs using either softphones or PAP2-like devices. What we would like to accomplish is setup a SER or OpenSER ("SER") server(s) in front of our Asterisk box such that all incoming and outgoing calls are handled by SER. The basic idea is to get set up for scaleability and redundancy. The goal is to be able to add additional Asterisk servers to spread our queue loads. Nothing fancy, maybe just separate clients on different boxes (not load balancing queues across multiple Asterisk boxes since that a totally different scope of project). We could then add additional SER boxes to protect our inbound and outbound SIP gateways to our SIP providers (all our calls are SIP-based - e.g. no TDM circuits). Lastly, all our agents would register against the SER server(s) instead of directly to the Asterisk boxes. Has anyone done this? Can anyone point me to some tips/documentation? Does anyone care to comment? If agents login using AgentCallBackLogin, will Asterisk know where the agents are and send the calls to them via SER? Thank you so much in advanced. - Daniel

1 0

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