- sr-users - kamailio.org

[Serusers] SIP client with video???
by Joao Pereira 06 Sep '06

06 Sep '06

Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software..... Does someone knows about a better software? Regards Joao Pereira

2 2

[Serusers] log ( history of action for one user)
by Legos Engineering 06 Sep '06

06 Sep '06

HI. I wonder, if there is some log (like messages or another) which permit to know the history of one user ( his last calls, when he was called ...). I want to know that in order to add, an history of each user in the webcare. Therefore, if someone can say me, if this log exist and where it is. Thank you.

2 1

[Users] How to redirect based on from_uri and uri
by Darren DeLitizia 06 Sep '06

06 Sep '06

Here is what I want to do. Domestic routing defining interlata and intralata based off of ani and dnis (from uri and uri). Now let's not dive too much into the whole domestic routing thing or we'll be here for the next century. Bottom line is, this what I want to do with openser. 1. Look at ani (from_uri) (NPA - first 3 digits) 2. Look at dnis (uri) - (NPA - first 3 digits) 3. If the NPA (first 3 digits) in ani is in same as the DNIS NPA then redirect 555dnis@ip_address 4. If ANI NPA is in same as DNIS NPA then redirect 999dnbis@ip_address I am redirecting so I can be and stay stateless I am able to redirect but unable to figure out correct syntax for it to look at ani and dnis then redirect appropriately. Thanks all

2 1

[Users] crash
by Helge Waastad 06 Sep '06

06 Sep '06

Hi, in my 1.1.0 version I syddenly got a core dump. Does any one know why? (gdb) bt #0 0x000000304448660b in strftime_l () from /lib64/tls/libc.so.6 #1 0x0000002a95b29de4 in time2mysql (_time=Variable "_time" is not available. ) at utils.c:54 #2 0x0000002a95b2a731 in val2str (_c=Variable "_c" is not available. ) at val.c:147 #3 0x0000002a95b26b13 in print_values (_c=0x643648, _b=0x2a95db5269 "'SIP/2.0 100 Trying\\r\\nVia: SIP/2.0/UDP 217.17.222.4;branch=z9hG4bK1957.c42a1af6.0\\r\\nVia: SIP/2.0/UDP 10.1.93.186:10001;rport=44323;received=217.17.222.6;branch=z9hG4bK2eb74daf34599095\\r\\nFrom: \\\"Har"..., _l=65431, _v=0x7fbfffe990, _n=10) at dbase.c:144 #4 0x0000002a95b27aa4 in db_insert (_h=0x643e08, _k=Variable "_k" is not available. ) at dbase.c:495 #5 0x0000002a97c26bdc in trace_sl_onreply_out (req=Variable "req" is not available. ) at siptrace.c:1078 #6 0x0000002a956b30ad in run_sl_callbacks (req=0x643f28, buffer=Variable "buffer" is not available. ) at sl_cb.c:88 #7 0x0000002a956b3608 in sl_send_reply (msg=0x643f28, code=100, text=0x608ee8 "Trying") at sl_funcs.c:149 #8 0x0000000000409c38 in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:701 #9 0x000000000040a87e in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #10 0x000000000040b43d in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #11 0x000000000040b496 in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #12 0x000000000040b762 in run_top_route (a=Variable "a" is not available. ) at action.c:89 #13 0x000000000043188c in receive_msg (buf=0x2a97e4d2f8 "\026", len=822, rcv_info=0x7fbffff810) at receive.c:155 #14 0x000000000044fc74 in udp_rcv_loop () at udp_server.c:465 #15 0x000000000041f7f4 in main_loop () at main.c:925 #16 0x0000000000420446 in main (argc=Variable "argc" is not available. ) at main.c:1477 -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 5

[Users] aliases in multi-domain
by Ricardo Carvalho 06 Sep '06

06 Sep '06

I'm using multi-domain and I can only call users by their aliases if I'm dialling from a phone registered with the same domain of the called user or if I'm dialling alias@domain_of_the_callee from a phone registered with other domain. How can I make Ser allow that every user can contact other users even if they are from other domain by just dialling their alias (without dialling also their domain when the callee is from other domain than the caller)? Regards, Ricardo.

2 3

[Users] "BYE" messages change path
by ravi reddy 06 Sep '06

06 Sep '06

Hi users, This is ngrep report regarding my problem; -----------------<when a sip proxy behind NAT and the user registerd to that sip-proxy calls>------------------------- U 82.102.69.105:39871 -> 81.21.33.35:5060 INVITE sip:99106883@81.21.33.35:5060 SIP/2.0. To: "99106883"<sip:99106883@81.21.33.35:5060>. From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4(a)192.168.1.102. CSeq: 47344 INVITE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 361. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 82.102.69.105.---------------------------------------------> this is the NAT for the sip-proxy , t=0 0. m=audio 5004 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. -----------------------------<when my SIP-SERVER standing on public ip recieved the "INV" message from the above"----------------------------------- U 81.21.33.35:5060 -> pstngw:5060 INVITE sip:99106883@pstngw:5060 SIP/2.0. Record-Route: <sip:99106883@81.21.33.35 :5060;nat=yes;ftag=c86b66ad8b9187c8;lr=on>. To: "99106883"<sip:99106883@81.21.33.35:5060>. From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK0ab.9522bc25.0. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport=39871. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4(a)192.168.1.102. CSeq: 47344 INVITE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:82.102.69.105:39871>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 360. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 81.21.33.35.--------------------------------------------->This is my SIP-SERVER standing on public ip (and is sending to pstngw) t=0 0. m=audio 60516 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. ---------------------------------------------<and the pstngw is making call>----------------------------------------------- U 81.21.33.35:5060 -> pstngw:39871 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105;branch=z9hG4bK-d87543-ac3b034487ba16641188e4c9e5ad0664-1--d87543-;rport=39871,SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK01c64e769ba6c176. From: "12345"<sip:12345@81.21.33.35:5060>;tag=9ca5c1fe3b43aad6. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E6010EC-1764. Date: Wed, 06 Sep 2006 10:27:40 GMT. Call-ID: 72d0e4e8adca2ab2(a)192.168.1.102. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 32352 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 237. . v=0. o=CiscoSystemsSIP-GW-UserAgent 54 9902 IN IP4 pstngw s=SIP Call. c=IN IP4 81.21.33.35.-------------------------------------<in pstngw it noticed the contact header of SIP-SERVER>--------------------- t=0 0. m=audio 60518 RTP/AVP 18 100. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:100 X-NSE/8000. a=fmtp:100 192-194. a=ptime:20. ---------------------------------------------<after talking some time if any of UA hung the phone (here i am showing from pstn hung up) the result goes like this---------------- U pstngw:52991 -> 81.21.33.35:5060 BYE sip:99106883@81.21.33.35:5060;nat=yes;ftag=1bab08cdaad1d6e8;lr=on SIP/2.0. Via: SIP/2.0/UDP 81.21.38.15:5060. From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292. To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 6. Route: <sip:192.168.1.100:5060>, <sip:82.102.69.105:39871>.------------------------> What is happening here ??????? Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. . # U 81.21.33.35:5060 -> 192.168.1.100:5060--------------------------------------------------->here unexpected problem arises for me it have to use 82.102.69.105 BYE sip:192.168.1.100:5060 SIP/2.0. Record-Route: <sip:81.21.33.35;ftag=7E66E3D8-1292;lr=on>. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bKb0f1.3fe32691.0. Via: SIP/2.0/UDP pstngw:5060. From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292. To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 5. Route: <sip:82.102.69.105:39871>. Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. P-hint: rr-enforced. So the call never ends up from SIP SERVER to other party Make some comments on above and assist me where am I going wrong: and i am using www.openser.org pstn default script as ser.cfg Hope to get some help Thanks, Ravi.

