- sr-users - kamailio.org

[Serusers] Why BYE doesn't accounted?
by Ryan Pagquil 05 Aug '05

05 Aug '05

Guys, Iv'e configured my SER to authenticate with freeradius, well thanks everybody. Yesterday I tried to make mysql accounting and it worked but the problem is when I hungup I can't see the BYE on the accounting, I only see INVITE and ACK. What should be the problem with my ser.cfg? Here is the configuration specifically with mysql accounting: modparam("acc", "log_flag", 1) modparam("acc", "log_level", 1) modparam("acc", "report_ack", 1) modparam("acc", "log_fmt", "cdfimorstup") … [View More]

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RE: [Serusers] Problem with SER and PAP2, I get a 407 message when trying a call
by Sam Lee 05 Aug '05

05 Aug '05

What exactly is the problem? I got a similar set up.... Regards, Sam -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Yiannis Martis Sent: Thursday, August 04, 2005 4:11 PM To: jiri(a)iptel.org; sisalem(a)iptel.org; serusers(a)lists.iptel.org Subject: [Serusers] Problem with SER and PAP2,I get a 407 message when trying a call Hello there, I am Network Administrator working for a company called Hellasfon Network S.A. I … [View More]use SER 0.9.3 with Vovida's STUN server. PAP2 UAis NATed and gets registered using my ADSL modem's IP and everything looks OK. The problem is when I try to make a call to another UA) of my service I get the following output from my server (ngrep):. Please help me if u can!!! U 2005/08/03 13:55:57.380953 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 INVITE sip:2810318308@XXXXXXXXXX.com SIP/2.0. Record-Route: <sip:XX.XX.XX.XX;ftag=7ab32f2bbf2689feo0;lr=on>. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Max-Forwards: 16. Contact: <sip:2810300308@XX.XX.XX.XX:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 420. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 13486 13486 IN IP4 XX.XX.XX.XX. s=-. c=IN IP4 XX.XX.XX.XX. t=0 0. m=audio 16420 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/08/03 13:55:57.381212 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a17 9. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="XXXXXXXXXX.com", nonce="42f0a3e9ea677d46038bdbeccaf8d95c18bce58e". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=4360 req_src_ip=XX.XX.XX.XX req_src_port=5060 in_uri=sip:2810318308@XXXXXXXXXX.com out_uri=sip:2810318308@XXXXXXXXXX.com via_cnt==2". . # U 2005/08/03 13:55:57.381353 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. Call-ID: b521746f-5119658a(a)192.168.1.2. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a17 9. CSeq: 102 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/linux)). Content-Length: 0. . # U 2005/08/03 13:55:57.664069 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a17 9. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="2810300308",realm="XXXXXXXXXX.com",nonce="42f0a3e8f78926d70014 0a7c490f06dd965ea748",uri="sip:2810318308@XXXXXXXXXX.com",algorithm=MD5, response="ac73134a0de530d527cfbe4fe1903707". Contact: <sip:2810300308@XX.XX.XX.XX:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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Re: [Serusers] lookup 2 uri
by Richard Z 05 Aug '05

05 Aug '05

The issue is that the locationis not always the same. I will try to fool around the register message and save function to see if there is a way to make it work. Richard On 8/3/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > If you know in advance the IP of 1002 and it is static, you can use serctl to add it's contact. This way doing a lookup("location") for 1003 will return both contacts...it's not a nice solution but it whould work. > > Samuel. > &… [View More]gt; Unclassified. > >>> Richard Z <rzheng(a)gmail.com> 08/03/05 08:55PM >>> > Hi Samuel, > > That's a very good detailed explanation! That's exactly what happened. > In my existing config, I have the logic to deal with the second ring > to 1003(a)test.com. It forks to itself and creates another ser > transaction. I'd like to make it simpler by doing it in a single > transaction. > > One way I am thinking about, but haven't tried yet, is when 1003 > registers, ser adds its location to both 1002 and 1003. So when ser > does lookup("location") for 1002, the real location for 1003 shows up. > Not sure that it is feasible. > > Thanks, > Richard > > > > On 8/2/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > Just one remark: only one of the two URI will be translated to the contact. > > > > I'll try to explain... > > > > imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup, > > lookup("aliases"); > > SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch. > > Req-URI: 1002(a)test.com > > branch: 1003(a)test.com > > Doing the second lookup, > > lookup("location"); > > will put the contact of 1002(a)test.com as Req-URI (if more than one contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched. > > Req-URI: 1002(a)10.1.1.102 > > branch: 1003(a)test.com > > This happens because lookup() only takes as input the req-URI. > > > > The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening. > > User 1002 will get the message without any problem. > > > > NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation. > > > > As SER is now coded, above can not be changed (as far as I know).... > > > > Hope it helps, > > > > Samuel. > > > > Unclassified. > > > > >>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>> > > You would need two steps here, looup("aliases"), then lookup("location"). > > > > Zeus > > > > > -----Original Message----- > > > From: serusers-bounces(a)lists.iptel.org > > > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z > > > Sent: Wednesday, 3 August 2005 3:33 AM > > > To: Atle Samuelsen > > > Cc: serusers(a)lists.iptel.org > > > Subject: Re: [Serusers] lookup 2 uri > > > > > > > > > Hi Atle, > > > > > > Alias doesn't replace uri with its real location. I'd like to use > > > lookup("location") to find out its real location. For > > > example, if 102(a)test.com is at 102(a)10.1.1.102 and > > > 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all > > > uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser > > > transaction logic. Would it be possible? > > > > > > Thanks, > > > Richard > > > > > > > > > On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote: > > > > > > > > Hi, > > > > > > > > you could do something like : > > > > > > > > serctl alias add 101 sip:102@test.com > > > > serctl alias add 101 sip:103@test.com > > > > > > > > This way the invite would be forked to both 102 and 103, > > > > > > > > just remember to have both lookup("aliases"); and > > > save("aliases"); in > > > > your config at the right spot :-) > > > > > > > > -Atle > > > > > > > > > > > > > > > > * Richard Z <rzheng(a)gmail.com> [050802 04:34]: > > > > > Hi, > > > > > > > > > > I am trying to implement a feature to ring two ip phones > > > > > simultaneously. For example, when I call 101(a)test.com, it > > > rings both > > > > > 101(a)test.com and 102(a)test.com. In this case, the ruri is > > > > > 101(a)test.com. I can use append_branch 102(a)test.com, but > > > is there a > > > > > way to lookup the locations of both 101 and 102? > > > > > > > > > > Thanks, > > > > > Richard > > > > > > > > > > _______________________________________________ > > > > > 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 > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > [View Less]

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[Serusers] SER and Cisco 1760V /w VIC2 2BRI-NT/TE
by Antanas Masevicius 04 Aug '05

04 Aug '05

Hello, does enyone had any experiance trying to run cisco router with 2BRI-NT/TE voip NIC with SER? I wonder if it is possible at least. I also didn't found how cisco should send sip-password to SER. Antanas

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[Serusers] What is unique to a call in acc database table??
by Eric Haskins 04 Aug '05

04 Aug '05

here are a few sip_callid's from the ACC table they are all from the same user. This is being used in a prepaid outgoing only. 855d2882-32ae277b(a)192.168.1.100 7cd4ca18-5abe85b0(a)192.168.1.100 We are using same ATA adapter for these calls. I just wanted to confirm this is unique to each call or I may have to write something to export info Eric

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[Users] force_rtp_proxy() problem.
by Ray Van Dolson 04 Aug '05

04 Aug '05

Thanks for the help earlier Bogdan. I've gotten things working for the most part, but am having a problem with the force_rtp_proxy function. Once again, here is my setup: SIP ATA - 10.0.201.196 OpenSER - eth0: 208.208.53.5/24 - eth1: 10.0.201.5/24 Asterisk - eth0: 65.234.34.5/24 - eth1: 10.0.201.6/24 Yes, Asterisk and OpenSER are on the same Private network currently. This is just for testing as clients were originally connecting directly to Asterisk. It will not always be the case. … [View More]

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[Serusers] What is unique to a call in acc database table??
by Eric Haskins 04 Aug '05

04 Aug '05

We are currently working on our Billing system and I was looking over our SER database and noticed sip_callid appeared to be unique to each call but when I looked back in the database I saw a bunch that appeared to be the same??? They were notify messages is this common or do I have to devise a different method? We have the hardest part done that calculates call price by Country Code , City Code, customer assigned rate or default Wholesale/Retail rate. We had it completed when running … [View More]

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[Users] PostgreSQL Version of "sc"
by Adam Sherman 04 Aug '05

04 Aug '05

There is a nice little script in sip-server/scripts/sc that allows for managing many of the tables. Has anyone done the necessary tweaking to port it to PostgreSQL? Thanks, A. -- Adam Sherman Technologist

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[Serusers] route loop quandary
by Daniel Poulsen 04 Aug '05

04 Aug '05

I am having a problem with calls that are parallel forked. The problem happens when a call comes in, gets forked using multiple aliases and one UA is unavailable. The question is, how do I handle the offline call leg? I can't do if (!lookup("location")) { break; (or sl_send_reply("404") } because then the whole call dies since each branch of the call uses the same callid. If I t_relay them both, then the fork works but I the offline leg gets repeatedly relayed unltil I get a 483 "Too … [View More]many hops". This scenario would be OK [althought certainly not ideal], except it does not allow me to add a condition to hunt to the next number upon a failed lookup("location"). If I do, the online UA will ring once and the offline UA will get routed immediately to voicemail which picks up after just one ring. How are other handling these problems? I think the answer lies in letting failure_route handle everything, but I don't know how to get that to work because if I don't handle the offline users in the (!lookup("location"), I end up with routing loops and eventualluy a 483. Anybody that has no online contacts whatsoever never makes it to failure_route (where the forward to VM is handled). Below is my ser.cfg. I've been playing around with things like if (isflagset(6)) { route(x) } but I still need some way to handle the failed portion of the fork. Is there a way to "null route" the failed portion or something? route { /* Registration and other logic snipped for brevity */ # Translate local address according to aliases table xlog("L_INFO", "%ci: Preparing to look up alias for %ru\n"); if(lookup("aliases")) { xlog("L_NOTICE", "%ci: alias lookup changed uri to %ru\n"); if (!(uri==myself)) { xlog("L_NOTICE", "%ci: Outbound aliases %ru\n"); route(4); break; }; }; if ( (uri=~"^sip:411@.*") | (uri=~"^sip:911@.*") | (uri=~"^sip:011[0-9]+@.*") | (uri=~"^sip:1[0-9]{10}@.*") ) { route(3); # 3: PSTN with authentication break; }; # Anything left is a local call. xlog("L_INFO", "%ci: Preparing to look up location for %ru\n"); if (!lookup("location")) { xlog("L_NOTICE", "%ci: no location for %ru\n"); setflag(5); } else { # Had a sucessful lookup setflag(6); }; route(4); } route[4] { # If an invitation, we want to hunt on failure if(method == "INVITE") { t_on_failure("1"); # first hunt }; append_hf("P-hint: relay\r\n"); xlog("L_NOTICE", "%ci: r4: t_relay %ru\n"); if (!t_relay()) { xlog("L_NOTICE", "%ci: %ru r4: Relay Failed\n"); sl_reply_error(); break; }; } failure_route[1] { revert_uri(); route(7); } ### Handle hunt sequences. [View Less]

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[Users] Trouble with freshly compiled OpenSer0.9.5
by Martin Kjeldsen 04 Aug '05