1 0

[Serusers] SER & Asterisk (IVR problem)
by Roberto Lopes 06 Sep '06

06 Sep '06

Hi to all! I've a big problem with SER & Asterisk I've a network where i put them working together. Asterisk is the media gateway with PSTN as well as being the server which is making all the services available (IVR, voicemail, conference calls, etc). SER just routes calls either to the softphones, VoIP phones or to Asterisk who will just forward the calls to PSTN (via Asterisk). Till now everything works well except the service IVR which is working on Asterisk. Details: 192.168.226.13 - IP SER 192.168.226.45 - IP Asterisk The problem is only after the user (roberto(a)192.168.226.13 - authenticated and authorized in RADIUS) calls IVR. Till then the user calls, IVR gives some information to the user and everything goes well. Problem is when he presses a number which will have to redirect the call to another user. And why is there a problem. Because the initially roberto(a)192.168.226.13 to be able to call IVR had to rewrite host & port 192.168.226.45 so that the call would reach Asterisk who would then call to the IVR. That rewriting turned roberto(a)192.168.226.13 into roberto(a)192.168.226.45. That is, I can say, that it became a "new user". While it's on the IVR there's no problem, but when call is diverted it has to give its credentials... Credentials that don't exist in RADIUS cause it only accepts usernames that end in @192.168.226.13 (IP of SER) Here is the following code, both of extensions file in Asterisk and configure file of SER I hope somebody knows how to solve it or at least find a way how to. extensions.conf - Asterisk [sip] exten => 74001,1,Answer exten => 74001,2,SetMusicOnHold(default) exten => 74001,3,Set(TIMEOUT(digit)=5) exten => 74001,4,Set(TIMEOUT(response)=10) exten => 74001,5,Background(menu) exten => 1,1,Dial(SIP/roberto@192.168.226.13:5060,,r) ... exten => 7,1,Goto(sip,74001,1) exten => 8,1,Hangup ser.cfg - SER ... if ((uri=~"^sip:7[0-9]{4}@.*")) { route(5); break; }; ... route[5] { # Redirecting to Asterisk # MsgLog xlog("L_ALERT","(WARNING) Sending to Asterisk request of %fu to %tu\n"); rewritehostport("192.168.226.45:5060"); route(1); } Thanks for your help, Roberto Lopes

2 1

[Users] "BYE" messages change path
by ravi reddy 06 Sep '06

06 Sep '06

Hi Users, I am using SER-0.9.6 on public domain . and I am using another Sip-proxy behind NAT (ip-address=192.168.1.100:5060) i have some clients registered to that nated sip-proxy and here that nated sip-proxy forwards PSTN calls to SER on public domain. when call make from client (registered to nated sip-proxy) the session establishing suscessfully , but when the client hung the phone the SER on public domain is taking the BYE message and is not sending the BYE message to the other end and it is sending directly to the Sip-proxy which is a private-ip . So, here SER is not getting ACK and its keep on sending the BYE requests to the privateip instead of sending to the NAT address . In some docs which jiri wrote on record route he mentioned the same problem below is the link ; but he dint mention how to solve that record route problem, If SER send messages back as it recieving path (from NAT)there is no problem , but its changing the route to send BYE , For the request "CANCEL","ACK","INVITE","200 OK" every thing is happening in the same Route but it is not happening only for the "BYE" request . Hope some body can help me. Thank You. here is the link which i told;-http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/…<http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/meetin…> Regards, Ravi.

1 0

[Serusers] aliases in multi-domain
by Ricardo Carvalho 06 Sep '06

06 Sep '06

I'm using multi-domain and I can only call users by their aliases if I'm dialling from a phone registered with the same domain of the called user or if I'm dialling alias@domain_of_the_callee from a phone registered with other domain. How can I make Ser allow that every user can contact other users even if they are from other domain by just dialling their alias (without dialling also their domain when the callee is from other domain than the caller)? Regards, Ricardo.

2 3

Re: [Users] more on accounting
by tele 06 Sep '06

06 Sep '06

Glenn Dalgliesh wrote: >This is from previous list > >"i Glenn, > >For this feature to work, you need to be sure that you do record_route() on >the INVITE and that "append_fromtag" RR module param is not disable. > >regards, >bogdan " > > Yes i have read that thread. i always record_route all messages before do things. if (!method=="REGISTER") record_route(); by the sample openser.cfg and for be sure i have also forced: modparam("rr","append_fromtag",1) >-----Original Message----- >From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf >Of tele >Sent: Wednesday, June 28, 2006 11:05 AM >To: users(a)openser.org >Subject: [Users] more on accounting > >Hi all, > >I'm using OpenSER v1.0.1 and i have done an accounting compatible with >my billing system. >Now i have problem with detection of flow direction for billing reason. > >i'm talking about openser 1.0.x: > >the problem is when the BYE come from callee. >i know the problem is solved in openser1.1.x with the > >modparam("acc", "detect_direction", 1) > >i have read in modules documentation of 1.0.x that i can use >is_direction() of RR module for detect call flow direction but it says >that this must be called after loose_route() > >so for the accounting i must call setflag before loose_route() but what >can i do if i want detect call flow >before of exec setflag? > >Hmmm i think that i must install OpenSER 1.1.x and test with it > >:tele > > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Serusers] Media Proxy and 2 interfaces (Private and Public)
by Kyriakos 06 Sep '06

06 Sep '06

Hello, Is there a way to have a machine that hosts media proxy have its one interface in a private network and the other on a public? >From what I read rtpproxy supports bridging which is basically what I want to do but I have not found anything for mediaproxy. Thanks KM

2 1

[Serusers] Sip Proxy behind Nat.
by ravi reddy 06 Sep '06

06 Sep '06

Hi Users, I am Using SER-0.9.6 on a public ip and using another sip-proxy(not SER) behind the NAT. and when the users who registerd to Sip-proxy which is behind NAT,, calls to the users who registered to public domain SER.. call is being established . but when hunging the phone the SER is not sending "BYE" messages to the NAT device it is sending to the Sip-proxy (not SER) that is for a private IP . so even when i hung the call from the Nated Sip proxy ,SER is getting messages through NAT but again SER is not sending "BYE" messages to other party its sending back to the nated sip-proxy having private IP , so the messages are going in a wrong direction and is being wasted . It looks like this is a BUG in SER , i would take this oppurtunity to report this. If this is not a bug please send me the suggestions ..... To solve my problem. thank you. regards, Ravi.

2 1

[Serusers] server error message " 500 I'm terribly sorry, server error occurred (1/SL)".
by ravi reddy 06 Sep '06

06 Sep '06

Hi users , I am proceeding in this way i have a SER-0.9.6 running fine :-) and I had given a username and password to a call-shop and this callshop owner with his username and password he connects to another 6 phones, Actually he bills to his 6 phones and I will bill him ."o.k overall Scenario is well and good". I made a little changes in ser.cfg and when the call made from the call shop up to call connecting is O.K but when we hung the phone the SER is not generating "BYE" messages to other party , so the call is on.. and i am not getting "Acct Stop" packet also SO How I can solve my problem :-( any suggestions will be appreciated . below is the message i am getting from SER when I hung the phone on one side -------------------<this message is came from callshop Nat address>---------<"it sends bye to my SER "------------------------------------------------------ U 82.102.69.105:32768 -> 81.21.33.35:5060 BYE sip:99106883@81.21.33.35:5060 SIP/2.0. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C. From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9. Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102. CSeq: 9533 BYE. Route: <sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a>. Route: <sip:192.168.1.100:5060>. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Proxy-Authorization: Digest username="12345", realm="81.21.33.35", algorithm=MD5, uri="sip:192.168.1.100:5060", nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde", response="4001812db780010a557b40683efd2f9e". Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 0. . ----------------<here it is as soon as SER recieve Bye message it has to send Bye to other party >------<But it is sending to the hung up phone itself>--------- # U 81.21.33.35:5060 -> 192.168.1.100:5060 BYE sip:192.168.1.100:5060 SIP/2.0. Record-Route: <sip:81.21.33.35;ftag=837e5b2ff0b4cf4a;lr=on>. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C. From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK936f.7ff7572.0. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105 ;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9. Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102. CSeq: 9533 BYE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Proxy-Authorization: Digest username="12345", realm="81.21.33.35", algorithm=MD5, uri="sip:192.168.1.100:5060", nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde", response="4001812db780010a557b40683efd2f9e". Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 0. . -----------------<And here I go iam getting this message and the call is not being stopped >---------------------------------- # U 81.21.33.35:5060 -> 82.102.69.105:32768 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C. From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768;received= 82.102.69.105. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9. Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102. CSeq: 9533 BYE. Content-Length: 0. Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=27773 req_src_ip= 82.102.69.105 req_src_port=32768 in_uri=sip:99106883@81.21.33.35:5060 out_uri=sip:192.168.1.100:5060 via_cnt==2". . any suggestions will be appreciated: Thank You.

2 1

[Serusers] Record Route problem :-(
by ravi reddy 06 Sep '06

06 Sep '06

Hi SER Users, I am using SER-0.9.6 on public domain . and I am using another Sip-proxy behind NAT (ip-address=192.168.1.100:5060) i have some clients registered to that nated sip-proxy and here that nated sip-proxy forwards PSTN calls to SER on public domain. when call make from client (registered to nated sip-proxy) the session establishing suscessfully , but when the client hung the phone the SER on public domain is taking the BYE message and is not sending the BYE message to the other end and it is sending directly to the Sip-proxy which is a private-ip . So, here SER is not getting ACK and its keep on sending the BYE requests to the privateip instead of sending to the NAT address . In some docs which jiri wrote on record route he mentioned the same problem below is the link ; but he dint mention how to solve that record route problem, If SER send messages back as it recieving path there is no problem , but its changing the route to send BYE , For the request "CANCEL","ACK","INVITE","200 OK" every thing is happening in the same Route but it is not happening only for the "BYE" request . Hope some body can help me. Thank You. here is the link which i told;- http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/meetin… Regards, Ravi.

1 0

[Users] t_forward() functionality with t_relay()?
by Greg Fausak 06 Sep '06

06 Sep '06

I've got a dumb question. Can I get t_relay() functionality (transactions and whatnot) without getting the Via and potentially Record-Route tags in the header??? I warned you it was dumb.... -g -- Greg Fausak greg(a)thursday.com

3 4

[Users] Error in installing Ruby...Please help...Please
by Ferianto siregar 06 Sep '06

06 Sep '06

Dear all, Thank you very much for your help. A few days ago, I have sent a message that to ask about openser admin installation. Thank you very much for your help. Thanks Now, I have tried to install it again. But, When I compile the rubygerms-0.9.0.tar.gz, I got error message. This is the task that I have done before: 1. Untar the rubygerms packet # tar -zxvf rubygems-0.9.0.tar.gz # cd rubygems-0.9.0 # ruby setup.rb # gem install --include-dependencies When I type this command, "gem install --include-dependencies", I got error message. This is the error message: Attempting local installation of 'rails' Local gem file not found: rails*.gem Attempting remote installation of 'rails' Updating Gem source index for: http://gems.rubyforge.org ERROR: While executing gem ... (Gem::RemoteSourceException) Error fetching remote gem cache: getaddrinfo: Temporary failure in name esolution What`s wrong? I do hope anybody can help me to solve this problem. Please help me...Please help because of my lack skills..Please Note: 1. I do not have connection through the internet, so for rails packet, I have downloaded the rails-1.1.6.tar.gz. I have untar it in root directory(same as rubygems) and type command rails # script/server But, Nothing happens. Why? Please... Thank you very much Regards, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail.