04 Aug '05

Hi all I was wondering if any of you might have a solution to my problem! The thing is that i've compiled and installed OpenSer0.9.5 from CVS, on a SUSE 9.2, added MySQL support and created a DB for OpenSer. And now I'm stuck at executing openserctl start and only ending up with the PID filed getting created and ultimately deleted, when OpenSer terminates with the message "PID file /home/ser/var/run/openser.pid does not exist -- OpenSER start failed", and path to the PID has been altered … [View More]in openserctl. I'm using debug mode 5 but cant realy figure out what might be causing this error! All help will be greatly appreciated. I have attached the output: (Sorry for the big post) ser@site1:~/running/openser/sbin> [21Popenserctl start Starting OpenSER : 0(14441) read 2733431526 from /dev/urandom 0(14441) seeding PRNG with 3856599290 0(14441) test random number 930545650 0(14441) loading module /home/ser/running/openser/lib/openser/modules/mysql.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/sl.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/tm.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/rr.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/maxfwd.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/usrloc.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/registrar.so 0(14441) loading module /home/ser/running/openser/lib/openser/modules/textops.so 0(14441) set_mod_param_regex: usrloc matches module usrloc 0(14441) set_mod_param_regex: found <db_url> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(14441) set_mod_param_regex: usrloc matches module usrloc 0(14441) set_mod_param_regex: found <db_mode> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(14441) set_mod_param_regex: rr matches module rr 0(14441) set_mod_param_regex: found <enable_full_lr> in module rr [/home/ser/running/openser/lib/openser/modules/rr.so] 0(14441) find_export: found <mf_process_maxfwd_header> in module maxfwd [/home/ser/running/openser/lib/openser/modules/maxfwd.so] 0(14441) find_export: found <sl_send_reply> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(14441) find_export: found <sl_send_reply> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(14441) find_export: found <record_route> in module rr [/home/ser/running/openser/lib/openser/modules/rr.so] 0(14441) find_export: found <loose_route> in module rr [/home/ser/running/openser/lib/openser/modules/rr.so] 0(14441) find_export: found <append_hf> in module textops [/home/ser/running/openser/lib/openser/modules/textops.so] 0(14441) find_export: found <append_hf> in module textops [/home/ser/running/openser/lib/openser/modules/textops.so] 0(14441) find_export: found <save> in module registrar [/home/ser/running/openser/lib/openser/modules/registrar.so] 0(14441) find_export: found <lookup> in module registrar [/home/ser/running/openser/lib/openser/modules/registrar.so] 0(14441) find_export: found <append_hf> in module textops [/home/ser/running/openser/lib/openser/modules/textops.so] 0(14441) find_export: found <lookup> in module registrar [/home/ser/running/openser/lib/openser/modules/registrar.so] 0(14441) find_export: found <sl_send_reply> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(14441) find_export: found <append_hf> in module textops [/home/ser/running/openser/lib/openser/modules/textops.so] 0(14441) find_export: found <t_relay> in module tm [/home/ser/running/openser/lib/openser/modules/tm.so] 0(14441) find_export: found <sl_reply_error> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(14441) routing table 0: 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x4027ffd0> 0(14441) , "10" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027ffd0> 0(14441) , "10" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "483" 0(14441) , "Too Many Hops" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) UNKNOWN 0(14441) <UNKNOWN> 0(14441) 2048 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "513" 0(14441) , "Message too big" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4027c7c0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: rr-enforced " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) UNKNOWN 0(14441) <UNKNOWN> 0(14441) 2048 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "513" 0(14441) , "Message too big" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4027c7c0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: rr-enforced " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4027c7c0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: rr-enforced " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4027a1c0> 0(14441) ); 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: rr-enforced " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) if ( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) { 0(14441) if ( 0(14441) method 0(14441) == 0(14441) "REGISTER" 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x4029bb30> 0(14441) , "location" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "aliases" 0(14441) ); 0(14441) if ( 0(14441) NOT( 0(14441) uri 0(14441) == 0(14441) _myself_ 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: outbound alias " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x40298490> 0(14441) , "location" 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170910> 0(14441) , "404" 0(14441) , "Not Found" 0(14441) ); 0(14441) drop( 0(14441) type<0> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) } else { 0(14441) }; 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x402a1450> 0(14441) , "P-hint: usrloc applied " 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) route( 0(14441) 1 0(14441) ); 0(14441) 0(14441) routing table 1: 0(14441) if ( 0(14441) NOT( 0(14441) external_module_call( 0(14441) f_ptr<0x4024c740> 0(14441) ); 0(14441) external_module_call( 0(14441) f_ptr<0x4024c740> 0(14441) ); 0(14441) ) 0(14441) ) { 0(14441) external_module_call( 0(14441) f_ptr<0x40170580> 0(14441) ); 0(14441) } else { 0(14441) }; 0(14441) Listening on udp: 212.97.202.73 [212.97.202.73]:5060 tcp: 212.97.202.73 [212.97.202.73]:5060 Aliases: tcp: osite1:5060 tcp: osite1.cuatro.dk:5060 udp: osite1:5060 udp: osite1.cuatro.dk:5060 WARNING: no fork mode 0(14441) shm_mem_init: success 0(0) core dump limits set to 4294967295 0(0) DEBUG: init_mod: sl_module stateless - initializing 0(0) DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(0) unixsock_register_cmd: New command (sl_stats) registered 0(0) DEBUG: MD5 calculated: ba950484b7cb9e0b78ff4f84c7272d40 0(0) DEBUG: init_mod: tm 0(0) TM - (size of cell=2780, sip_msg=760) initializing... 0(0) Call-ID initialization: '773ffe9e' 0(0) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(0) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered 0(0) DEBUG: register_fifo_cmd: new command (t_hash) registered 0(0) DEBUG: register_fifo_cmd: new command (t_reply) registered 0(0) unixsock_register_cmd: New command (t_uac_dlg) registered 0(0) unixsock_register_cmd: New command (t_uac_cancel) registered 0(0) unixsock_register_cmd: New command (t_hash) registered 0(0) unixsock_register_cmd: New command (t_reply) registered 0(0) DEBUG: lock_initialize: lock initialization started 0(0) DEBUG: register_fifo_cmd: new command (t_stats) registered 0(0) unixsock_register_cmd: New command (t_stats) registered 0(0) DEBUG: MD5 calculated: 5818e8718167640aef9d06ecea0d94fe 0(0) DEBUG: MD5 calculated: 55629ad9d6de7376eb3d3c6f0e1dea9c 0(0) DEBUG: init_mod: rr 0(0) rr - initializing 0(0) DEBUG: init_mod: maxfwd 0(0) Maxfwd module- initializing 0(0) DEBUG: init_mod: usrloc 0(0) usrloc - initializing 0(0) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(0) unixsock_register_cmd: New command (ul_stats) registered 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) find_mod_export: found <db_use_table> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_init> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_close> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_query> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_raw_query> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_free_result> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_insert> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_delete> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) find_mod_export: found <db_update> in module mysql [/home/ser/running/openser/lib/openser/modules/mysql.so] 0(0) DEBUG: init_mod: registrar 0(0) registrar - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/home/ser/running/openser/lib/openser/modules/sl.so] 0(0) find_export: found <ul_bind_usrloc> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_register_udomain> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_get_all_ucontacts> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_insert_urecord> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_delete_urecord> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_get_urecord> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_lock_udomain> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_unlock_udomain> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_release_urecord> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_insert_ucontact> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_delete_ucontact> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_get_ucontact> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_update_ucontact> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_register_watcher> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_unregister_watcher> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) find_export: found <ul_register_ulcb> in module usrloc [/home/ser/running/openser/lib/openser/modules/usrloc.so] 0(0) DEBUG: init_mod: textops textops - initializing 0(0) fixing /home/ser/running/openser/lib/openser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /home/ser/running/openser/lib/openser/modules/sl.so sl_send_reply 0(0) fixing /home/ser/running/openser/lib/openser/modules/sl.so sl_send_reply 0(0) fixing /home/ser/running/openser/lib/openser/modules/rr.so record_route 0(0) fixing /home/ser/running/openser/lib/openser/modules/rr.so loose_route 0(0) fixing /home/ser/running/openser/lib/openser/modules/textops.so append_hf 0(0) fixing /home/ser/running/openser/lib/openser/modules/textops.so append_hf 0(0) fixing /home/ser/running/openser/lib/openser/modules/registrar.so save 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain(): Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /home/ser/running/openser/lib/openser/modules/registrar.so lookup 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain(): Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /home/ser/running/openser/lib/openser/modules/textops.so append_hf 0(0) fixing /home/ser/running/openser/lib/openser/modules/registrar.so lookup 0(0) fixing /home/ser/running/openser/lib/openser/modules/sl.so sl_send_reply 0(0) fixing /home/ser/running/openser/lib/openser/modules/textops.so append_hf 0(0) fixing /home/ser/running/openser/lib/openser/modules/tm.so t_relay 0(0) fixing /home/ser/running/openser/lib/openser/modules/sl.so sl_reply_error 0(0) INFO: udp_init: SO_RCVBUF is initially 111616 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 225280 0(0) DEBUG: setting SO_RCVBUF; set=225280,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 227328 0(0) DEBUG: setting SO_RCVBUF; set=227328,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 229376 0(0) DEBUG: setting SO_RCVBUF; set=229376,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 231424 0(0) DEBUG: setting SO_RCVBUF; set=231424,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 233472 0(0) DEBUG: setting SO_RCVBUF; set=233472,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 235520 0(0) DEBUG: setting SO_RCVBUF; set=235520,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 237568 0(0) DEBUG: setting SO_RCVBUF; set=237568,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 239616 0(0) DEBUG: setting SO_RCVBUF; set=239616,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 241664 0(0) DEBUG: setting SO_RCVBUF; set=241664,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 243712 0(0) DEBUG: setting SO_RCVBUF; set=243712,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 245760 0(0) DEBUG: setting SO_RCVBUF; set=245760,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 247808 0(0) DEBUG: setting SO_RCVBUF; set=247808,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 249856 0(0) DEBUG: setting SO_RCVBUF; set=249856,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 251904 0(0) DEBUG: setting SO_RCVBUF; set=251904,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 253952 0(0) DEBUG: setting SO_RCVBUF; set=253952,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 256000 0(0) DEBUG: setting SO_RCVBUF; set=256000,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 258048 0(0) DEBUG: setting SO_RCVBUF; set=258048,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 260096 0(0) DEBUG: setting SO_RCVBUF; set=260096,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 262144 0(0) DEBUG: setting SO_RCVBUF; set=262144,verify=262142 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/openser_fifo 0(0) DEBUG: fifo /tmp/openser_fifo opened, mode=777 0(0) ERROR: init_fifo_server: fifo_write did not open: Permission denied 0(0) initializing fifo server failed 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(14441) shm_mem_destroy 0(14441) destroying the shared memory lock PID created! total 0 PID file /home/ser/var/run/openser.pid does not exist -- OpenSER start failed Kind regards Martin [View Less]

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[Users] regexp problem
by Klaus Darilion 04 Aug '05

04 Aug '05

Hi! I have a problem with search(); I try to match a contact header with IP address: if !(search("Contact:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+:")) { works fine, whereas if !(search("Contact:.*@[0-9]+\.[0-9]+\.[0-9]+\.\d+:")) { does not work. ^^ Looks like search() does not like \d Are there any known limitations for the regular expressions? regards klaus

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RE: SV: [Serusers] SER core dump on SuSE
by Pat wang 04 Aug '05

04 Aug '05

Thanks Roger, You are right. I reinstalled SuSE with all the available development packages cause I am not sure what else is needed if there are always something missing. SER works fine with authentication now. Thanks, Pat >From: "Roger Lewau" <roger.lewau(a)serverhallen.com> >To: "'Pat wang'" <wangyu39(a)hotmail.com>,<serusers(a)lists.iptel.org> >Subject: SV: [Serusers] SER core dump on SuSE >Date: Wed, 3 Aug 2005 19:01:08 +0200 > >Hi, > >Did you … [View More]install a precompiled version of SER and modules? I had the exact >same problem on FreeBSD but solved it by downloading the source and compile >the mysql.so module by hand and then replacing the precompiled module. > >Dont know the exact reason for the crash I had, but the above step solved >it. Could be woth a try. > >Kind regards >Roger > > >-----Ursprungligt meddelande----- >Från: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] För >Pat >wang >Skickat: den 3 augusti 2005 17:53 >Till: serusers(a)lists.iptel.org >Ämne: [Serusers] SER core dump on SuSE > >Hi, > >I am having trouble running SER on SuSE Linux. SER crashes when >authentication module is used. Whenever a REGISTER arrives to SER, it gives >a core dump. If I turn authentication off then everything is fine. I have >tried this on different versions of SuSE including: 9.0 (2.4 Kernel) and >9.1 >and 9.3 (2.6 Kernel). They all give this core dump: > >Core was generated by `ser -E -w .'. >Program terminated with signal 11, Segmentation fault. > >warning: current_sos: Can't read pathname for load map: Input/output error > >Reading symbols from /lib/libdl.so.2...done. >Loaded symbols for /lib/libdl.so.2 >Reading symbols from /lib/libresolv.so.2...done. >Loaded symbols for /lib/libresolv.so.2 >Reading symbols from /lib/tls/libc.so.6...done. >Loaded symbols for /lib/tls/libc.so.6 >Reading symbols from /lib/ld-linux.so.2...done. >Loaded symbols for /lib/ld-linux.so.2 >Reading symbols from /usr/local/lib/ser/modules/mysql.so...done. >Loaded symbols for /usr/local/lib/ser/modules/mysql.so >Reading symbols from /usr/lib/libmysqlclient.so.12...done. >Loaded symbols for /usr/lib/libmysqlclient.so.12 Reading symbols from >/lib/libz.so.1...done. >Loaded symbols for /lib/libz.so.1 >Reading symbols from /lib/libcrypt.so.1...done. >Loaded symbols for /lib/libcrypt.so.1 >Reading symbols from /lib/libnsl.so.1...done. >Loaded symbols for /lib/libnsl.so.1 >Reading symbols from /lib/tls/libm.so.6...done. >Loaded symbols for /lib/tls/libm.so.6 >Reading symbols from /usr/local/lib/ser/modules/sl.so...done. >Loaded symbols for /usr/local/lib/ser/modules/sl.so Reading symbols from >/usr/local/lib/ser/modules/tm.so...done. >Loaded symbols for /usr/local/lib/ser/modules/tm.so Reading symbols from >/usr/local/lib/ser/modules/rr.so...done. >Loaded symbols for /usr/local/lib/ser/modules/rr.so Reading symbols from >/usr/local/lib/ser/modules/maxfwd.so...done. >Loaded symbols for /usr/local/lib/ser/modules/maxfwd.so >Reading symbols from /usr/local/lib/ser/modules/usrloc.so...done. >Loaded symbols for /usr/local/lib/ser/modules/usrloc.so >Reading symbols from /usr/local/lib/ser/modules/registrar.so...done. >Loaded symbols for /usr/local/lib/ser/modules/registrar.so >Reading symbols from /usr/local/lib/ser/modules/textops.so...done. >Loaded symbols for /usr/local/lib/ser/modules/textops.so >Reading symbols from /usr/local/lib/ser/modules/auth.so...done. >Loaded symbols for /usr/local/lib/ser/modules/auth.so >Reading symbols from /usr/local/lib/ser/modules/auth_db.so...done. >Loaded symbols for /usr/local/lib/ser/modules/auth_db.so >Reading symbols from /lib/libnss_files.so.2...done. >Loaded symbols for /lib/libnss_files.so.2 #0 qm_malloc (qm=0x80ffac0, >size=20) at q_malloc.c:290 >290 q_malloc.c: No such file or directory. > in q_malloc.c >(gdb) >(gdb) >(gdb) >(gdb) >(gdb) bt >#0 qm_malloc (qm=0x80ffac0, size=20) at q_malloc.c:290 >#1 0x4016322c in new_result () at res.c:199 >#2 0x40162035 in store_result (_h=0x8106138, _r=0xbfffee4c) at dbase.c:298 >#3 0x080998ec in table_version (dbf=0x4039b320, connection=0x8106138, >table=0xbfffeeb0) > at db.c:176 >#4 0x40398eaf in auth_db_ver (db_url=0x0, name=0x0) at authorize.c:337 >#5 0x40397fba in str_fixup (param=0x8106c30, param_no=2) at >authdb_mod.c:238 >#6 0x0806ec55 in fix_actions (a=<value optimized out>) at route.c:243 >#7 0x0806f136 in fix_expr (exp=0x8106c48) at route.c:134 >#8 0x0806f17e in fix_expr (exp=0x8106c78) at route.c:99 >#9 0x0806ed1b in fix_actions (a=<value optimized out>) at route.c:221 #10 >0x0806ed3e in fix_actions (a=<value optimized out>) at route.c:225 >#11 0x0806ed3e in fix_actions (a=<value optimized out>) at route.c:225 >#12 0x0806f026 in fix_rls () at route.c:726 >#13 0x0805df35 in main (argc=4, argv=0xbffff1b4) at main.c:1558 >(gdb) > >I have to use Redhat playing SER now. I am really appreciate if someone can >take a look at the cause. By the way, I tried to install SER from binary >package as well as compile from the source code, result is the same. > >Regards, > >Pat > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > [View Less]

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[Serusers] Segmentation fault-fifo_server fgets failed: Illegal seek
by Ashutosh Kumar 04 Aug '05

04 Aug '05

Hi , I am getting a segmentation fault when my sip server runs. It happens mostly when a client tries to call another client phone. I think that this has got something to do with ser_fifo file permissions. For that reason, the relevant code from my ser.cfg file is as belows; --------------- fifo="/tmp/ser_fifo" fifo_mode=0666 --------------- I am running SER-0.8.14.The debug trace is as produced below: 0(3972) check_self - checking if host==us: 15==15 && [sip.myownphone.com] ==… [View More] [sip. myownphone.com] 0(3972) check_self - checking if port 5060 matches port 5060 0(3972) rwrite(): Rewriting Request-URI with 'sip:satya@210.7.75.189:5060' 0(3972) DEBUG: t_addifnew: msg id=3 , global msg id=1 , T on entrance=0xffffffff 0(3972) parse_headers: flags=-1 0(3972) parse_headers: flags=60 0(3972) t_lookup_request: start searching: hash=13553, isACK=0 0(3972) DEBUG: RFC3261 transaction matching failed 0(3972) DEBUG: t_lookup_request: no transaction found 0(3972) DBG: callback type 2, id 3 entered 0(3972) DEBUG: t_lookupOriginalT: searching on hash entry 13553 0(3972) DEBUG: RFC3261 transaction matching failed 0(3972) DEBUG: t_lookupOriginalT: no CANCEL maching found! 0(3972) DEBUG: t_lookupOriginalT completed 0(3972) DEBUG: mk_proxy: doing DNS lookup... 0(3972) check_via_address(61.16.236.34, 192.168.11.37, 0) 2(3975) ERROR: fifo_server fgets failed: Illegal seek 2(3975) ERROR: fifo_server fgets failed: Illegal seek 2(3975) ERROR: fifo_server fgets failed: Illegal seek 2(3975) ERROR: fifo_server fgets failed: Illegal seek 2(3975) INFO: signal 15 received 2(3975) Memory status (pkg): 2(3975) fm_status (0x80c2c80): 2(3975) heap size= 1047440 2(3975) dumping free list: 2(3975) hash = 1 fragments no.: 11, bucket size: 8 - 8 (first 8) 2(3975) hash = 2 fragments no.: 2, bucket size: 16 - 16 (first 16) 2(3975) hash = 4 fragments no.: 1, bucket size: 32 - 32 (first 32) 2(3975) hash = 15 fragments no.: 1, bucket size: 120 - 120 (first 120) 2(3975) hash = 20 fragments no.: 3, bucket size: 160 - 160 (first 160) 2(3975) hash = 265 fragments no.: 1, bucket size: 524288 - 1048576 (first 1030944) 2(3975) TOTAL: 19 free fragments = 1031696 free bytes 2(3975) ----------------------------- 1(3974) INFO: signal 15 received 1(3974) Memory status (pkg): 1(3974) fm_status (0x80c2c80): 1(3974) heap size= 1047440 1(3974) dumping free list: 1(3974) hash = 1 fragments no.: 11, bucket size: 8 - 8 (first 8) 1(3974) hash = 2 fragments no.: 1, bucket size: 16 - 16 (first 16) 1(3974) hash = 4 fragments no.: 1, bucket size: 32 - 32 (first 32) 1(3974) hash = 6 fragments no.: 1, bucket size: 48 - 48 (first 48) 1(3974) hash = 8 fragments no.: 1, bucket size: 64 - 64 (first 64) 1(3974) hash = 20 fragments no.: 3, bucket size: 160 - 160 (first 160) 1(3974) hash = 32 fragments no.: 1, bucket size: 256 - 256 (first 256) 1(3974) hash = 78 fragments no.: 1, bucket size: 624 - 624 (first 624) 1(3974) hash = 265 fragments no.: 1, bucket size: 524288 - 1048576 (first 1030048) 1(3974) TOTAL: 21 free fragments = 1031656 free bytes 1(3974) ----------------------------- Segmentation fault Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com [View Less]

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[Serusers] serweb 0.9.3 config on RedHat ES4
by Amit Rastogi 04 Aug '05

04 Aug '05

Hello, I am trying to install serweb 0.9.3 release. I already have ser 0.9.3 running with mysql support. When i try to access the http:<host>/serweb/admin.php, it displays a blank page and nothing else.. Here are the steps i have done so far.. untar package to <root>/serweb directory copy ./html to /var/www/html/serweb directory copy ./phplib to /var/www/phplib directory modified php.ini to have short_open_tag to on Added alias in httpd.conf as Alias /serweb/ "/var/www/html/… [View More]

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Re: [Serusers] Implementing Codec Translator with Asterisk
by Noel Sharpe 04 Aug '05

04 Aug '05

Frank Kostin wrote: > > BTW, I heard that someone might need a licence for G.729 !? It's a proprietory codec, so yes you need a licence for each and every channel you transcode. If you want to support the transcoding of 100 simultaneous conversations, you need 100 licences. You don't need a licence to just use G729; you only need licences when you need to encode or decode using G729. If you use asterisk, you can purchase a licence from Digium.