2 1

[Serusers] NAT Solution
by Asep Saifudin 06 Sep '06

06 Sep '06

Hi, Does anyone can help me to choose best NAT solution for SER/OpenSER ?NatHelper or Mediaproxy ? And if they compare with STUN, TURN, ICE technology Nathelper or Mediaproxy better than them ? My target is I want my SIP Server can pass all NAT/Firewall just like SkyPE which hard to be block by firewall and can pass all NAT system. ~Asep

2 1

[Serusers] SER not responding
by Henry Duan 06 Sep '06

06 Sep '06

hi, I'm new to SER. I downloaded SER0.9.6 source to my local Fedora Core 5. I run make all and make prefix=/ install, and then set SIPDOMAIN, and run ser. However it didn't respond with incomging Registration(it showed incoming Registration from Ethereal). (I used default ser.cfg). Can anybody tell me what's the problem, or how to debug? Thanks so much! Henry

3 2

[Users] OSS on openSER and Redriect .
by raviprakash sunkara 06 Sep '06

06 Sep '06

Hi Users, Can any one tell about OSS ( Operation Support System) on openSER, Why we need ? But i think to reduce the load balance on openSER server, Suppose is I use 3 OSS's on SIP server. Sip users should also insert in to 3 OSS sip. And other doubt is ? Suppose if we using the redirect , that forwording into other openser system, When openSER redirect to other openSER, the user should also insert, But i think that .... OSS and redirect on openSER are the same. Please for me in English, I know its naughty questioin, please excuse me ! -- -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Serusers] Free DNS Service
by Tracy Lofton 06 Sep '06

06 Sep '06

Hi All, I am new to OpenSER/SER but have been receiving quite a bit of help and would like to express my gratitude by letting you all know of the FREE DNS Service that I am offering to the Internet Community. Go to http://www.t4tm.net to find out more, signup and enjoy the benefits. Tracy _________________________________________________________________ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

1 0

[Serusers] Preregistering a sip address?
by Ignacio Mas Ivars 06 Sep '06

06 Sep '06

Hi all! I have just started using the SIP Express Router and I am looking for a way to have a user preregistered when the router starts. Is there any way to configure that user1 should be registered by default at an IP address without needing to send the registration? I guess it will have to do with messing a bit with the routing engine, but I don't dare to touch it so far and I don't know how to do it. Any help would be very much appreciated! Thanks very much for any help! /Nacho --- ,,, (o o) --ooO--(_)---Ooo----------- Three may keep a secret, if two of them are dead.- --oo0--------0oo-----------

2 1

[Serusers] Free DNS Service
by Tracy Lofton 06 Sep '06

06 Sep '06

Hi All, I am new to SER/OpenSER and I have received much help and in return, I am offering free Dynamic DNS services to whoever is interested. To find out more go to http://www.t4tm.net, signup and enjoy the benefits. Tracy _________________________________________________________________ Got something to buy, sell or swap? Try Windows Live Expo ttp://clk.atdmt.com/MSN/go/msnnkwex0010000001msn/direct/01/?href=http://exp…

1 0

[Serusers] delay to write things in DB when using serctl
by Ricardo Carvalho 06 Sep '06

06 Sep '06

When I add with serctl some alias to the DB, only after some time it appears written in the aliases table of DB. How can I force that to be simultaneously written in DB? Any parameter I have to tune in ser.cfg? Regards, Ricardo.

2 1

[Users] Re: Request for discussion: t_relay() internal error processing
by Mark Kent 05 Sep '06

05 Sep '06

>> 2) t_relay will return several error codes: >> -1 = no transaction created -> need to use sl_* functions >> -2 = transaction created, relay failed -> may destroy If I understand this correctly, the difference between these two return codes really has to do with who it is that cares about it... -2 relates to this exact call, at this exact moment, chances are the very next call the openser box handles will be OK. It looks to me like -1 means serious trouble, for the entire call switching platform. It sounds like it could happen because of a resource shortage (memory), possibly because of a bug in openser, or a bug in the openser.cfg. If you get -1 once, then the chances that you'll get the same code on the next call are very high. Let's say we've run out of memory. Then, what tools are available at the openser.cfg script level that can handle this? Is there a way to force some internal garbage collection? Or to do a warm-restart (e.g., trash all transactions, but not registrations)? My point here is this: if I'm right about -1 meaning serious trouble that is likely to mean serious problems from now on, then this is a more general problem than just t_relay, and much bigger than just handling this one call or a call to one particular neighbour. It's true that the application may halt and catch fire, but there are plenty of errors that are catchable by openser and the larger question seems to be: what do we do then? If there were script-level tools that could be used to try to manage the failure, possibly clean up (in a large way) and try to keep going, then maybe we should have openser_failure route blocks that the user can register with openser to handle various errors. Thanks, -mark

2 1

[Users] Request for discussion: t_relay() internal error processing
by Bogdan-Andrei Iancu 05 Sep '06

05 Sep '06

Hi all, there are some topics largely discussed lately on the mailing list; in order to fix them problems first we need to find a solution for logical point of view - and here I think that several opinions are better than one :) In order to have failover based on serial forking, it is mandatory to be able to detect from script if the initial t_relay() fails. The problem is that now t_relay() masks any error by internally generating an error SIP reply. Of course, it can be internal errors (like lack of memory), msg errors (parse errors) or transport errors (DNS failed, send/connect failed, etc). Now, disabling the auto replying in case of error is not a problem. The problem is how to handle the errors from the script; two make distinction between the error classes a listed above will not be the purpose of this email ( I will open another one for this ;) ). What is important here is to be able to properly resume from the point were the error popped up. What I mean is: the t_relay() function builds the transaction and sends the requests. If the error occurred during transaction creation, from script, you will had to use only stateless functions to send err. replies. If the transaction was already created and the error popped up during relaying stage, you will need from the script to take care of the transaction and either destroy it, either reply it. Based on this, here is what I suggest: 1) have a new param for disabling the auto reply in t_relay() 2) t_relay will return several error codes: -1 = no transaction created -> need to use sl_* functions -2 = transaction created, relay failed -> may destroy transaction, reply via TM with error, trigger failover. any comments on this ? regards, bogdan

4 4

[Users] mediaproxy error
by Richard Z 05 Sep '06

05 Sep '06

Hi, I tried the lateset mediaproxy. When starting, got these errors and dispatcher won't start, /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/proxydispatcher.py", line 65, in ? /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 39, in ? /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/modules/accounting.py", line 14, in ? /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/modules/mysql.py", line 8, in ? #ls /usr/local/mediaproxy/modules/ accounting.py daemon.py dispatcher.py mysql.py request.py serialize.py version.pyc accounting.pyc daemon.pyc dispatcher.pyc mysql.pyc request.pyc utilities.py configuration.py datatypes.py external options.py rtphandler.py utilities.pyc configuration.pyc datatypes.pyc formatting.py options.pyc rtphandler.pyc version.py Any suggestion? Thanks, Richard

1 0

[Serusers] No calls shown in serweb under accounting tab
by Chahn John Kim 05 Sep '06

05 Sep '06

Sorry for double posting. I forgot to put subject line in previous posting. Hi, all. I just installed serweb and it is working however there is are no calls shown under "accounting" tab, saying "no calls". Mysql table has call records with matching flag set in ser.cfg. Records seem to be correct for numerous test calls made. It just that serweb doesn't seem to pull those records out of mysql at all. Any idea where I should look at? Thanks in advance. -- Chahn John Kim

1 0

[Serusers] (no subject)
by Chahn John Kim 05 Sep '06

05 Sep '06

Hi, all. I just installed serweb and it is working however there is are no calls shown under "accounting" tab, saying "no calls". Mysql table has call records with matching flag set in ser.cfg. Records seem to be correct for numerous test calls made. It just that serweb doesn't seem to pull those records out of mysql at all. Any idea where I should look at? Thanks in advance. -- Chahn John Kim

1 0

[Serusers] prevent a dialled username without dialling the domain to ring a phone registered with that username
by Ricardo Carvalho 05 Sep '06

05 Sep '06

How can I prevent a dialled username (without dialling the domain) to ring a phone registered with that username? I pretend to use a multi-domain architecture where apart from aliases of users, only dialling with username@domain will ring registered phones. Thanks, Ricardo.

2 1

[Users] Proxy to Proxy Communication
by shiva sankar padhy 05 Sep '06

05 Sep '06

Dear Members! I am very new to This VOIP Technology. I have just installed the Openser. I am now able to Do the Communication between the users who are resisterd in the proxy but how do I establish the calls to another proxy.I mean Lets consider I have two Proxies and I want a user of Proxy 1 to coomunicate with a user of Proxy2.How will I edit the configuration files so that I can achieve that. And One more thing.How do I achieve the Nat Traversal in openser.Let us consider I am running my Server in 24.10.65.90(Public and static).and I have a Lan connection In the client side, When the two phones from the same LAN want to talk with each other by using the proxy at 24.10.65.90, Its not possible.I got Disconnected from the internet connection and also when I tried this.If I am calling to a Phone other than this LAN using proxy at 24.10.65.90, I am able to establish.What may be the problem to the previous. I hope I have explained my problems in a user Understandable way. I am expecting a sooner reply. Thanking you In Advance, Shiva

2 1

[Users] Freeradius dictionary additions..
by nick 05 Sep '06

05 Sep '06

When I add my dictionary.openser to Freradius v 1.0.1-3 on Centos 4.3, and after following all the steps outlined on the webpage: http://www.openser.org/docs/openser-radius-1.0.x.html When I get to testing the configuration, radiusd -X i get this: read_config_files: reading dictionary Errors reading dictionary: dict_init: No ATTRIBUTE "Sip-Method" defined for VALUE "Other" Errors reading radiusd.conf Unfortunately freeradius.org is now a parked domain (seems like a rather large oversight on their part...) so I can't ask them.. Has anyone here ever seen an issue like this? BTW, I'm using openser 1.1.0-notls compiled from source. Not that seems to apply here though.

6 10

[Serusers] DNS SRV Query's
by Ricardo Martinez 05 Sep '06

05 Sep '06

Hello list. I was doing some debugs on my SER machine and i found this these DNS query's: 9.381051 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 9.390312 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 9.390559 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 9.399772 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 11.337343 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _mediaproxy._tcp.sipproxy.mydomain.net 11.339614 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 11.345067 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 11.349127 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 11.349584 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 11.358696 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 17.247771 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 17.253948 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 17.254042 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 17.263298 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 18.988320 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 18.998219 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 18.998329 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 19.008071 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name What modules is generating this querys' and why ? Thanks Ricardo

2 1

[Users] Mediaproxy with t38
by Leonid Fainshtein 05 Sep '06

05 Sep '06

Pavel, I think you have to check if your VoIP gateways can send UDPTL "no-signal" messages. Sometimes the caller gateway doesn't send any messages until it receives CED from callee. Because of that neither MediaProxy nor RTPProxy can't complete the ip/port binding for the media session. Also, check which ports are used for UDPTL session. If they are differ from the ports that were used during the previous voice (RTP) session than you probably have to re-establish mediaproxy session. At least it is true for RTP Proxy that I use. For example, I call unforce_rtp_proxy() and then force_rtp_proxy() on each re-INVITE. To be honest, I have never tried the MediaProxy but I suppose that it works in the similar way like the RTP Proxy. Best regards, Leonid Fainshtein Product Manager IP Gear Ltd., a division of Qualmax "Get your VoIP network "In Gear" with Claro gateways from IP Gear" e-mail: LFAIN(a)ipgear.net Tel: +972-4-9907-525 Fax: +972-4-9990-334 www.ipgear.net