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Re: [Serusers] lookup 2 uri
by Samuel Osorio Calvo 04 Aug '05

04 Aug '05

If you know in advance the IP of 1002 and it is static, you can use serctl to add it's contact. This way doing a lookup("location") for 1003 will return both contacts...it's not a nice solution but it whould work. Samuel. Unclassified. >>> Richard Z <rzheng(a)gmail.com> 08/03/05 08:55PM >>> Hi Samuel, That's a very good detailed explanation! That's exactly what happened. In my existing config, I have the logic to deal with the second ring to 1003(a)test.com. It forks … [View More]to itself and creates another ser transaction. I'd like to make it simpler by doing it in a single transaction. One way I am thinking about, but haven't tried yet, is when 1003 registers, ser adds its location to both 1002 and 1003. So when ser does lookup("location") for 1002, the real location for 1003 shows up. Not sure that it is feasible. Thanks, Richard On 8/2/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Just one remark: only one of the two URI will be translated to the contact. > > I'll try to explain... > > imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup, > lookup("aliases"); > SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch. > Req-URI: 1002(a)test.com > branch: 1003(a)test.com > Doing the second lookup, > lookup("location"); > will put the contact of 1002(a)test.com as Req-URI (if more than one contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched. > Req-URI: 1002(a)10.1.1.102 > branch: 1003(a)test.com > This happens because lookup() only takes as input the req-URI. > > The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening. > User 1002 will get the message without any problem. > > NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation. > > As SER is now coded, above can not be changed (as far as I know).... > > Hope it helps, > > Samuel. > > Unclassified. > > >>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>> > You would need two steps here, looup("aliases"), then lookup("location"). > > Zeus > > > -----Original Message----- > > From: serusers-bounces(a)lists.iptel.org > > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z > > Sent: Wednesday, 3 August 2005 3:33 AM > > To: Atle Samuelsen > > Cc: serusers(a)lists.iptel.org > > Subject: Re: [Serusers] lookup 2 uri > > > > > > Hi Atle, > > > > Alias doesn't replace uri with its real location. I'd like to use > > lookup("location") to find out its real location. For > > example, if 102(a)test.com is at 102(a)10.1.1.102 and > > 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all > > uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser > > transaction logic. Would it be possible? > > > > Thanks, > > Richard > > > > > > On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote: > > > > > > Hi, > > > > > > you could do something like : > > > > > > serctl alias add 101 sip:102@test.com > > > serctl alias add 101 sip:103@test.com > > > > > > This way the invite would be forked to both 102 and 103, > > > > > > just remember to have both lookup("aliases"); and > > save("aliases"); in > > > your config at the right spot :-) > > > > > > -Atle > > > > > > > > > > > > * Richard Z <rzheng(a)gmail.com> [050802 04:34]: > > > > Hi, > > > > > > > > I am trying to implement a feature to ring two ip phones > > > > simultaneously. For example, when I call 101(a)test.com, it > > rings both > > > > 101(a)test.com and 102(a)test.com. In this case, the ruri is > > > > 101(a)test.com. I can use append_branch 102(a)test.com, but > > is there a > > > > way to lookup the locations of both 101 and 102? > > > > > > > > Thanks, > > > > Richard > > > > > > > > _______________________________________________ > > > > 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 > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > [View Less]

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[Serusers] t_uac_dlg pb
by mma 04 Aug '05

04 Aug '05

I want to send an invite request to ser's fifo using t_uac_dlg command and exec_msg, my pb is that i can't construct a valid invite header , when i check the log of ser, i see that there is errors in parsing headers. i'm trying to send something like this : cat > /tmp/ser_fifo << EOF :t_uac_dlg:hh INVITE sip: user2(a)sip.server.net . From: sip:user1@sip.server.net To: sip:user2@sip.server.net . Content-Type: application/sdp EOF please can someone correct it. Regards PS : the … [View More]

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[Serusers] No stop record for pstn calls
by Anders Hansson 04 Aug '05

04 Aug '05

Hi all, I have a problem accounting transactions to the PSTN network. I do only get the Stop record - no Start record... Anyone knows what could be the reason? I´m using radiusserver for the accounting records. This is part of my ser.cfg file. In the log the messages INVITE accounted and BYE or CANCEL accounted can be seen, but nothing in the radius file, except for the stop record. To make it worse, not all of the stop records can be seen in radius - it looks like only some destinations … [View More]

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[Serusers] Problem with SER and PAP2, I get a 407 message when trying a call
by Yiannis Martis 04 Aug '05

04 Aug '05

Hello there, I am Network Administrator working for a company called Hellasfon Network S.A. I use SER 0.9.3 with Vovidas STUN server. PAP2 UAis NATed and gets registered using my ADSL modems IP and everything looks OK. The problem is when I try to make a call to another UA) of my service I get the following output from my server (ngrep):. Please help me if u can!!! U 2005/08/03 13:55:57.380953 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 INVITE sip:2810318308@XXXXXXXXXX.com SIP/2.0. … [View More] Record-Route: <sip:XX.XX.XX.XX;ftag=7ab32f2bbf2689feo0;lr=on>. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Max-Forwards: 16. Contact: <sip:2810300308@XX.XX.XX.XX:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 420. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 13486 13486 IN IP4 XX.XX.XX.XX. s=-. c=IN IP4 XX.XX.XX.XX. t=0 0. m=audio 16420 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/08/03 13:55:57.381212 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="XXXXXXXXXX.com", nonce="42f0a3e9ea677d46038bdbeccaf8d95c18bce58e". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=4360 req_src_ip=XX.XX.XX.XX req_src_port=5060 in_uri=sip:2810318308@XXXXXXXXXX.com out_uri=sip:2810318308@XXXXXXXXXX.com via_cnt==2". . # U 2005/08/03 13:55:57.381353 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. Call-ID: b521746f-5119658a(a)192.168.1.2. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. CSeq: 102 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/linux)). Content-Length: 0. . # U 2005/08/03 13:55:57.664069 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="2810300308",realm="XXXXXXXXXX.com",nonce="42f0a3e8f78926d700140a7c490f06dd965ea748",uri="sip:2810318308@XXXXXXXXXX.com",algorithm=MD5,response="ac73134a0de530d527cfbe4fe1903707". Contact: <sip:2810300308@XX.XX.XX.XX:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ [View Less]

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RE: RE : [Serusers] is SER Dependable for prepaid System ? Anyone usingitfor prepaid system?
by Sam Lee 04 Aug '05

04 Aug '05

Dear all, How does the b2bua and the Jradius work ? It's the first time I've seen it. Does the billing system rely on radius ? I already have a SER running on digest authentication. How do I integrate them with this prepaid system? Regards, Sam -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Olivier Taylor Sent: Thursday, August 04, 2005 12:00 AM To: 'Iqbal'; 'Bruno Lopes F. Cabral' Cc: serusers(a)lists.iptel.org Subject: … [View More]

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RE: RE : [Serusers] is SER Dependable for prepaid System ? Anyone usingitfor prepaid system?
by Sam Lee 04 Aug '05

04 Aug '05

-----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Olivier Taylor Sent: Thursday, August 04, 2005 12:00 AM To: 'Iqbal'; 'Bruno Lopes F. Cabral' Cc: serusers(a)lists.iptel.org Subject: RE : [Serusers] is SER Dependable for prepaid System ? Anyone usingitfor prepaid system? With Jradius, we disconnect the user when credit limit is reached. Olivier -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-… [View More]

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[Users] Mediaproxy, fix_contact, and ip:port checks
by Kanakatti Mahesh Subramanya 03 Aug '05

03 Aug '05

Currently, fix_contact immediately returns if src_ip == contact_ip However, there are a bunch-a situations where src_ip==contact_ip, but src_port != contact_port (Dont even ask. Just blame the *really* bizarre cable routers that Comcast deploys every now and then) I *could* go about modifying FixContact in functions.c to check both the ip and port. Question is, should I? Is there something I am missing as to why we check only the IP? Should I be explicitly *not* checking the port? Cheers

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[Serusers] pstn -> ser -> asterisk voicemail
by Jack Wei 03 Aug '05

03 Aug '05

i'm having a minor problem with pstn to ser forwarding to asterisk voicemail. when ser forwards the call to asterisk, the destination ua keeps ringing while the regular phone hears the voicemail. this doesn't happen when it's a ip-to-ip call. anyone knows why this is happening? ser.cfg ================= route { ... if ( lookup( "location" ) ) # Destination UA is online { xlog("L_INFO", "%ru is online (in … [View More]

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[Serusers] UDP
by Sebastian Kühner 03 Aug '05

03 Aug '05

Hi! I'm getting INVITES from another Cisco server, but my ser isn't catching it. The difference between a "normal message" (from one of our phones) is, that a dump shows us "UDP" like protocol (Cisco)... and SIP from our phones. Do I have to respect different SIP signallings?? I made a log on "INVITES"... but ser isn't catching it :-( Thanks! Sebastian

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[Serusers] RTPProxy issues with G729 at different frame lengths
by Noel Sharpe 03 Aug '05

03 Aug '05

Hi I am have some difficulties using the latest RTPProxy. Everything seems to work fine unless you use G729 codec with 10ms frame lengths. Has anyone else experienced this issue, and does anyone have any suggestions to work around the issue? Kind Regards Noel

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Re: [Serusers] lookup 2 uri
by Richard Z 03 Aug '05

03 Aug '05

Hi Samuel, That's a very good detailed explanation! That's exactly what happened. In my existing config, I have the logic to deal with the second ring to 1003(a)test.com. It forks to itself and creates another ser transaction. I'd like to make it simpler by doing it in a single transaction. One way I am thinking about, but haven't tried yet, is when 1003 registers, ser adds its location to both 1002 and 1003. So when ser does lookup("location") for 1002, the real location for 1003 shows up. … [View More]Not sure that it is feasible. Thanks, Richard On 8/2/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Just one remark: only one of the two URI will be translated to the contact. > > I'll try to explain... > > imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup, > lookup("aliases"); > SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch. > Req-URI: 1002(a)test.com > branch: 1003(a)test.com > Doing the second lookup, > lookup("location"); > will put the contact of 1002(a)test.com as Req-URI (if more than one contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched. > Req-URI: 1002(a)10.1.1.102 > branch: 1003(a)test.com > This happens because lookup() only takes as input the req-URI. > > The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening. > User 1002 will get the message without any problem. > > NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation. > > As SER is now coded, above can not be changed (as far as I know).... > > Hope it helps, > > Samuel. > > Unclassified. > > >>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>> > You would need two steps here, looup("aliases"), then lookup("location"). > > Zeus > > > -----Original Message----- > > From: serusers-bounces(a)lists.iptel.org > > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z > > Sent: Wednesday, 3 August 2005 3:33 AM > > To: Atle Samuelsen > > Cc: serusers(a)lists.iptel.org > > Subject: Re: [Serusers] lookup 2 uri > > > > > > Hi Atle, > > > > Alias doesn't replace uri with its real location. I'd like to use > > lookup("location") to find out its real location. For > > example, if 102(a)test.com is at 102(a)10.1.1.102 and > > 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all > > uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser > > transaction logic. Would it be possible? > > > > Thanks, > > Richard > > > > > > On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote: > > > > > > Hi, > > > > > > you could do something like : > > > > > > serctl alias add 101 sip:102@test.com > > > serctl alias add 101 sip:103@test.com > > > > > > This way the invite would be forked to both 102 and 103, > > > > > > just remember to have both lookup("aliases"); and > > save("aliases"); in > > > your config at the right spot :-) > > > > > > -Atle > > > > > > > > > > > > * Richard Z <rzheng(a)gmail.com> [050802 04:34]: > > > > Hi, > > > > > > > > I am trying to implement a feature to ring two ip phones > > > > simultaneously. For example, when I call 101(a)test.com, it > > rings both > > > > 101(a)test.com and 102(a)test.com. In this case, the ruri is > > > > 101(a)test.com. I can use append_branch 102(a)test.com, but > > is there a > > > > way to lookup the locations of both 101 and 102? > > > > > > > > Thanks, > > > > Richard > > > > > > > > _______________________________________________ > > > > 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 > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > [View Less]

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[Serusers] Need to do load balancing with two ser servers.
by Prajyot 03 Aug '05

03 Aug '05

Hi, We are using ser v 0.8.14, we are looking for a solution in which I can use two or more SER servers to do load balancing like a cluster design. Any ideas how can I achieve this? Thanks in advance. Regards, Prajyot Shirsolkar

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[Serusers] Record-Route/Via headers not added when protocol=TCP
by Erich Nahum 03 Aug '05

03 Aug '05

Howdy, I'm playing with SER 0.9.3 with sipp as the workload generator on a bunch of linux boxes. When the transport is UDP, openser adds the record-route and via headers like it is supposed to as a proxy. But when I use TCP these headers are not added. This is a UAC->proxy->UAS scenario. I'm attaching ethereal dumps of the UDP-based UAC, TCP-based UAC, UDP-based UAS, and TCP-based UAS for reference. Any suggestions appreciated. Thanks, -Erich -- Erich M. Nahum … [View More]

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RE: [Serusers] is SER Dependable for prepaid System ? Anyone using itfor prepaid system?
by Sam Lee 03 Aug '05

03 Aug '05

I have been working for quite some time on deploying SER as a prepaid system. Currently it is not done yet as there's still quite a number of things to be done. Probably we can share a few ideas down the road. Regards, Sam -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Sanjim Sanjim Sent: Tuesday, August 02, 2005 7:02 PM To: serusers(a)lists.iptel.org Subject: [Serusers] is SER Dependable for prepaid System ? Anyone using … [View More]

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[Serusers] Call Quality
by John Clements 03 Aug '05

03 Aug '05

I am using SER 0.9.3, MediaProxy 1.4.2 and sometimes the call quality gets pretty bad. Parts of the conversation drops and sometimes there will be a delay of 10 or 15 seconds. The SER server has a public IP and is not behind a firewall. The UAs are spread out at different locations behind different firewalls. Sometimes it works perfect though, with a 45min conversation having no problems whatsoever. Does anyone have any suggestions on what I might be able to do to track down/fix this? -John

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[Users] check_from
by Timur Irmatov 03 Aug '05

03 Aug '05

Hi! I am new to SER, and have following question. How can I provide that: 1) Users can register only their sip addresses; 2) Users can use their username in sip From: field. In my installation sip username and authentication usernames differ a little. For example, sip username could be 112233, but auth username is sip112233. So, I want to be sure that user that can successfully autheticate as sip112233 will not use 345678 username, and always will be using 112233 in From? I need something … [View More]

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[Serusers] Routing
by Sebastian Kühner 03 Aug '05