1 0

[Users] Not Hung uping...
by raviprakash sunkara 05 Sep '06

05 Sep '06

Good Morning, In local network openser is working fine...... In Behind the NAt the voice is also fine.... But problem is Not hung. upping.... I think problem in Record router header.... Actually it should reply as " Record Router < sip: public.ip:...... > " , But for me it reply as " Record Router <sip: private.ip.... > " That why i think Bye request is not reaching to openser server and its not hung upping.... and my guess is router(firewall) causing this problem , But voice is fine.... Below one is log file.... sip_domain= public_ip It s not hung upping....... openser server system ip 192.168.2.75 And UAC are outside the NAT The media is fine..... but not hung upping at a same time..... I think the record router is not correct... which is red color marked one...... 59.93.74.129:41010 -> 192.168.2.75:5060 INVITE sip:9001@public_ip SIP/2.0..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@6 1.17.248.68>;tag=587704647..To: <sip:9001@public_ip>..Contact: < sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2. 34..CSeq: 35636 INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 3 5180906 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=00..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a =rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2.34 ..CSeq: 35636 INV ITE..Server: OpenSer (1.1.0-notls (i386/linux))..Content-Length: 0..Warning: 392 192.168.2.75:5060 "Noisy feedback tells: pid=4700 req_src_ip=59.93.74. 129 req_src_port=41010 in_uri=sip:9001@public_ip out_uri= sip:9001@59.93.74.129:5060 via_cnt==1".... # U 192.168.2.75:5060 -> 59.93.74.129:5060 INVITE sip:9001@59.93.74.129:5060 SIP/2.0..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b 487.0..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public _ip>;tag=58770464 7..To: <sip:9001@public_ip>..Contact: <sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..M ax-Forwards: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 35180906 IN IP4 59.93.74.1 29..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtp map:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;br anch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: < sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2.34 ..CSeq: 3 5636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;b ranch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: < sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647> ..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip >;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip: 192.168.2.75 ;lr=on;ftag=587704647 >..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip: 192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>.. Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129 :5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: < sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647> ..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75 ;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: < sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip: 192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>. .Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129 :5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: < sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647> ..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Users] 415 unsopported document format.
by Javier Ramirez 04 Sep '06

04 Sep '06

Hi, I have a problem, when I send a PUBLISH with personal status, if have accent, the openser send me 415 unsopported document format. How to encode for openser accept my PUBLISH ? best regards Javier

2 1

[Users] How to configure multi-domain?
by Ricardo Carvalho 04 Sep '06

04 Sep '06

Besides from adding the following lines in configuration file and adding 2 diferent domains to the domain table, what else should I do to implement multi-domain? modparam("usrloc", "use_domain", 1) modparam("registrar", "use_domain", 1) My phones still register, but none of them can call any other, even if I call someone only by username or by username@domain. Regards, Ricardo.

2 1

[Users] run_failure_handlers warning message
by Federico Giannici 04 Sep '06

04 Sep '06

Since we upgraded to 1.1.0 from time to time the following warning is logged: WARNING:tm:run_failure_handlers: no UAC or CANCEL support (1, 2) Is it "normal" or does it indicate some problem some were? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 3

[Users] redirect and proxy functionality
by Mark Kent 04 Sep '06

04 Sep '06

I have a clash of concepts in my head, and it's messing with my openser config. 1) On one hand I think that if a phone is configured to use an outbound proxy then it should use it. Period. 2) On the other hand, I now do a "300 Redirect" and the phone sends back, to the same box, the same Invite but with the redirected r-uri. This is consistent with 1), but in this case I don't want this behaviour... If things are supposed to work like 2) then why wouldn't I just rewritehostport() and t_relay it without bothering with the redirect? In any case, I don't want to do that (rewritehostport), I want the redirect to be "try over there, and don't bug me again." I don't want to relay the dialog. So, in the real world, what should I hope to get out of a redirect (3xx) code? Thanks, -mark

2 1

[Users] uac_redirect out of memory?
by Greg Fausak 04 Sep '06

04 Sep '06

We've been doing a little bit of stress testing on the auc_redirect module. Throwing 60 calls per second at it. After quite a while I start getting error messages like: Sep 1 15:33:43 www openser[74528]: ERROR: sip_msg_cloner: cannot allocate memory Sep 1 15:33:43 www openser[74499]: redirect.g2.sbsqa2.com-LOOSE: time_t=1157142823 ^LINE=47 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=5723-28564(a)10.88.129.253 ^cseq=3 ^contact=<sip:18001002132@10.88.129.253> ^from=sip:+18001002132@10.88.129.253 ^fromtag=18001002132 ^to=sip:+16661012126@10.88.128.116 ^totag=5724 ^method=BYE ^ruri=sip:+116661 012126@10.88.129.251:5060 ^messageid=263458^remark=Doing it Sep 1 15:33:43 www openser[74528]: ERROR: new_t: out of mem: Sep 1 15:33:43 www openser[74592]: redirect.g2.sbsqa2.com-LOOSE: time_t=1157142823 ^LINE=47 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=5724-28564(a)10.88.129.253 ^cseq=3 ^contact=<sip:18001002133@10.88.129.253> ^from=sip:+18001002133@10.88.129.253 ^fromtag=18001002133 ^to=sip:+16661012127@10.88.128.116 ^totag=5725 ^method=BYE ^ruri=sip:+116661 012127@10.88.129.251:5060 ^messageid=263192^remark=Doing it Sep 1 15:33:43 www openser[74528]: ERROR: t_newtran: new_t failed Sep 1 15:33:43 www openser[74499]: ERROR: sip_msg_cloner: cannot allocate memory Sep 1 15:33:43 www openser[74592]: ERROR: new_t: out of mem: Sep 1 15:33:43 www openser[74499]: ERROR: new_t: out of mem: Sep 1 15:33:43 www openser[74592]: ERROR: t_newtran: new_t failed in the log file. I have seen similar problems with regular t_relay when we approach 200 calls per second. Is this a shared memory problem? How much shared memory to I need to set up? Have a great weekend! ---greg -- Greg Fausak greg(a)thursday.com

3 3

[Users] Not Hung uping
by raviprakash sunkara 04 Sep '06

04 Sep '06

Hi.... please forgive me in egnlish...... Below one is log file.... sip_domain= public_ip It s not hung upping....... openser server system ip 192.168.2.75 And UAC are outside the NAT The media is fine..... but not hung upping at a same time..... I think the record router is not correct... which is red color marked one...... 59.93.74.129:41010 -> 192.168.2.75:5060 INVITE sip:9001@public_ip SIP/2.0..Via: SIP/2.0/UDP 59.93.74.129:5060;rport;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@6 1.17.248.68>;tag=587704647..To: <sip:9001@public_ip>..Contact: < sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2. 34..CSeq: 35636 INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 3 5180906 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=00..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a =rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INV ITE..Server: OpenSer (1.1.0-notls (i386/linux))..Content-Length: 0..Warning: 392 192.168.2.75:5060 "Noisy feedback tells: pid=4700 req_src_ip=59.93.74. 129 req_src_port=41010 in_uri=sip:9001@public_ip out_uri= sip:9001@59.93.74.129:5060 via_cnt==1".... # U 192.168.2.75:5060 -> 59.93.74.129:5060 INVITE sip:9001@59.93.74.129:5060 SIP/2.0..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b 487.0..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=58770464 7..To: <sip:9001@public_ip>..Contact: <sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..M ax-Forwards: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 35180906 IN IP4 59.93.74.1 29..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtp map:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;br anch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: <sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 3 5636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;b ranch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: <sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip >;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647 >..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Users] another permissions documentation issue
by Juha Heinanen 04 Sep '06

04 Sep '06

http://www.openser.org/docs/modules/1.1.x/permissions.html does not include allow_refer_to(), but does include allow_uri(), which is not part of release 1.1.0. -- juha

2 1

[Users] ENUM
by Andrew And 04 Sep '06

04 Sep '06

I have problem with ENUM, below I am showing my DNS configuration (Bind version 9): $ORIGIN 1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 8.8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server2.com!" . When I use DIG "1.8.8.5.2.1.2.3.3.e164.arpa" or " 1.8.8.5.3.1.2.3.3.e164.arpa", I don´t have problem. But When I try "1.8.8.3.2.1.2.3.3.e164.arpa" DIG don´t obtain resolution. I see that I need to add in DNS the tree between 1.2.3.3 and 8.5.2.1.2.3.3: $ORIGIN 2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . I have a lot of servers and this is not scalable, then I can´t use the tree between 1.2.3.3 and 8.5.2.1.2.3.3. Is there another configuration that I don´t need to use the tree between 1.2.3.3 and 8.5.2.1.2.3.3?

2 1

[Users] Record router problem in NAT RTPproxy.
by raviprakash sunkara 04 Sep '06

04 Sep '06

Hi Users , Good Morning, In local network openser is working fine...... In Behind the NAt the voice is also fine.... But problem is Not hung. upping.... I think problem in Record router header.... Actually it should reply as " Record Router < sip: public.ip:...... > " , But for me it reply as " Record Router <sip:private.ip.... > " That why i think Bye request is not reaching to openser server and its not hung upping.... and my guess is router(firewall) causing this problem , But voice is fine.... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Serusers] SER ,asterisk registration
by Siqhamo Sifo 04 Sep '06

04 Sep '06

How do I let clients registered on SER make calls through asterisk and virse versa

2 1

[Users] Mediaproxy with t38
by Pavel D. Kuzin 04 Sep '06

04 Sep '06

Hello! Somebody using mediaproxy with t38? Can you provide me some configuration examples? Thank you. -- Pavel D.Kuzin System Administrator Nodex ISP St. Petersburg, Russia pk(a)nodex.ru http://nodex.ru

5 7

[Serusers] UA, SER, RTPPROXY in local net, some external clients with public ip
by Rustam Yusupov 04 Sep '06

04 Sep '06

Hello, serusers. I am newbie for SIP. I installed ser, rtpproxy and read manual very many times :-), but... Ser server and some clients in local network : ser - 192.168.100.1 UA1 - 192.168.100.2 UA2 - 192.168.100.3 Some clients have public ip address, not behind firewall... All work fine between local clients, but not work with external clients. I think that local clients register with local ip in Contact field from local network and cannot be fix with fix_nated_ function, and so can't access from external... Simple config or config for rtpproxy from gettingstart not help me... Can anybody help me with simple config for this network configuration? -- Rustam mailto:ru@aspec.ru

2 1

[Serusers] Final ACK problem
by Abdul Qadir 04 Sep '06

04 Sep '06

Hi I am trying to setup SER with nathelper module and portaone proxy. All other messages are sent on the nated IP of both user agents, except final ACK. Which is sent on private IP of called party. Here is the sequence .... I have also pasted my ser.cfg file My ser is running on public IP and both user agents are behind firewall. UA1 SER UA2 INVITE ===============> INVITE ===============> TRYING <=============== Ringing <=============== Ringing <=============== OK <=============== OK <=============== ACK ===============> ACK (sent on UA2's Private IP) ===============> debug=7 fork=yes log_stderror=yes listen=some_public_ip port=5060 check_via=no dns=no rev_dns=no server_signature=no sip_warning=no fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/xlog.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/exec.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("nathelper","rtpproxy_sock", "udp:some_public_ip:port") modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 0) modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 30) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # ------------------------- request routing logic ------------------- # main routing logic route { if(!mf_process_maxfwd_header("10")) { log(1, "too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if(msg:len >= max_len ) { log(1, "message too big\n"); sl_send_reply("513", "Message too big"); break; }; if(loose_route()) { t_relay(); break; }; if(uri==myself) { if(method=="REGISTER") { fix_nated_register(); force_rport(); if(!save("location")) { log(1, "* * * * * < < < could not save location > > > * * * * *\n"); sl_reply_error(); }; break; }; if(method=="INVITE") { fix_nated_contact(); record_route(); if(!lookup("location")) { sl_send_reply("404", "subscriber is offline. Pls try later"); break; } force_rport(); force_rtp_proxy("l"); fix_nated_sdp("1"); } }; if(!t_relay()) { log(1, "relay error\n"); sl_reply_error(); break; }; } --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