03 Aug '05

Hi! Has anyone of you tried to use the database to make routing? I need to get information of the database and then rewritehost with the result... Thanks! Sebastian

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Re: [Serusers] UDP
by Sebastian Kühner 03 Aug '05

03 Aug '05

Hello, Yes, the INVITE goes to PORT 5060. Here is my ngrep: U 2005/08/02 15:43:56.584366 xxx.xxx.xxx.xxx:51403 -> xxx.xxx.xxx.xxx:5060 INVITE sip:10099999999999@xxx.xxx.xxx.xxx:5060 SIP/2.0. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:OutTrunkGroup". From: "anonymous" <sip:54@xxx.xxx.xxx.xxx>;tag=CEA94420-184A. To: <sip:10099999999999@xxx.xxx.xxx.xxx>. Date: Tue, 02 Aug 2005 18:43:56 GMT. Call-ID: 375AFBE2-2BC11DA-B615D293-1A0569D1(a)xxx.xxx.xxx.xxx. Supported: … [View More]

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[Serusers] Implementing Codec Translator with Asterisk
by Frank Kostin 03 Aug '05

03 Aug '05

Hi everybody, Looking to implement Codec Translator with Asterisk - loading codec modules API in Asterisk to support transcoding. Has anyone experienced this issue, and does anyone have any suggestions or hint, simple scripts, whatever ? Thanks in advance and kind Regards, Frank --------------------------------- Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

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[Users] Radius or SQL
by Jayesh Nambiar 03 Aug '05

03 Aug '05

Hi Bogdan, Thanx for the information about using allow_trusted. It works fine. I just wanted to clarify one thing, is using SQL more efficient or Radius is better. I wanted to know performance wise. What would be the overhead in both cases. one more thing, whenever we add a new IP address to the trusted table openser needs to be restarted. Is there any alternative to this. Which type of calls would be affected when we restart openser. please let me know some information about this. Thanx a lot. Jayesh

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[Users] more TCP problems
by Erich Nahum 03 Aug '05

03 Aug '05

Thanks to Bogdan, who pointed out that my UAC was not behaving properly. Having solved that problem in sipp, I now have a new issue. The UAC is opening a TCP connection to the proxy, but the proxy then forwards the INVITE to the UAS over UDP. I'm using the stock openser.cfg file. Any suggestions? Thanks, -Erich -- Erich M. Nahum IBM T.J. Watson Research Center Research Staff Member P.O. Box 704 nahum(a)watson.ibm.com Yorktown Heights NY 10598

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[Serusers] 407 Proxy Authentication Required
by Yiannis Martis 03 Aug '05

03 Aug '05

Hello there, I have installed ser 0.9.3 with vovida stun server and I use Linksys PAP2 UAs. This is what I get with ngrep when I try to make an IP-to-IP call to another UA: Please Help me... U 2005/08/03 13:55:57.380953 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 INVITE sip:2810318308@XXXXXXXXXX.com SIP/2.0. Record-Route: <sip:XX.XX.XX.XX;ftag=7ab32f2bbf2689feo0;lr=on>. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;… [View More]rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Max-Forwards: 16. Contact: <sip:2810300308@XX.XX.XX.XX:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 420. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 13486 13486 IN IP4 XX.XX.XX.XX. s=-. c=IN IP4 XX.XX.XX.XX. t=0 0. m=audio 16420 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/08/03 13:55:57.381212 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="XXXXXXXXXX.com", nonce="42f0a3e9ea677d46038bdbeccaf8d95c18bce58e". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=4360 req_src_ip=XX.XX.XX.XX req_src_port=5060 in_uri=sip:2810318308@XXXXXXXXXX.com out_uri=sip:2810318308@XXXXXXXXXX.com via_cnt==2". . # U 2005/08/03 13:55:57.381353 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. Call-ID: b521746f-5119658a(a)192.168.1.2. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. CSeq: 102 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/linux)). Content-Length: 0. . # U 2005/08/03 13:55:57.664069 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="2810300308",realm="XXXXXXXXXX.com",nonce="42f0a3e8f78926d700140a7c490f06dd965ea748",uri="sip:2810318308@XXXXXXXXXX.com",algorithm=MD5,response="ac73134a0de530d527cfbe4fe1903707". Contact: <sip:2810300308@XX.XX.XX.XX:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ [View Less]

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[Users] ovpops for serial forking and its cures
by Francesco Bottà 03 Aug '05

03 Aug '05

Hi all, I'm playing around serial forking with avpops module. My goal is to have one PSTN number mapped to N voip number; so, when the first voip number is busy, OPENSER catch 486 reply and then try with the second AOR and so on. But here raise the problem: the OPENSER forward the reply upstream with a code 407 Proxy Auth Required and the PSTN GW send his ACK and all stop here. Below ther's a snippet of mi openser.cfg: modparam("avpops", "avp_aliases", "serial_fork=i:665") [..skipping..] #… [View More]

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RE: [Serusers] is SER Dependable for prepaid System ? Anyone using itfor prepaid system?
by Sam Lee 03 Aug '05

03 Aug '05

Jim, Your prepaid system concept is a little different from mine. For mine, the SER is just a middleman. I route all my calls to Cisco CallManager which will then route accordingly. So I just get all my accounting info from the Callmanager CDR. I then run ASP scripts to check the users balances and if it ever finds a low-in-credit user, it will go into the mysql db by odbc and insert the user into a grp (acl is preconfigured to block this grp from accessing the PSTN). For your case, probably … [View More]the only thing u can do is to match 2 things, the dest. Country code and the different local charges for each country. Do comment about my set up. Regards, Sam -----Original Message----- From: Sanjim Sanjim [mailto:sanjim007@gmail.com] Sent: Wednesday, August 03, 2005 5:33 PM To: Sam Lee Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] is SER Dependable for prepaid System ? Anyone using itfor prepaid system? Sure Sam, I am currently developing a script which will check the credit balance of the user from mysql and accordingly rewrite the uri. If the balance is low... it will forward to SEMS announcement. Like that when SER receives a BYE, the Script should check the destination and total mins and calculate the amount and reduce it from the user credits. Currently the difficult part is the destination country code. I have stored all the codes in the mysql prefix table. But each country have sub codes for states. For example UK proper is only .02 cents, but mobile is .40 cents We have to differentiate the code properly. If you have any other suggestions kindly let me know. Thanks, JiM On 8/3/05, Sam Lee <sam.lee(a)super.net.sg> wrote: > I have been working for quite some time on deploying SER as a prepaid > system. Currently it is not done yet as there's still quite a number > of things to be done. > Probably we can share a few ideas down the road. > > Regards, > Sam > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] > On Behalf Of Sanjim Sanjim > Sent: Tuesday, August 02, 2005 7:02 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] is SER Dependable for prepaid System ? Anyone > using itfor prepaid system? > > is SER Dependable for prepaid System ? Anyone using it for prepaid > system? > > I very much like the SER performance in the test environment. Only > drawback(maybe advantage if we provide unlimited plan) ) is stateless > operation. > > But I want to know anyone using it in realtime prepaid business? > > > > Thanks, > JiM > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > [View Less]

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[Serusers] Can I know Call ID of UAC from Proxy server?
by Abhijit A.Mahajani 03 Aug '05

03 Aug '05

Hi, Can I know the Call ID of a UAC / UAS who has been registered on th eproxy server? I need to send the MESSAGE to the Particular "UAC" but I do not know the Call ID ? There is no INVITE nor Call Established.But I need to send IM to that particular UAC. As Far As I Know, Call id is mandatory , Hence I need to know the Call id from the prosy server. How should I proceed? Thanks and Regards. Abhijit

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RE: [Users] Radius or SQL
by Jayesh Nambiar 03 Aug '05

03 Aug '05

hi bogdan, Thanx for the help. I am sorry to ask again but how do i use the "trusted_reload" command in the script. I tried different techniques but i could not get the results. please help!! Thanx again

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[Serusers] SER got error during the start due to MySQL connection 2
by ser＠ipcb.net 03 Aug '05

03 Aug '05

Hi all. I have configured ProxySER for my test clients and connected they via SER, but... I can't configure SER - MySQL interoperability. I've read "SIP.edu Cookbook" by ProxySER afterward I made the changes into ser.cfg and db configuration. After SER startup I've got the next log: Aug 2 19:21:21 fwd ser[11846]: Maxfwd module- initializing Aug 2 19:21:21 fwd ser[11849]: new_connection(): Access denied for user 'serro'@'localhost' (using password: YES) Aug 2 19:21:21 fwd ser[11849]: … [View More]db_init(): Could not create a connection Aug 2 19:21:21 fwd ser[11849]: ERROR: auth_db_bind: unable to connect to the database Aug 2 19:21:21 fwd ser[11849]: init_mod_child(): Error while initializing module auth_db Aug 2 19:21:21 fwd ser[11849]: ERROR: open_uac_fifo: init_child failed Aug 2 19:21:21 fwd ser[11849]: starting fifo server failed I've had the next config: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ # debug=7 # debug level (cmd line: -dddddddddd) # fork=yes # log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "//lib/ser/modules/mysql.so" loadmodule "//lib/ser/modules/sl.so" loadmodule "//lib/ser/modules/tm.so" loadmodule "//lib/ser/modules/rr.so" loadmodule "//lib/ser/modules/maxfwd.so" loadmodule "//lib/ser/modules/usrloc.so" loadmodule "//lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//lib/ser/modules/auth.so" loadmodule "//lib/ser/modules/auth_db.so" loadmodule "//lib/ser/modules/acc.so" loadmodule "//lib/ser/modules/exec.so" # loadmodule "//lib/ser/modules/group.so" # loadmodule "//lib/ser/modules/msilo.so" loadmodule "//lib/ser/modules/print.so" loadmodule "//lib/ser/modules/textops.so" loadmodule "//lib/ser/modules/uri.so" # loadmodule "//lib/ser/modules/uri_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://ser:mypass@fwd.serout.com/myserdb") modparam("auth", "secret", "myphrase") # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- skipped... Where I can dig for solving this trouble? Thank you in advance. Alexander Naumov. [View Less]

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[Serusers] Ask Billing Server for SER
by Asep Saifudin 03 Aug '05

03 Aug '05

Hi All, Does anyone knows Billing Server for SIP based on SER ? whether for prepaid or postpaid system. I would be glad if there's Opensource SER Billing system but Licensed Billing also ok for me. Thanks & Regards, Asep

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RE: [Serusers] lookup 2 uri
by Samuel Osorio Calvo 03 Aug '05

03 Aug '05

Just one remark: only one of the two URI will be translated to the contact. I'll try to explain... imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup, lookup("aliases"); SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch. Req-URI: 1002(a)test.com branch: 1003(a)test.com Doing the second lookup, lookup("location"); will put the contact of 1002(a)test.com as Req-URI (if more than one … [View More]contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched. Req-URI: 1002(a)10.1.1.102 branch: 1003(a)test.com This happens because lookup() only takes as input the req-URI. The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening. User 1002 will get the message without any problem. NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation. As SER is now coded, above can not be changed (as far as I know).... Hope it helps, Samuel. Unclassified. >>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>> You would need two steps here, looup("aliases"), then lookup("location"). Zeus > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z > Sent: Wednesday, 3 August 2005 3:33 AM > To: Atle Samuelsen > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] lookup 2 uri > > > Hi Atle, > > Alias doesn't replace uri with its real location. I'd like to use > lookup("location") to find out its real location. For > example, if 102(a)test.com is at 102(a)10.1.1.102 and > 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all > uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser > transaction logic. Would it be possible? > > Thanks, > Richard > > > On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote: > > > > Hi, > > > > you could do something like : > > > > serctl alias add 101 sip:102@test.com > > serctl alias add 101 sip:103@test.com > > > > This way the invite would be forked to both 102 and 103, > > > > just remember to have both lookup("aliases"); and > save("aliases"); in > > your config at the right spot :-) > > > > -Atle > > > > > > > > * Richard Z <rzheng(a)gmail.com> [050802 04:34]: > > > Hi, > > > > > > I am trying to implement a feature to ring two ip phones > > > simultaneously. For example, when I call 101(a)test.com, it > rings both > > > 101(a)test.com and 102(a)test.com. In this case, the ruri is > > > 101(a)test.com. I can use append_branch 102(a)test.com, but > is there a > > > way to lookup the locations of both 101 and 102? > > > > > > Thanks, > > > Richard > > > > > > _______________________________________________ > > > 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 > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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[Users] Looking for openser's gentoo ebuild
by Cheng Zhang 03 Aug '05

03 Aug '05

Hi, there In INSTALL doc [http://www.openser.org/docs/INSTALL] section 3-C, it seems to me that a gentoo's ebuild is already available in the official portage tree for OpenSER. Unfortunately I couldn't find it in my daily-rsync'ed Gentoo box, nor I can find it in http://packages.gentoo.org. Searching in both users and devel mailing lists and Google search couldn't find anything useful either. Am I missing something obvious here? Could someone give me a hint please? Thanks. -- Cheng

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[Serusers] How to rewrite source IP -- SIP packet forwarding.
by Ray Van Dolson 03 Aug '05

03 Aug '05

I'm using nathelper with SER. Here's a quickie diagram of my setup: [SIP ATA] <-------------> [SER] <------------------> [Asterisk] 10.0.201.5 eth1 - 10.0.201.7 eth0 - <public_ip1> eth0 - <public_ip0> I'm using the basic nathelper example configuration, except for one modification. I removed the call to t_relay and replaced it with: t_relay_to("<public_ip1>", "5060"); Basically I want a SIP packet to come in to … [View More]

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[Serusers] How to catch re-invite and signal rtpproxy ??
by jimmy 03 Aug '05

03 Aug '05

Hi all: I am using ser 0.8.14 with rtpproxy and kphone 0.4.1.1, When kphone holds the call, then unholds later, sometimes kphone will change the rtp port in SDP differ from previously used. Therefore, the rtp packet can't be forward correctly via rtpproxy if ser doesn't signal rtpproxy to change the forwarding port mapping I have asked before about this problem, and Zeus Ng have give me some information that the key factor is to catch the re-invite, but i don't know how … [View More]

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[Serusers] Re: Problem on pthread and dtmf
by Nicky 03 Aug '05

03 Aug '05

Please help..... ----- Original Message ----- From: Nicky To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org Sent: Tuesday, July 05, 2005 6:56 PM Subject: Problem on pthread and dtmf Dear all, Anyone encountered the problem below,please help. Thank you. ----- Original Message ----- From: Nicky To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org Sent: Thursday, June 16, 2005 3:51 PM Subject: Problem on pthread and dtmf Dear all, 1.… [View More]

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RE: [Serusers] Start and fnish call
by Sam Lee 03 Aug '05

03 Aug '05

Method==Bye ? Try a ngrep during a call transaction to see what are the header sent. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Anderson Alves de Albuquerque Sent: Wednesday, August 03, 2005 12:20 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Start and fnish call With SER how can I know when the client SIP finish a call? I need that SER can start a script when the client SIP finish a call. … [View More]

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RE: [Serusers] serctl add user error
by Sam Lee 03 Aug '05

03 Aug '05

Try adding an alias in it. Eg. Alias=192.168.2.2 -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of han yibing Sent: Wednesday, August 03, 2005 10:08 AM To: serusers(a)lists.iptel.org Subject: [Serusers] serctl add user error hi serctl add 8800 8800 8800(a)192.168.2.2 using above command to add user ,but get following error ERROR:400;check if you use aliases in SER,in fact ,i don't use alisses in ser.cfg file. please help … [View More]

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[Serusers] serctl add user error
by han yibing 03 Aug '05

03 Aug '05

hi serctl add 8800 8800 8800(a)192.168.2.2 using above command to add user ,but get following error ERROR:400;check if you use aliases in SER,in fact ,i don't use alisses in ser.cfg file. please help me. thanks ___________________________________________________________ 雅虎邮箱超强增值服务－2G超大空间、pop3收信、无限量邮件提醒 http://cn.mail.yahoo.com/mail_alert/promo1.html

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[Serusers] rtpproxy problem.
by Ray Van Dolson 03 Aug '05

03 Aug '05

I'm running into a problem with SER & rtpproxy trying to get a NAT scenario to work correctly. I'm still learning about SER, so bear with me... :) Here is my network layout in a nutshell: SIP ATA - 10.0.201.196 (private only, no NAT to internet) SER - eth0: 208.208.53.5/24 - eth1: 10.0.201.5/24 Asterisk - eth0: 65.234.34.5/24 *Everything* is configured to proxy through SER to Asterisk currently. My configuration file is here: http://webdev.digitalpath.net/~rayvd/voice/ser.cfg … [View More]

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[Serusers] failure_route problem
by davor jovanovic 03 Aug '05

03 Aug '05

Hi all, I have a problem with sending a call to PSTN gateway when user is offline. I always get "404 Not found" what is normal because user is not in location database, but I don't understand why failure_route[1] doesn't work. Below is my ser.cfg so if someone could help. I'm using ser 0.9.3. Any help would be appreciated. Thanks __________________________________ route{ # initial sanity checks -- messages with # max_forwards==0, or excessively … [View More]

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[Serusers] lookup 2 uri
by Richard Z 02 Aug '05

02 Aug '05

Hi, I am trying to implement a feature to ring two ip phones simultaneously. For example, when I call 101(a)test.com, it rings both 101(a)test.com and 102(a)test.com. In this case, the ruri is 101(a)test.com. I can use append_branch 102(a)test.com, but is there a way to lookup the locations of both 101 and 102? Thanks, Richard

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[Users] avp questions
by Klaus Darilion 02 Aug '05

02 Aug '05

Hi! I'm playing with AVPs stored in a database and wonder about the following example found in the doc: avp_db_load("$from","i:678"); This loads all AVPs with attribute==678, regardless of the from URI. Is there a usecase for this? If not, why not use the specifier /uri as default if no specifier is selected. Also, in avp_db_load the source may be an avp-alias. Thus, how do you differ between the From: header URI "$from" and the avp-alias "$from"? regards, Klaus PS: Maybe I'm a little … [View More]

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[Users] about the next release
by Daniel-Constantin Mierla 02 Aug '05

02 Aug '05

The features we proposed (and many other) for next release are almost done. As you can see in the roadmap (http://openser.org/roadmap.php), mandatory items were achieved, a few optional are still to be worked on and other that would be good to have in next release popped up on the mailing lists. The major item which is still pending is the authentication via uac module. Another thing that should be there is the clean up of serctl + support for the other database types. Other topics … [View More]

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[Serusers] Start and fnish call
by Anderson Alves de Albuquerque 02 Aug '05

02 Aug '05

With SER how can I know when the client SIP finish a call? I need that SER can start a script when the client SIP finish a call.

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[Serusers] URI rewrite utils
by Pavol Segeč 02 Aug '05

02 Aug '05

Hi, Ser has a few cmd which allows rewrite request uri or some part of it. Does SER have some cmd which allow rewrite caller uri or some part of it? For example, my uri is 1767(a)mydomain.com but when I'm calling outside of my VoIP world to PSTN (for example through Cisco call manager) I would like to identify myself by E.164 address format, i.e. +4214151341767(a)mydomain.com. It should be helpful for me change my URI. Thanks, palo

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[Serusers] MARK RODDY/LAKE is out of the office.
by mark.roddy＠lakecommunications.com 02 Aug '05

02 Aug '05

I will be out of the office starting 08/02/2005 and will not return until 08/18/2005. I will not be checking my mail. For Daidalos related issues please contact Stephen Butler. *********************************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify postmaster(a)… [View More]

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[Serusers] is SER Dependable for prepaid System ? Anyone using it for prepaid system?
by Sanjim Sanjim 02 Aug '05

02 Aug '05

is SER Dependable for prepaid System ? Anyone using it for prepaid system? I very much like the SER performance in the test environment. Only drawback(maybe advantage if we provide unlimited plan) ) is stateless operation. But I want to know anyone using it in realtime prepaid business? Thanks, JiM

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[Users] via header not added when transport=TCP?
by Erich Nahum 02 Aug '05

02 Aug '05

Howdy, I'm playing with OPENSER 0.9.5 with sipp as the workload generator on a bunch of linux boxes. When the transport is UDP, openser adds a via header like it is supposed to as a proxy. But when I use TCP no via header is added. This is a UAC->proxy->UAS scenario. Here's the header as seen at the UAS: | UDP message received [633] bytes : | | INVITE sip:service@10.0.1.41:5060 SIP/2.0 | Record-Route: <sip:10.0.1.42;ftag=1;lr=on> | Via: SIP/2.0/UDP 10.0.1.42;branch=… [View More]

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[Users] lcr from_gw in reply_route
by Klaus Darilion 02 Aug '05

02 Aug '05

Hi! Currently it is not possible to use the from_gw function in a reply_route. I'm no code expert, but to me it looks like it would be safe also to use it in a reply_route. Can please someone review this? regards, klaus

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[Serusers] Please help me with NAT...
by Yiannis Martis 02 Aug '05

02 Aug '05

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[Users] current calls
by Ozan Blotter 02 Aug '05

02 Aug '05

dear list, i'm still working on the billing system... there'r a few things left like active calls and active registers... from openser database, in "location" table i can read current registered users, and it's easy to make a PHP from there to show "active registers" from a web page. but in addition to it, i wan to see active calls, is there a specific way to do it from ser.cfg or database ? debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX port=5060 children=4 alias=212.XXX.104.XXX … [View More]dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transactions", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.XXX.104.XXX:5060;nat=yes"); } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { record_route(); setflag(1); } }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # this is our PSTN gateway route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } [View Less]

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[Users] Problems with digest authentication
by aimable 02 Aug '05

02 Aug '05

I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost and to the the IP address of the server, … [View More]

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[Serusers] Xlite not working with SER+SEMS
by yu wang 02 Aug '05

02 Aug '05

Hi, I've been working on conference with Xlite but it is not work. The problem is SEMS starts sending media (default wav) back to Xlite once SEMS receives the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the media port after sending the INVITE thus it replys with ICMP port unreachable. But I don't see anyone complain this on this list. Is something wrong with my configuration? Please please help me.... Thanks, Pat

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Re: [Serusers] URI rewrite utils
by Klaus Darilion 02 Aug '05

02 Aug '05

Pavol Segeč wrote: > Hi, > > But as I see in rfc3261 such header as Remote-Party-ID is not there, or I'm > wrong? What this headre means? In which rfc it is described? I suppose that It never reached the RFC state, it was only a draft, but is supported by Cisco, Asterisk, ser, Inalp, ... http://www.ietf.org/proceedings/02mar/I-D/draft-ietf-sip-privacy-04.txt > for wished functionality should we need rewrite From and Contact headers Do not rewrite From: as it is forbidden (… [View More]

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[Serusers] Radius Authentication
by Ryan Pagquil 02 Aug '05

02 Aug '05

Guys, I'm testing SER to authenticate with radius. But when I start authenticating I see this on the radius log: rad_recv: Access-Request packet from host 127.0.0.1:1131, id=222, length=262 User-Name = "rpagquil@server4all" Digest-Attributes = "\n\nrpagquil" Digest-Attributes = "\001\014server4all" Digest-Attributes = "\002*42ef48a123c4e75c2d998852eaa5d4fb14bc9917" Digest-Attributes = "\004\020sip:server4all" Digest-Attributes = "\003\… [View More]nREGISTER" Digest-Response = "1df283adcf333605c0007d8a86a2e332" Service-Type = Sip-Session Sip-URI-User = "rpagquil" Cisco-AVPair = "call-id=CE373D63037311DABFB500E04CAB4AB4@server4all" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 modcall: entering group authorize for request 150 modcall[authorize]: module "preprocess" returns ok for request 150 modcall[authorize]: module "chap" returns noop for request 150 modcall[authorize]: module "eap" returns noop for request 150 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "rpagquil" Digest-Realm = "server4all" Digest-Nonce = "42ef48a123c4e75c2d998852eaa5d4fb14bc9917" Digest-URI = "sip:server4all" Digest-Method = "REGISTER" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 150 rlm_realm: Looking up realm "server4all" for User-Name = "rpagquil@server4all" rlm_realm: No such realm "server4all" modcall[authorize]: module "suffix" returns noop for request 150 users: Matched rpagquil@server4all at 138 modcall[authorize]: module "files" returns ok for request 150 modcall[authorize]: module "mschap" returns noop for request 150 modcall: group authorize returns ok for request 150 rad_check_password: Found Auth-Type Digest auth: type "Digest" modcall: entering group Auth-Type for request 150 A1 = rpagquil:server4all:test A2 = REGISTER:sip:server4all KD = 94b43b69398cb3ca2eef355e9875d36f:42ef48a123c4e75c2d998852eaa5d4fb14bc9917:33f62c1688cd77a13c84b07b3877bb1c *rlm_digest: FAILED authentication modcall[authenticate]: module "digest" returns reject for request 150 modcall: group Auth-Type returns reject for request 150 auth: Failed to validate the user. Login incorrect: [rpagquil@server4all/<no User-Password attribute>] (from client me2 port 5060)* It says that there is no User-Password attribute contained in my authentication request. What could be the problem? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com [View Less]

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[Users] How to rewrite source IP -- SIP packet forwarding.
by Ray Van Dolson 02 Aug '05

02 Aug '05

I'm using nathelper with OpenSER 0.10.x. Here's a quickie diagram of my setup: [SIP ATA] <-------------> [OpenSER] <--------------> [Asterisk] 10.0.201.5 eth1 - 10.0.201.7 eth0 - <public_ip1> eth0 - <public_ip0> I'm using the basic nathelper example configuration, except for one modification. I removed the call to t_relay and replaced it with: t_relay_to_udp("<public_ip1>", "5060"); Basically I want a SIP packet … [View More]

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[Serusers] SER <==> SEMS fifo permisions problem
by Pavol Segeč 02 Aug '05

02 Aug '05

Hi All ! I'm implementing SER, SEMS server, I've configured fifo as communication method between them (folowing install manual). In ser.cfg I've enabled conference part in INVITE message route handling as follows : if (uri=~"sip:100.*@") { if (!t_newtran()){ sl_send_reply("500","Could not create transaction"); break; }; t_reply("100","Trying - just wait a minute !"); # assumes that Sems configuration parameter 'socket_name=' … [View More]

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[Serusers] can not open dictionary from freeradiusd!
by zhu lizhong 02 Aug '05

02 Aug '05

hello, guys: I installed freeradiusd and SER. now, i want to connect the two together. the problme is that when i include dictionary.ser in raddb/dictionary, and run radiusd -X, then the messages came out as follow: ------------------------------------- read_config_files: reading dictionary Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary[14]: Couldn't open dictionary " /usr/local/share/freeradius/dictionary": No such file or directory Errors reading radiusd.conf ---------… [View More]

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[Serusers] Problem with digest authentication
by aimable 02 Aug '05

02 Aug '05

Hi serusers, I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost, the IP address, with no luck.… [View More]

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[Serusers] No Audio
by Ashutosh Kumar 02 Aug '05

02 Aug '05

Hi, I am using the following ser.cfg with ser and mediaproxy, but I am getting the audio only in one direction. This does not seem a problem of NAT, as the nathelper is correclt rewriting the sdp messages, and the aor shows all clients bound with public IP. Earlier t_on_reply call in route[1] was causing an error sl_reply_error used: I'm terribly sorry, server error occured (1/SL) when one calls from one phone to another. Earlier I was getting proper bi-directional audio when I was using … [View More]STUN instead of mediaproxy. Is there something wrong with my ser.cfg #-------------------------------------------------- # # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode #debug=7 #fork=no #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.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/domain.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/msilo.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("uri", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 3) modparam("acc", "db_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) modparam("acc", "log_level", 2) #modparam("acc", "radius_missed_flag", 3) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") # -- rr params -- modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registrar@mydomainxyz123.com") modparam("msilo", "db_url", "sql://ser:heslo@mydomainxyz123.com/ser") modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("uri", "subscriber_table", "subscriber") modparam("uri", "uri_table", "uri") modparam("uri", "uri_user_column", "username") modparam("uri", "uri_domain_column", "domain") modparam("uri", "uri_uriuser_column", "uri_user") modparam("uri", "subscriber_user_column", "username") modparam("uri", "subscriber_domain_column", "domain") # !! Nathelper 1 #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 # !! NAthelper 1 ends # ------------------------- request routing logic ------------------- # main routing logic 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 too big");break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- # we record-route all messages -- to make sure that subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities# use different transport protocol if (method=="INVITE"){ # && client_nat_test("3")) { record_route_preset("mydomainxyz123.com:5060;nat=yes"); route(3); } else if (method!="REGISTER") {record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- # if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- lookup("aliases"); if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) {setflag(6); #use_media_proxy(); }; ; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # if the request is for other domain use UsrLoc (in case, it does not work, use the following command # with proper names and addresses in it) if (!method=="BYE") { force_rport(); fix_nated_contact(); }; if (uri!=myself) {route(1);break;}; if (uri==myself) { if (method=="CANCEL") {route(3); break;}; if (method=="REGISTER") {route(2); break;}; if (method=="MESSAGE") {route(4); break;}; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ){ route(3);break; }; sl_send_reply("404", "Not Foundbusip1"); break; } else {setflag(1);t_relay();break;}; # if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- # t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { #end_media_session(); ; }; # sl_send_reply("404", "balleballeUser Not Found"); # sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); #if (!search("^Contact:\ +\*") && client_nat_test("7")) { # setflag(6); #fix_nated_register(); # force_rport(); #}; #if (!radius_www_authorize("")) { #if (!www_authorize("","subscriber")) { # www_challenge("","0"); # break; #}; #if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # break; #}; consume_credentials(); save("location"); m_dump(); acc_rad_request("Some comment"); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- #if (client_nat_test("3")) { # setflag(7); # force_rport(); # fix_nated_contact(); #}; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); acc_db_request("404 missed call", "missed_calls"); acc_rad_request("404 missed call"); setflag(3); break; }; if (method=="CANCEL") { route(1); break; }; #if (!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # break; #} else if (!check_from()) { # sl_send_reply("403", "Use From=ID"); # break; #}; consume_credentials(); if (isflagset(6) || isflagset(7)) { # use_media_proxy(); ; }; route(1); } route[4] { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; ; }; if (client_nat_test("1")) { fix_nated_contact(); }; ; } Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com [View Less]

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[Serusers] Non eixsting File in mediaproxy modparam
by Ashutosh Kumar 02 Aug '05

02 Aug '05

Hi, In my ser.cfg being used with mediaproxy for NAT traversal, there are two lines reuiqred by this package. modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") But these two files do not exist (*anywhere*) on my machine. I suppose that they are a part of the mediaproxy package.Am I missing them or is the package outdated. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin … [View More]

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[Serusers] Xlite not working with SER+SEMS
by yu wang 02 Aug '05

02 Aug '05

Hi, I've been working on conference with Xlite but it is not work. The problem is SEMS starts sending media (default wav) back to Xlite once SEMS receives the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the media port after sending the INVITE thus it replys with ICMP port unreachable. But I don't see anyone complain this on this list. Is something wrong with my configuration? Please please help me.... Thanks, Pat

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[Serusers] serctll add error
by chris 02 Aug '05

02 Aug '05

hello i tried running serctl add 1001 1001 chris_cleofe2yahoo.com but i got this error: awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed pls advise. thnks.

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[Serusers] ser and database
by Sebastian Kühner 01 Aug '05

01 Aug '05

Hi! Has anyone of you used a database to make routing? For example: If the caller marks 001 -> Rewritehost("xxx.xxx.xxx.111"); If the caller marks 0049 -> Rewritehost("xxx.xxx.xxx.222"); The two host names are in my postgres database... Once I used a command like this to stop calls of users that don't have credit: if (!exec_msg(' POS1=`expr index "$SIP_HF_FROM" sip: + 4`; LENGTH=`expr index "$SIP_HF_FROM" @ - $POS1`; … [View More]

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[Serusers] gw-pstn.3.07.cfg with Call-Transfer support
by Bastian Schern 01 Aug '05

01 Aug '05

Hello, I tried out the SER config "gw-pstn.3.07.cfg" from www.onsip.org. Nearly everything works fine but "Call-Transfer" is not working. Is it possible to integrate "Call-Transfer" support into the "gw-pstn.3.07.cfg"? Does anybody integrates this already? Regards Bastian debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/… [View More]ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(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); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.0.2.245"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } [View Less]

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[Serusers] codec question
by Innocent Evil 01 Aug '05

01 Aug '05

Scenario: IP Phone --> SIP Server --> Asterisk --> (Voice Mailbox) (G723.1, G729) -> (GSM) -> (GSM, G729) GSM coded file Is ipphone with codec g723.1 and g729 cabable to retrived voice mail from the above picture? If not, please explain a little more. Thanks a lot.

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[Serusers] UAC module status of CVS HEAD version
by Innocent Evil 01 Aug '05

01 Aug '05

Hello, Anybody know about the status of UAC module of CVS HEAD version. Does it work? Thanks,

4 4

[Serusers] NAT-flag and multiple branches
by Klaus Darilion 01 Aug '05

01 Aug '05

Hi! I wonder how the NAT flag will be set during lookup("location") in case of multiple branches and some of the contacts are behind NAT and some not. regards, klaus

2 1

AW: [Serusers] dbtext: table not loaded. example desired
by Müller Thorsten 01 Aug '05

01 Aug '05

Hi all, I'm still at dbtext: I have a line in ser.cfg which is now: modparam("auth_db", "db_url","dbtext:///usr/local/etc/ser/dbtext") where I have a file produced with the script from cesc called "subscriber". Given by DEBUG-mode from dbt_load_file the module "dbtext" searches this file in a directory called "/usr/local/etc/ser" and NOT in the dir given by the "modparam"-line Can somebody give me a hint, why this occurs? Another problem I had with the strncpy/strncat of database and … [View More]table-name. I had to change a line in db_file.c at line 82 from strncpy(path+dbn->len+1,tbn->s,tbn->len) to strncat(path,tbn->s,tbn->len) Is this a bug or is something wrong with my ser.cfg? Thanks Thorsten ser-version: 0.8.14 mt> -----Ursprüngliche Nachricht----- mt> Von: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] mt> Gesendet: Freitag, 29. Juli 2005 10:55 mt> An: Müller Thorsten mt> Cc: serusers(a)lists.iptel.org mt> Betreff: Re: [Serusers] dbtext: table not loaded. example desired mt> mt> mt> Hello, mt> mt> On 07/29/05 11:16, Müller Thorsten wrote: mt> mt> > Hi all, mt> > mt> > I try to set up ser with dbtext as authentication basis. mt> I run in some mt> > problems because the database is not found: mt> > DBT:db_query: table not loaded. mt> > mt> > After some googling a found that Cesc mt> [cesc.santa(a)gmail.com] had also mt> > done some work with dbtext and mailed that to "serusers" mt> or "serdev" mt> > mailinglists mt> > mt> > In a mail from May and June 05 he told that he has mt> written a "serctl" mt> > for dbtext and a script to produce the necessary files. mt> > mt> > Has somebody got these things and can mail me these mt> files! I could not mt> > found them in the mailing history. mt> > mt> > Has somebody also dealed with dbtext and can give me some example mt> > files for ser.cfg and dbtext files for mt> "authentication/subscriber" and mt> > "location". mt> > mt> rhe readme file of dbtext module has some examples of table mt> structures mt> as well as a simple config file. mt> mt> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/mod mt> ules/dbtext/README?rev=HEAD&content-type=text/vnd.viewcvs-markup mt> mt> Daniel mt> mt> > Thanks in advance mt> > mt> > Thorsten Mueller mt> > mt> > ser 0.8.14 cross-compiled for arm mt> > My ser.cfg: mt> > #################################################### mt> > ... mt> > # Added by mt for authentication with dbtext mt> > loadmodule "/usr/local/lib/ser/modules/auth.so" mt> > loadmodule "/usr/local/lib/ser/modules/auth_db.so" mt> > loadmodule "/usr/local/lib/ser/modules/dbtext.so" mt> > mt> > modparam("auth_db", "db_url","dbtext:///var/dbtext/ser") mt> > mt> > # -- auth params -- mt> > modparam("auth_db", "calculate_ha1", 1) mt> > modparam("auth_db", "password_column", "password") mt> > modparam("auth_db", "user_column", "username") mt> > modparam("auth_db", "domain_column", "domain") mt> > ... mt> > mt> > if (uri==myself) { mt> > mt> > if (method=="REGISTER") { mt> > mt> > if (!www_authorize("domain.com", "subscriber")) { mt> > www_challenge("domain.com", "0"); mt> > break; mt> > }; mt> > save("location"); mt> > break; mt> > } mt> > ... mt> > ##################################################### mt> > mt> > I have a file in /var/dbtext/ser which is called "subscriber": mt> > ##################################################### mt> > username(str) password(str) ha1(str) domain(str) ha1b(str) mt> > suser:supasswd:xxx:domain.com:xxx mt> > ##################################################### mt> > mt> >----------------------------------------------------------- mt> ------------- mt> > mt> >_______________________________________________ mt> >Serusers mailing list mt> >serusers(a)lists.iptel.org mt> >http://lists.iptel.org/mailman/listinfo/serusers mt> > mt> > mt> [View Less]

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[Serusers] Problem with digest authentication
by aimable 01 Aug '05

01 Aug '05

Hi serusers, I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost, the IP address, with no luck.… [View More]

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[Serusers] Adding Asterisk Support for Voice Messages
by ntandoh＠termilink.net 01 Aug '05

01 Aug '05

Does anyone have info on adding Asterisk Support for Voice Messages, that page is missing from the SIP - Getting Started - Issue 04a.doc www.ONsip.org ?

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[Serusers] sip_method
by Szabolcs Szasz 01 Aug '05

01 Aug '05

Hi my friends! I have been tested my SER config for a few weeks (ser-0.8.14). It's OK, but in the ser's MySQL database in the acc table appears a few row with sip_method=PRACK and sip_method=INFO (there are > 1100 rows with INVITE,ACK and BYE). What is this meaning? I cannot regenerate this row. can anybody help me? Thanks Szabolcs

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[Serusers] Existing Radius Authentication
by Ryan Pagquil 01 Aug '05

01 Aug '05

Hi, Can I use my existing radius server as my login authentication for ser? The existing radius uses the system to read the user accounts, but explained on the radius howto i must create the user accounts on users file of the freeradius. Please help. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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RE: [Serusers] 1 way voice issue
by Sam Lee 01 Aug '05

01 Aug '05

Could the 183 session progress have anything to do with the one way voice ? ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, August 01, 2005 4:54 PM To: Sam Lee; serusers(a)lists.iptel.org Subject: Re: [Serusers] 1 way voice issue Very hard to say without a complete ngrep trace. If both parties are on public IPs, there is probably something with the signalling, not the SDP payload. The 183 Session Progress is a so-called provisional reponse. … [View More]

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[Users] Using permissions
by jayesh nambiar 01 Aug '05

01 Aug '05

hi there, Can you please tell me if I can use the permissions module to trust the requests coming towards my SIP server using source IP. But my problem is I want to use flat files to store the source IPs I trust. Is it possible And if the IPs are trusted I need to forward them without any authentication. I need to store the trusted IPs in flat files since I dont need the overhead of querying DB. Please help me out. Thanx --------------------------------- Start your day with Yahoo! - make … [View More]

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[Serusers] Any known problems with running SER behind NAT?
by Alan 01 Aug '05

01 Aug '05

Hi List, I wonder if you could share your experience with me of running SER behind NAT. Are there any known problems with doing it this way? - Server side NAT and have SER fix NAT'ed clients and (Media)Proxying them where needed. My specific setup is as follows:- SER(0.9.13) with a static private IP Firewall performing static on-to-one NAT Users will be on public IP's/(behind NAT) Does this cause any problems with the SIP/SDP packets? Also it is the intention to run Asterisk in the … [View More]

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[Serusers] 1 way voice issue
by Sam Lee 01 Aug '05

01 Aug '05

Hi, I've recently encountered some problem with my SIP service whereby i call out to a specific number and i encounter a one way voice. If i'm the initiator, i cannot hear the other party but he can hear me. At first i thought it was a return route issue (as i'm going thru NAT) , so i switch my SIP to a public IP but i still face the same problem. Its really only that specific PSTN number that i have dialed facing this problem. The only difference that i can think of is that PSTN number is on … [View More]

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[Serusers] STUN for NAT traversal...
by Roger Lewau 01 Aug '05

01 Aug '05

I'm trying to use STUN for NAT traversal for all clients not using 1 to 1 NATing, and mediaproxy for the rare few who does.But it seems that all traffic is proxied anyway, dont really understand why. I do call the routeblock for mediaproxy before relaying the call, but since STUN is used it should not detect that NAT is in place. Or am I missing some logic? Any Idea on how to accomplish this? Kind regards Roger - Getting one step closer to a really god service everyday ;)

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[Serusers] Mediaproxy Load Balacing
by smy＠addm.com 01 Aug '05

01 Aug '05

Hi all, I've tried playing with Mediaproxy for sometime now, and thou with some clarification from Greger, I'm still having some problems with getting it up and running. I'm trying to have 1 SER with multiple Mediaproxy, and here's my configuration. I'm desperate since I've experienced no improvement with the system after over a week playing with it! :< Any help is really appreciated. --------- ser.cfg --------- loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/… [View More]

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[Serusers] NAT Problem (Mediaproxy)... Pls help
by Yan Yu Lim 01 Aug '05

01 Aug '05

Hi Guys, I have a NAT problem, whereby I cannot get the RTP packets from a NATed user. Hence, I am trying to implement the Mediaproxy solution from ONSIP. 1. Mediaproxy and Ser are installed on the same PC, say 202.123.45.85. I've set SER to listen at 202.123.45.85, port 5060. How should I configure the ini file for Mediaproxy? Can someone please provide me with a simple example of how the mediaproxy ini file should be configured? 2. Assuming that I set mediaproxy to listen at 202.123.45.85, … [View More]

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[Serusers] Mediaproxy -- pls help!
by yy lim 01 Aug '05

01 Aug '05

Hi guys, I'm trying to get my mediaproxy working without much success. Both mediaproxy and ser are installed on the same machine, i.e. 201.23.13.145 for example. SER is listening on port 5060, and mediaproxy is set up as below. Please kindly advise. Rgrds, Zan ----- Mediaproxy.ini ------ [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = ser listen = 201.23.13.145:5062 allow = any proxyIP = 201.23.13.145 portRange = 5062:5064 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 … [View More]forceClose = 0 accounting = off ------- ser.cfg ------- debug=3 fork=yes listen=201.23.13.145 log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:XXXXX@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # -- mediaproxy NAT modules --- loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- auth_radius params -- modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:XXXXXX@localhost/ser") modparam("acc", "db_url", "mysql://ser:XXXXXX@localhost/ser") # -- usrloc params -- # store user location in memory, not using database modparam("usrloc", "db_mode", 0) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", 1) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # mysql accounting # -- acc params -- modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) # -- 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",15) modparam("tm","fr_timer",10) modparam("tm","fr_inv_timer_avp", "inv_timeout") #modparam("permissions", "db_mode", 1) #modparam("permissions", "trusted_table", "trusted") # -- Mediaproxy params -- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # main routing logic route{ # ----------------------------- # Sanity check # ----------------------------- # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # ----------------------------- # Record Route Section # ----------------------------- # perform NAT check if (method=="INVITE" && client_nat_test("3")) { record_route_preset("201.23.13.145:5060;nat=yes"); log(1,"Invite + NAT test- Record Route\n"); } # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol else if(method!="REGISTER") { record_route(); }; # ------------------------------- # Call Tear Down Section # ------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); log(1,"Tear Down Section\n"); }; # ------------------------- # loose-route processing # ------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; log(1,"Loose route processing\n"); }; route(1); break; }; setflag(1); # ----------------------------- # 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); t_on_failure("1"); break; } else if(method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; route(1); } # ------------------------------- # Route Processing Section # ------------------------------- # ------------------------------ # Default Message Handler # ------------------------------ route[1] { t_on_reply("1"); if(!t_relay()) { if(method=="INVITE" || method=="ACK") { end_media_session(); log(1,"Route 1 default message handler\n"); }; sl_reply_error(); }; } # REGISTER Message Handler # ---------------------------------------------------------------- route[2] { sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); log(1,"Nated register\n"); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } # ----------------------------------------------------------------- # CANCEL & INVITE Message Handler # ----------------------------------------------------------------- route[3] { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if(method=="INVITE" && !(src_ip==10.10.10.4)) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { # attempt handoff to PSTN if (uri=~"^sip:8[0-9]*@") { log(1, "Forwarding to PSTN\n"); route(4); break; }; # retrieve voicemail if (uri=~"^sip:2[0-9]*@") { log(1, "Retrieving voicemail\n"); route(7); break; }; sl_send_reply("404", "User Not Found!"); break; }; if(method=="CANCEL") { route(1); break; }; route(5); route(1); } # ----------------------------- # PSTN Handler # ----------------------------- route[4] { rewritehostport("10.10.10.4:5060"); avp_write("i:45", "inv_timeout"); # ------------------------------- # RTP Proxy Enabler # ------------------------------- route[5] { if(isflagset(6)||isflagset(7)) { use_media_proxy(); log(1,"Using rtp proxy\n"); }; } # ------------------------------- # ACK # ------------------------------- route[6] { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found!"); break; }; route(5); route(1); } route(5); route(1); } # ------------------------------ # Asterisk voicemail # ------------------------------ route[7] { #redirect now! rewritehostport("201.23.13.145:5061"); append_branch(); t_relay_to_udp("201.23.13.145","5061"); } failure_route[1] { if(!(method=="BYE") && !(method=="CANCEL")) { log(1,"Activating voicemail!!\n"); }; #revert_uri(); #redirect now! rewritehostport("201.23.13.145:5061"); append_branch(); t_relay_to_udp("201.23.13.145","5061"); } onreply_route[1] { log(1,"On reply route/n"); if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com [View Less]

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Re: [Serusers] serctll add error
by chris 01 Aug '05

01 Aug '05

hi kofi, were you able to run and test my config file? any developments? :) much thanks. chris ----- Original Message ----- From: chris To: Kofi Obiri-Yeboah ; serusers(a)lists.iptel.org Sent: Wednesday, July 27, 2005 11:46 AM Subject: Re: [Serusers] serctll add error hi kofi, i did commented out the listen=IP line. but i gives me the same error. ERROR: bad config file (1 errors) by the way, when i tried the hello_world.cfg from onsip i did not encountered any … [View More]problem and ser was able to run. thnks kofi, i will wait for you reply next week. chris. ----- Original Message ----- From: Kofi Obiri-Yeboah To: chris ; serusers(a)lists.iptel.org Sent: Wednesday, July 27, 2005 11:19 AM Subject: RE: [Serusers] serctll add error Chris, I will run your config file over the weekend and will get back to you. Clearly that fact that you are receiving a bad config file error is an indication that there is a problem. Could you please comment out the listen=IP address line and rerun ?????? KOFI -----Original Message----- From: chris [mailto:chris@fivestartel.com] Sent: Tuesday, July 26, 2005 10:19 AM To: Kofi Obiri-Yeboah; serusers(a)lists.iptel.org Subject: Re: [Serusers] serctll add error hi kofi, attached is my ser.cfg i got this sample script from onsip.org getting started doc. this is their auth-mysql.3.05.cfg sample script, i just change the IP address on the listen variable. running using this cfg file leads me to an error. ERROR: bad config file (1 errors) maybe u can help me fix this. i have no idea how i can give you more details on the error cause that is the only line the system gives me. thnks. debug=3 fork=yes log_stderror=no listen=203.115.128.41 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } ----- Original Message ----- From: Kofi Obiri-Yeboah To: chris ; serusers(a)lists.iptel.org Sent: Tuesday, July 26, 2005 11:39 PM Subject: RE: [Serusers] serctll add error please send your config file -----Original Message----- From: chris [mailto:chris@fivestartel.com] Sent: Tuesday, July 26, 2005 6:34 AM To: Kofi Obiri-Yeboah; serusers(a)lists.iptel.org Subject: Re: [Serusers] serctll add error hi, i still got the same error bash-2.05# serctl add 1001 1001 chris(a)yahoo.com awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed bash-2.05# pls. advise on how to fix this. thnks. ----- Original Message ----- From: Kofi Obiri-Yeboah To: chris ; serusers(a)lists.iptel.org Sent: Tuesday, July 26, 2005 8:06 PM Subject: RE: [Serusers] serctll add error syntax is <username> <password> <email address> you have a typo in the email address use "@" instead of "2" use shift-2 before yahoo.com HTH -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of chris Sent: Tuesday, July 26, 2005 12:58 AM To: serusers(a)lists.iptel.org Subject: [Serusers] serctll add error hello i tried running serctl add 1001 1001 chris_cleofe2yahoo.com but i got this error: awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed pls advise. thnks. 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Re: [Serusers] Problem : Can SER process the reINVITE messages properly?
by Greger V. Teigre 01 Aug '05

01 Aug '05

BTW, I sketched out a few other solutions, where I believe at least a couple should be valid attempts. Did you try those and what were the results? g-) Ricardo Martinez wrote: > Hello Greger and SER users. > About this issue here are some test that i made. I added the > lookup("location") as you recommend me : > > # > > > > ----------------------------------------------------------------- # > Loose Route Section # > -----------------------------------… [View More]------------------------------ if > (loose_route()) { > > if (has_totag() && (method=="INVITE" || > method=="ACK")) { lookup("location"); > if (isflagset(6) || client_nat_test("3") || > search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > break; > }; > > > Unfortunately, i'm still unable to have voice in both ways, but this > modification seems to handle better the reINVITE message. And this > is why. For the normal approach (i mean the initial onsip_pstn > configuration) the second INVITE was not using the mediaproxym as > Greger pointed. Now, as you can see in the debugs that i'm > attaching, the second invite is using the mediaproxy, and therefore > the "c" parameter in the SDP message is being correctly modified by > SER. So, i don't understand why this time the call is failing. Could > it be that mediaproxy is not capable to handle the reINVITEs?. For > example in the debug from mediaproxy (i'm also attaching this one), i > see a warning message like this : warning: Received packet from a > third party: 200.0.0.7:16432. It seems like mediaproxy is not > accepting "voice packets" from the second IP (the one indicated by > the reINVITE). Why? > Also, when the call is established this is the output from the > "session" command : > > Caller Via Called > Status Duration Codec Type Traffic > ---------------------------------------------------------------------------- > -------------------------------------- > 200.0.0.6:17480 - 200.0.0.5:35094 - 200.0.0.4:46115 inactive > 0'24" G729 Audio 0/16.75k/16.75k > > The call is established but between the first IP (Asterisk box) and > the endpoint public IP. > So, could this be a problem with mediaproxy?. I read the changelog > from the version 1.3.1 and this is what i found : > > Changes from version 1.2.1 to 1.3.0 > ----------------------------------- > > - Only create a new session if a request belongs to the first INVITE > (no longer creates new sessions on re-INVITEs) > > could it be related?. > > I also have a general question about the reINVITE. As you can see > the first INVITE has in his URI the domain name for the number : > 5555848114(a)sipvoiss.desarrollo.redvoiss.net (Frame 1 from the debug). > The reINVITE sent by Asterisk has the URI : 5555848114(a)200.0.0.4. Is > this ok?, despite of this the reINVITE is correctly redirected to the > endpoint. (You can check this in the frame 13 from the debug). > > Thanks again. > > Regards, > > Ricardo Martinez.- > > > >> -----Mensaje original----- >> De: Greger V. Teigre [mailto:greger@teigre.com] >> Enviado el: Miércoles, 20 de Julio de 2005 3:06 >> Para: Ricardo Martinez; serusers(a)lists.iptel.org >> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >> messages properly? >> >> >> Hm. This was a tricky one. I would like to ask a second >> opinion from you >> NAT gurus on my musings below. What is the most appropriate >> response to >> Ricardo's scenario? WARNING to others: This is getting fairly >> technical on >> NAT/SDP... ;-) >> >> The reINVITE from Asterisk has a valid IP address and a valid >> (but new) SDP >> media ip (C=IN IP4 200.0.0.7). nat=yes is correctly not >> present in the >> reINVITE from Asterisk (it is not NATed). Thus >> use_media_proxy is correctly >> not used in the reINVITE, i.e. client_nat_test("3") and test >> on nat=yes will >> not trigger under the has_totag if statement (loose route >> processing): >> >> if (loose_route()) { >> >> if (has_totag() && (method=="INVITE" || method=="ACK")) { >> >> if (client_nat_test("3") || search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> }; >> >> route(1); >> break; >> }; >> >> In the OK from the Linksys: >> U 200.0.0.5:5060 -> 200.0.0.6:5060 >> SIP/2.0 200 OK. >> To: >> <sip:5555848114@sipvoiss.desarrollo.redvoiss.net>;tag=64d7492d >> b018b13ai0. From: "5565625858196" >> <sip:5565625858196@sipvoiss.desarrollo.redvoiss.net>;tag=as5d352c9a. >> Call-ID: >> 0848ef78297116742562204e2c04cbcb(a)sipvoiss.desarrollo.redvoiss.net. >> CSeq: 104 INVITE. >> Via: SIP/2.0/UDP 200.0.0.6:5060;branch=z9hG4bK518d870e. >> Record-Route: <sip:200.0.0.5;ftag=as5d352c9a;lr=on>. >> Contact: 5555848114 <sip:5555848114@200.0.0.4:46081>. >> Server: Linksys/PAP2-2.0.10(LSb). >> Content-Length: 237. >> Content-Type: application/sdp. >> . >> v=0. >> o=- 33068 33068 IN IP4 192.168.15.200. >> s=-. >> c=IN IP4 192.168.15.200. >> t=0 0. >> m=audio 16424 RTP/AVP 18 100 101. >> a=rtpmap:18 G729/8000. >> a=rtpmap:100 NSE/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-15. >> a=ptime:30. >> a=sendrecv. >> >> ...you have 192.168.15.200, a private c= address in SDP, but >> the Contact has >> correctly been fixed (200.0.0.4:46081). So it seems the OK is >> processed >> correctly in onreply even though it does not show up in the >> debug. Flags 6 >> and 7 are not set, so use_media_proxy is correctly not run. >> The problem is >> that flag 6 should have been set (the Linksys is NATed), but >> as the reINVITE >> is not processed using lookup("location") the flag will not >> be picked up >> from usrloc. >> ---------------------------------------- >> >> However, now that Asterisk in the reINVITE is requesting >> another (valid) IP >> as media destination, the question is: what is propor behavior? >> >> The most natural would be to add a lookup("location") after: >> if (has_totag() && (method=="INVITE" || method=="ACK")) >> and then add isflagset(6) to the or test: >> if (isflagset(6) || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> However, I'm not sure the lookup will work, because the R-URI >> is likely to >> be the direct address (5555848114@200.0.0.4:46081) and not >> the registered >> URI (@domain.com). It would be tempting to try to see if we >> somehow could >> make nat=yes set in the early INVITE. We would have to split the >> record_route section in two. One to handle !="REGISTER && >> !="INVITE" with a >> regular record_route() and wait until AFTER >> lookup("location") (maybe in >> route[1], the default handler) to add the following: >> if(method=="INVITE")){ >> if(isflagset(6)) { >> # INSERT YOUR IP ADDRESS HERE >> record_route_preset("192.0.2.13:5060;nat=yes"); >> } else { >> record_route(); >> } >> } >> >> This would make sure that not only dialogs where caller is >> nated, but also >> where callee is nated, will be tagged as nated. This again >> would cause >> nat=yes to be set and rtp proxy forced, further flag 6 is >> set, and thus the >> OK also processed with use_media_proxy(). >> >> BUT: Would this nat=yes cause any problems in other >> scenarios??? I'm not >> sure, but I don't think so. If that is correct, this is >> maybe the right >> solution. >> --------------------------------------- >> OTHER (SIMPLER?) SOLUTIONS: >> ---------- >> Instead you could add ||client_nat_test("1") together with >> isflagset 6 and 7 >> in your reply route: >> if ((isflagset(6) || isflagset(7) || client_nat_test("1")) && >> (status=~"(180)|(183)|2[0-9][0-9]")) >> >> but it will not work because the reINVITE did not get >> use_media_proxy(). >> >> Or, you could instead add fix_nated_sdp("2"); in your onreply route: >> if (client_nat_test("1")) { >> fix_nated_contact(); >> fix_nated_sdp("2"); >> }; >> WARNING! You will have to add else in between this test and >> the previous to >> avoid messing up use_media_proxy(). >> >> This will fix the SDP (c=IN IP4), but it will probably not >> help, because the >> port is still unknown. You could use "3" instead in >> fix_nated_sdp, this >> will add direction=active to the OK. However, another thread >> stated that >> Asterisk does not support direction=active. If it does, it >> will probably >> work and you get a non-proxied stream between Asterisk and >> the Linksys. >> >> Or finally, the third option would be to add the (hopefully) >> fixed IP in the >> loose_route test: >> if (src_ip==200.0.0.6 || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> This would cause the OK automatically to get use_media_proxy. >> >> Ricardo, could you please test these various solutions and >> see if they >> really work or if (where...) I go wrong :-) >> g-) >> >> Ricardo Martinez wrote: >>> Hello Greger. >>> Thanks for your answer on this topic. Now i'm attaching more debug >>> information (the /var/log/messages from mediaproxy, the ngrep output >>> and some xlog statements in the ser.cfg file) beside some >> comments in >>> the file reINVITE_debug_problem.txt. >>> >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>> >>> At least i have ringback tone. Then, when the call is answered i >>> have an OK and a ACK message coming to my SER box, then inmediatly >>> the reINVITE message arrives Asterisk box. >>> >>> >>> For what i can see from the debug the "nat=yes" is never reached >>> because the caller has a "valid ip" and therefore the first "if" in >>> the statament >>> >>> if (method=="INVITE" && client_nat_test("3")) { >>> # INSERT YOUR IP ADDRESS HERE >>> record_route_preset("64.76.148.246:5060;nat=yes"); >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> invite & client_nat_test(3) TRUE ,record_route_preset [%rm]\n"); >>> } else if (method!="REGISTER") { >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> record_route [%rm]\n"); >>> record_route(); >>> }; >>> >>> from the RECORD ROUTE SECTION is FALSE. >>> >>> Also, i don't understand why the second OK (the one from the >>> reINVITE) is not procesed in the ONREPLY ROUTE, or at least i don't >>> see any statement from the "xlog" in the debug. Is this normal? >>> >>> Thanks.! >>> Regards, >>> >>> Ricardo Martinez.- >>> >>> >>>> -----Mensaje original----- >>>> De: Greger V. Teigre [mailto:greger@teigre.com] >>>> Enviado el: Lunes, 18 de Julio de 2005 2:45 >>>> Para: Ricardo Martinez; serusers(a)lists.iptel.org >>>> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >>>> messages properly? >>>> >>>> >>>> Hi Ricardo, >>>> Thanks for a detailed analysis. Some questions: >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>>> - You didn't show the ngrep trace. The script uses nat=yes in >>>> the Route >>>> header of the INVITE to detect a nat'ed client. Can you >>>> verify that the >>>> reINVITE has the nat=yes? >>>> - You haven't showed the mediaproxy log (it will show the >>>> callers reporting >>>> in etc). That could help (default /var/log/messages) >>>> - You can put a log statement in the loose_route section >>>> after the test for >>>> nat=yes to see if use_media_proxy was called >>>> >>>> g-) >>>> >>>> Ricardo Martinez wrote: >>>>> Hello. >>>>> I'm having problems trying to make SER, NAT'd endpoints >>>> and reINVITE >>>>> work together. >>>>> I was using the "gw-pstn3.07.cfg" file from onsip.org to do some >>>>> tests, and this is what i have. In one side i have an >> Asterisk with >>>>> an endpoint registered in it (let's call it A). In the >> other side i >>>>> have a PAP2 under NAT (let's call it B). >>>>> >>>>> >>>>> A ---------- Asterisk ----------- SER ----------- B (NAT'd) >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>>> 10.0.0.4 >>>>> >>>>> When i make a call from "A" to "B" this is what i see (in terms of >>>>> SDP). Looking from SER. >>>>> >>>>> A --------- Asterisk ------------ SER ------------ B (NAT'd) >>>>> Public: >>>>> 200.0.0.4 >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>> Inside: >>>>> 10.0.0.1 >>>>> >>>>> INVITE >>>>> c:200.0.0.6:19996 >>>>> -------------------> >>>>> INVITE >>>>> c:200.0.0.5:35010 >>>>> ----------------> >>>>> >>>>> >>>>> Caller Via Called Status >>>>> Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> ------------ >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - ?.?.?.?:? inactive 0'04" >>>>> Unknown Audio 0/0/0 >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> >>>>> So far is ok..........and the phone is answered >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- (the phone is >>>>> answered) >>>>> OK >>>>> c:200.0.0.5:35010 >>>>> <--------------------- >>>>> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ---------------------> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ----------------> >>>>> >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <--------------------- >>>>> >>>>> Finally according to the "session" information : >>>>> >>>>> Caller Via Called >>>>> Status Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> -------------- >>>>> ---------- >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - 200.0.0.7:16420 inactive >>>>> 0'26" G729 Audio 0/11.48k/11.48k >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> And the audio is only in one way. :( >>>>> >>>>> So. you can see the reINVITE message apparently is not being >>>>> processed as a call to a NAT'd endpoint and therefore is not using >>>>> the mediaproxy, you can see the second "OK" messsage has >> the invalid >>>>> IP from the NAT'd user is in his sdp information. >>>>> As i said it before i am using the gw-pstn configuration >>>> file from the >>>>> onsip.org and as far as i can remember this configuration >> can handle >>>>> the reINVITE? isn't >>>>> I'm also using the last version of the mediaproxy (1.3.1). >>>>> Can someone tell me what i'm doing wrong? >>>>> >>>>> Hope someone could help me here. >>>>> Thanks in advance. >>>>> Regards... >>>>> >>>>> Ricardo Martinez.- >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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Re: [Serusers] Problem : Can SER process the reINVITE messages properly?
by Greger V. Teigre 01 Aug '05

01 Aug '05

Ricardo, I think you are correct in your analysis of mediaproxy not accepting a third party. Also, I saw the post from somebody on the newest mediaproxy version. Have you tried it? Version 1.2.1 created a new session for reINVITEs, that's why the nat=yes is used in the ONsip.org configs. Version 1.3.0 added support for detecting that a reINVITE came in and would reuse the session (which I btw don't think is good for your scenario as you change the IP). 1.4.2 seems to address your problem,… [View More] so it would be interesting to hear if you have any success. As for the reINVITE with the IP address instead of domain, that's ok. g-) Ricardo Martinez wrote: > Hello Greger and SER users. > About this issue here are some test that i made. I added the > lookup("location") as you recommend me : > > # > > > > ----------------------------------------------------------------- # > Loose Route Section # > ----------------------------------------------------------------- if > (loose_route()) { > > if (has_totag() && (method=="INVITE" || > method=="ACK")) { lookup("location"); > if (isflagset(6) || client_nat_test("3") || > search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > break; > }; > > > Unfortunately, i'm still unable to have voice in both ways, but this > modification seems to handle better the reINVITE message. And this > is why. For the normal approach (i mean the initial onsip_pstn > configuration) the second INVITE was not using the mediaproxym as > Greger pointed. Now, as you can see in the debugs that i'm > attaching, the second invite is using the mediaproxy, and therefore > the "c" parameter in the SDP message is being correctly modified by > SER. So, i don't understand why this time the call is failing. Could > it be that mediaproxy is not capable to handle the reINVITEs?. For > example in the debug from mediaproxy (i'm also attaching this one), i > see a warning message like this : warning: Received packet from a > third party: 200.0.0.7:16432. It seems like mediaproxy is not > accepting "voice packets" from the second IP (the one indicated by > the reINVITE). Why? > Also, when the call is established this is the output from the > "session" command : > > Caller Via Called > Status Duration Codec Type Traffic > ---------------------------------------------------------------------------- > -------------------------------------- > 200.0.0.6:17480 - 200.0.0.5:35094 - 200.0.0.4:46115 inactive > 0'24" G729 Audio 0/16.75k/16.75k > > The call is established but between the first IP (Asterisk box) and > the endpoint public IP. > So, could this be a problem with mediaproxy?. I read the changelog > from the version 1.3.1 and this is what i found : > > Changes from version 1.2.1 to 1.3.0 > ----------------------------------- > > - Only create a new session if a request belongs to the first INVITE > (no longer creates new sessions on re-INVITEs) > > could it be related?. > > I also have a general question about the reINVITE. As you can see > the first INVITE has in his URI the domain name for the number : > 5555848114(a)sipvoiss.desarrollo.redvoiss.net (Frame 1 from the debug). > The reINVITE sent by Asterisk has the URI : 5555848114(a)200.0.0.4. Is > this ok?, despite of this the reINVITE is correctly redirected to the > endpoint. (You can check this in the frame 13 from the debug). > > Thanks again. > > Regards, > > Ricardo Martinez.- > > > >> -----Mensaje original----- >> De: Greger V. Teigre [mailto:greger@teigre.com] >> Enviado el: Miércoles, 20 de Julio de 2005 3:06 >> Para: Ricardo Martinez; serusers(a)lists.iptel.org >> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >> messages properly? >> >> >> Hm. This was a tricky one. I would like to ask a second >> opinion from you >> NAT gurus on my musings below. What is the most appropriate >> response to >> Ricardo's scenario? WARNING to others: This is getting fairly >> technical on >> NAT/SDP... ;-) >> >> The reINVITE from Asterisk has a valid IP address and a valid >> (but new) SDP >> media ip (C=IN IP4 200.0.0.7). nat=yes is correctly not >> present in the >> reINVITE from Asterisk (it is not NATed). Thus >> use_media_proxy is correctly >> not used in the reINVITE, i.e. client_nat_test("3") and test >> on nat=yes will >> not trigger under the has_totag if statement (loose route >> processing): >> >> if (loose_route()) { >> >> if (has_totag() && (method=="INVITE" || method=="ACK")) { >> >> if (client_nat_test("3") || search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> }; >> >> route(1); >> break; >> }; >> >> In the OK from the Linksys: >> U 200.0.0.5:5060 -> 200.0.0.6:5060 >> SIP/2.0 200 OK. >> To: >> <sip:5555848114@sipvoiss.desarrollo.redvoiss.net>;tag=64d7492d >> b018b13ai0. From: "5565625858196" >> <sip:5565625858196@sipvoiss.desarrollo.redvoiss.net>;tag=as5d352c9a. >> Call-ID: >> 0848ef78297116742562204e2c04cbcb(a)sipvoiss.desarrollo.redvoiss.net. >> CSeq: 104 INVITE. >> Via: SIP/2.0/UDP 200.0.0.6:5060;branch=z9hG4bK518d870e. >> Record-Route: <sip:200.0.0.5;ftag=as5d352c9a;lr=on>. >> Contact: 5555848114 <sip:5555848114@200.0.0.4:46081>. >> Server: Linksys/PAP2-2.0.10(LSb). >> Content-Length: 237. >> Content-Type: application/sdp. >> . >> v=0. >> o=- 33068 33068 IN IP4 192.168.15.200. >> s=-. >> c=IN IP4 192.168.15.200. >> t=0 0. >> m=audio 16424 RTP/AVP 18 100 101. >> a=rtpmap:18 G729/8000. >> a=rtpmap:100 NSE/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-15. >> a=ptime:30. >> a=sendrecv. >> >> ...you have 192.168.15.200, a private c= address in SDP, but >> the Contact has >> correctly been fixed (200.0.0.4:46081). So it seems the OK is >> processed >> correctly in onreply even though it does not show up in the >> debug. Flags 6 >> and 7 are not set, so use_media_proxy is correctly not run. >> The problem is >> that flag 6 should have been set (the Linksys is NATed), but >> as the reINVITE >> is not processed using lookup("location") the flag will not >> be picked up >> from usrloc. >> ---------------------------------------- >> >> However, now that Asterisk in the reINVITE is requesting >> another (valid) IP >> as media destination, the question is: what is propor behavior? >> >> The most natural would be to add a lookup("location") after: >> if (has_totag() && (method=="INVITE" || method=="ACK")) >> and then add isflagset(6) to the or test: >> if (isflagset(6) || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> However, I'm not sure the lookup will work, because the R-URI >> is likely to >> be the direct address (5555848114@200.0.0.4:46081) and not >> the registered >> URI (@domain.com). It would be tempting to try to see if we >> somehow could >> make nat=yes set in the early INVITE. We would have to split the >> record_route section in two. One to handle !="REGISTER && >> !="INVITE" with a >> regular record_route() and wait until AFTER >> lookup("location") (maybe in >> route[1], the default handler) to add the following: >> if(method=="INVITE")){ >> if(isflagset(6)) { >> # INSERT YOUR IP ADDRESS HERE >> record_route_preset("192.0.2.13:5060;nat=yes"); >> } else { >> record_route(); >> } >> } >> >> This would make sure that not only dialogs where caller is >> nated, but also >> where callee is nated, will be tagged as nated. This again >> would cause >> nat=yes to be set and rtp proxy forced, further flag 6 is >> set, and thus the >> OK also processed with use_media_proxy(). >> >> BUT: Would this nat=yes cause any problems in other >> scenarios??? I'm not >> sure, but I don't think so. If that is correct, this is >> maybe the right >> solution. >> --------------------------------------- >> OTHER (SIMPLER?) SOLUTIONS: >> ---------- >> Instead you could add ||client_nat_test("1") together with >> isflagset 6 and 7 >> in your reply route: >> if ((isflagset(6) || isflagset(7) || client_nat_test("1")) && >> (status=~"(180)|(183)|2[0-9][0-9]")) >> >> but it will not work because the reINVITE did not get >> use_media_proxy(). >> >> Or, you could instead add fix_nated_sdp("2"); in your onreply route: >> if (client_nat_test("1")) { >> fix_nated_contact(); >> fix_nated_sdp("2"); >> }; >> WARNING! You will have to add else in between this test and >> the previous to >> avoid messing up use_media_proxy(). >> >> This will fix the SDP (c=IN IP4), but it will probably not >> help, because the >> port is still unknown. You could use "3" instead in >> fix_nated_sdp, this >> will add direction=active to the OK. However, another thread >> stated that >> Asterisk does not support direction=active. If it does, it >> will probably >> work and you get a non-proxied stream between Asterisk and >> the Linksys. >> >> Or finally, the third option would be to add the (hopefully) >> fixed IP in the >> loose_route test: >> if (src_ip==200.0.0.6 || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> This would cause the OK automatically to get use_media_proxy. >> >> Ricardo, could you please test these various solutions and >> see if they >> really work or if (where...) I go wrong :-) >> g-) >> >> Ricardo Martinez wrote: >>> Hello Greger. >>> Thanks for your answer on this topic. Now i'm attaching more debug >>> information (the /var/log/messages from mediaproxy, the ngrep output >>> and some xlog statements in the ser.cfg file) beside some >> comments in >>> the file reINVITE_debug_problem.txt. >>> >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>> >>> At least i have ringback tone. Then, when the call is answered i >>> have an OK and a ACK message coming to my SER box, then inmediatly >>> the reINVITE message arrives Asterisk box. >>> >>> >>> For what i can see from the debug the "nat=yes" is never reached >>> because the caller has a "valid ip" and therefore the first "if" in >>> the statament >>> >>> if (method=="INVITE" && client_nat_test("3")) { >>> # INSERT YOUR IP ADDRESS HERE >>> record_route_preset("64.76.148.246:5060;nat=yes"); >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> invite & client_nat_test(3) TRUE ,record_route_preset [%rm]\n"); >>> } else if (method!="REGISTER") { >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> record_route [%rm]\n"); >>> record_route(); >>> }; >>> >>> from the RECORD ROUTE SECTION is FALSE. >>> >>> Also, i don't understand why the second OK (the one from the >>> reINVITE) is not procesed in the ONREPLY ROUTE, or at least i don't >>> see any statement from the "xlog" in the debug. Is this normal? >>> >>> Thanks.! >>> Regards, >>> >>> Ricardo Martinez.- >>> >>> >>>> -----Mensaje original----- >>>> De: Greger V. Teigre [mailto:greger@teigre.com] >>>> Enviado el: Lunes, 18 de Julio de 2005 2:45 >>>> Para: Ricardo Martinez; serusers(a)lists.iptel.org >>>> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >>>> messages properly? >>>> >>>> >>>> Hi Ricardo, >>>> Thanks for a detailed analysis. Some questions: >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>>> - You didn't show the ngrep trace. The script uses nat=yes in >>>> the Route >>>> header of the INVITE to detect a nat'ed client. Can you >>>> verify that the >>>> reINVITE has the nat=yes? >>>> - You haven't showed the mediaproxy log (it will show the >>>> callers reporting >>>> in etc). That could help (default /var/log/messages) >>>> - You can put a log statement in the loose_route section >>>> after the test for >>>> nat=yes to see if use_media_proxy was called >>>> >>>> g-) >>>> >>>> Ricardo Martinez wrote: >>>>> Hello. >>>>> I'm having problems trying to make SER, NAT'd endpoints >>>> and reINVITE >>>>> work together. >>>>> I was using the "gw-pstn3.07.cfg" file from onsip.org to do some >>>>> tests, and this is what i have. In one side i have an >> Asterisk with >>>>> an endpoint registered in it (let's call it A). In the >> other side i >>>>> have a PAP2 under NAT (let's call it B). >>>>> >>>>> >>>>> A ---------- Asterisk ----------- SER ----------- B (NAT'd) >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>>> 10.0.0.4 >>>>> >>>>> When i make a call from "A" to "B" this is what i see (in terms of >>>>> SDP). Looking from SER. >>>>> >>>>> A --------- Asterisk ------------ SER ------------ B (NAT'd) >>>>> Public: >>>>> 200.0.0.4 >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>> Inside: >>>>> 10.0.0.1 >>>>> >>>>> INVITE >>>>> c:200.0.0.6:19996 >>>>> -------------------> >>>>> INVITE >>>>> c:200.0.0.5:35010 >>>>> ----------------> >>>>> >>>>> >>>>> Caller Via Called Status >>>>> Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> ------------ >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - ?.?.?.?:? inactive 0'04" >>>>> Unknown Audio 0/0/0 >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> >>>>> So far is ok..........and the phone is answered >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- (the phone is >>>>> answered) >>>>> OK >>>>> c:200.0.0.5:35010 >>>>> <--------------------- >>>>> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ---------------------> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ----------------> >>>>> >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <--------------------- >>>>> >>>>> Finally according to the "session" information : >>>>> >>>>> Caller Via Called >>>>> Status Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> -------------- >>>>> ---------- >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - 200.0.0.7:16420 inactive >>>>> 0'26" G729 Audio 0/11.48k/11.48k >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> And the audio is only in one way. :( >>>>> >>>>> So. you can see the reINVITE message apparently is not being >>>>> processed as a call to a NAT'd endpoint and therefore is not using >>>>> the mediaproxy, you can see the second "OK" messsage has >> the invalid >>>>> IP from the NAT'd user is in his sdp information. >>>>> As i said it before i am using the gw-pstn configuration >>>> file from the >>>>> onsip.org and as far as i can remember this configuration >> can handle >>>>> the reINVITE? isn't >>>>> I'm also using the last version of the mediaproxy (1.3.1). >>>>> Can someone tell me what i'm doing wrong? >>>>> >>>>> Hope someone could help me here. >>>>> Thanks in advance. >>>>> Regards... >>>>> >>>>> Ricardo Martinez.- >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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RE: [Serusers] LCR on onsip.org - No DB tables created?
by Dave 30 Jul '05

30 Jul '05

I used exactly that script. But It did not create any LCR related tables.!!! --- Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > Hello Dave. > If i'm not wrong, there is a shell script included > in the last SER > version to create all the DB tables related to SER. > I think there is in : > /sip_router/scripts/ and is called "ser_mysql.sh" > You can try that!. > > Regards, > Ricardo Martinez > > > > > -----Mensaje original----- &… [View More]

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RE: [Serusers] LCR on onsip.org - No DB tables created?
by Dave 30 Jul '05

30 Jul '05

Thanks Ricardo - Do I need to add these to "versions" table in ser database? If yes, what is the version number for these tables? Dave --- Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > Hello Dave. > I was almost sure that the tables for LCR were > defined there!, my mystake. > But, i have the table definitions (i can't rememeber > where i found it). You > can check with Juha if there is a mistake in the > tables. > > # > # Table structure for table … [View More]'gw' > # > > CREATE TABLE gw ( > gw_name VARCHAR(128) NOT NULL, > ip_addr INT UNSIGNED NOT NULL, > port SMALLINT UNSIGNED, > grp_id INT UNSIGNED NOT NULL, > PRIMARY KEY (gw_name), > KEY (grp_id) > ); > > # > # Table structure for table 'gw_grp' > # > > CREATE TABLE gw_grp ( > grp_id INT UNSIGNED NOT NULL AUTO_INCREMENT > PRIMARY KEY, > grp_name VARCHAR(64) NOT NULL > ); > > # > # Table structure for table 'lcr' > # > > CREATE TABLE lcr ( > prefix varchar(16) NOT NULL, > from_uri varchar(128) NOT NULL DEFAULT '%', > grp_id INT UNSIGNED NOT NULL, > priority TINYINT UNSIGNED NOT NULL, > KEY (prefix), > KEY (from_uri), > KEY (grp_id) > ); > > > Regards > Ricardo.- > > > -----Mensaje original----- > > De: Dave [mailto:ddx66@yahoo.com] > > Enviado el: Viernes, 29 de Julio de 2005 11:30 > > Para: Ricardo Martinez; serusers(a)lists.iptel.org > > Asunto: RE: [Serusers] LCR on onsip.org - No DB > tables created? > > > > > > I used exactly that script. But It did not create > any > > LCR related tables.!!! > > > > > > --- Ricardo Martinez <rmartinez(a)redvoiss.net> > wrote: > > > > > Hello Dave. > > > If i'm not wrong, there is a shell script > included > > > in the last SER > > > version to create all the DB tables related to > SER. > > > I think there is in : > > > /sip_router/scripts/ and is called > "ser_mysql.sh" > > > You can try that!. > > > > > > Regards, > > > Ricardo Martinez > > > > > > > > > > > > > -----Mensaje original----- > > > > De: Dave [mailto:ddx66@yahoo.com] > > > > Enviado el: Viernes, 29 de Julio de 2005 11:16 > > > > Para: serusers(a)lists.iptel.org > > > > Asunto: [Serusers] LCR on onsip.org - No DB > tables > > > created? > > > > > > > > > > > > Hi, I downloaded and installed the ser-0.9.3 > from > > > > onsip.org with the intention of using LCR. > > > Although, > > > > the LCR module was complied, I do not see the > > > tables > > > > lcr, gw in the DB? How do I create them? > > > > Thanks > > > > > > > > > __________________________________________________ > > > > Do You Yahoo!? > > > > Tired of spam? Yahoo! Mail has the best spam > > > protection around > > > > http://mail.yahoo.com > > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > serusers(a)lists.iptel.org > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam > protection around > > http://mail.yahoo.com > > > ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs [View Less]

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[Serusers] Parrallel forking using SER and Mediaproxy
by Ramu Yadav 30 Jul '05

30 Jul '05

Hi all, I am experiencing some problems while configuring SER in order to fork between different contacts (a SIP UA and a SIP Gateway) and passing through a mediaproxy: The scenario is the following: (3 calls) UA1 -> SER : INVITE sip:ramu at xyz.com SER forks: SER -> UA2 : INVITE sip:ramu@[local-ip] {SDP is changed by mediaproxy module} SER -> GW: INVITE sip:PSTN-NUMBER@[gateway-ip] {SDP is changed by mediaproxy module} SER -> GW: INVITE sip:PSTN-NUMBER@[gateway-ip] {… [View More]

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