2 1

Re: [Serusers] Bandwidth parameter modification (b:) from ser.cfg
by Greger V. Teigre 04 Sep '06

04 Sep '06

Ah! Try: http://www.ietf.org/rfc/rfc3556.txt?number=3556 for description of b-parameter in SDP. g-) M. Victoria Peláez wrote: > Hi! > I will re-try with nathelper, mediaproxy and mangle modules ...... But > till now I haven't found anything about b parameter. About bandwidth > types I also don't now what that mean, I have written it down because > I have read it in some articles ... > > My problem is that I have one application that exchange a b parameter > (b=CT 140) during INVITE and isn't able to work with any other SIP > based application. And the only suitable explanation we have found is > that the b parameter is disturbing our communication. > > So thank you very much! We'll try something else! > > Best regards, > Victoria > > Greger V. Teigre escribió: >> Victoria, >> SIP and SER do not really look at the SDP payload (ideal world). >> However, nathelper, mediaproxy and the mangle (have a look at it) >> modules allow you to do tricks with SDP fooling the user agents >> involved in the dialog in some way (for example by making both talk >> to a rtp proxy server). >> As for bandwidth types (not sure what you mean here?), maybe somebody >> on the list knows of something, but it's slightly off topic on this >> list. >> g-) >> >> M. Victoria Peláez wrote: >>> Hi! >>> >>> I am a beginner with SIP/SER systems. I have been having a look into >>> ser.cfg, in order to modify some SDP parameters, for instance, >>> fix_nated_sdp to change connection IP, and I have found enough >>> information to do it. But, now, I am traying to change b parameter, >>> but cannot find where to do it (Which module manage it?). I am also >>> looking for a document with a description of all bandwidth types >>> (CT, AS). Could you tell me something about this stuff, please? >>> >>> Thank you very much in advance for your help. >>> >>> I am very sorry if this is a very silly question! >>> >>> Best regards, >>> Victoria >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> >

1 0

[Serusers] RTP Proxy errors
by sip 04 Sep '06

04 Sep '06

So... I followed the instructions on OnSip on how to set up RTPProxy. However, I made a modification because I'm running it on a different machine... changing the modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpptoxy.sock") to modparam("nathelper", "rtpproxy_sock", "udp:XX.XX.XX.XX:22222") I've run RTPproxy on the remote machine as such: ./rtpproxy -f -l XX.XX.XX.XX -s udp:* 22222 And it's running. I start the ser server, and I see a slew of commands hit the RTPProxy console: received command "26742_0 V" sending reply "26742_0 20040107 " received command "26745_0 V" sending reply "26745_0 20040107 " received command "26791_0 V" sending reply "26791_0 20040107 " received command "26797_0 V" sending reply "26797_0 20040107 " received command "26748_0 V" sending reply "26748_0 20040107 " received command "26750_0 V" sending reply "26750_0 20040107 " received command "26751_0 V" sending reply "26751_0 20040107 " received command "26752_0 V" sending reply "26752_0 20040107 " received command "26784_0 V" sending reply "26784_0 20040107 etc, etc ad infinitum After a moment, the SER console spits back: ug 30 14:07:48 death ser[26742]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:48 death ser[26742]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:48 death ser[26742]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:48 death ser[26745]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:48 death ser[26745]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:48 death ser[26745]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26791]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26791]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:49 death ser[26791]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26797]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26797]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:49 death ser[26797]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26748]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26748]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:49 death ser[26748]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26750]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26751]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26752]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26784]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26789]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26792]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy etc, etc, ad infinitum I've done a search on this on Google and found a couple of people with the same questions and NO people responding with answers. Any ideas what's going on? N.

4 12

Re: [Users] Error when installing openseradmin.Please help..Please
by Ferianto siregar 04 Sep '06

04 Sep '06

Dear Nick, Thank you very much for your kind-hearted to help me. Thank you. Nick, If I can make a wish for some help to you, would you mind help me in these questions: 1. What do you mean by installing mysql? because when I run # rpm -qa|grep mysql, I got the version of mysql are : mysql-3.23.54a-11 mysql-server-3.23.54a-11 php-mysql-4.2.2-17 Do I need install mysql again? If I have to, How? Please 2. Is there any configuration in websever? or I just have to restart the webserver by using "# /etc/rc.d/init.d/httpd restart"? If I have to configure the webserver, Would you mind share the file..Please.. Please Help me Nick..Please...because my skill is too lack. Please... I am so sorry for bothered and waste your time Nick. I am Sorry. Thank you very much Regards, Ferianto nick <nick(a)mobilia.it> wrote: Ferianto siregar wrote: > Dear all, > > Thank you very much to all of you for your kind-hearted to reply my > questions. > Thank you very much. > > All, Now I try to build the openser admin in my openser server. I have > tried to build it for many times, but never successfully. I know that my > skill is too low. So I try to ask it to you. Please help me..Please. > > I have tried to build the openser admin packet by using README .But, I > am confused. Maybe it is because of my lack skill to understand it. > Here is the task that I have done: > 1. I have download Ruby and Rails packet. The versions that I have got > were rubygems-0.9.0.tar.gz and rails-1.1.6.tar.gz > > a. Untar the Rubygems packet > # tar -zxvf rubygems-0.9.0.tar.gz > # cd rubygems > # ruby setup.rb No errors on the install? > b. Untar the Rails packet > # tar -zxvf rails-1.1.6.tar.gz > # cd rails > # ruby script/server If you have gems installed you can just gem install rails --include-dependencies > > c. Untar the openseradmin packet > # tar -xjvf openseradmin-0.3.tar.bz2 > # cd openseradmin-0.3 > That's correct, but without rails installed it probably won't work.. > > d.Logging into the database server > #mysql -u root -p > > Connect openser; > > GRANT ALL ON openser.* TO 'openser'@'localhost' IDENTIFIED > BY 'openserrw'; > > USE openser; > > SOURCE /root/openseradmin-0.3/db/database_tables.sql; > > EXIT > > Here is the confusing problem that I got : > 1. When I try to run "gem install mysql", I got the error message. So, I > can not execute it. What`s wrong? Try gem install mysql --with-mysql-config=/path/to/mysql/lib > 2. When I try to access it by using web browser by typing > "http://localhost:3000", I get nothing. What`s wrong? You need to start the web server, but without mysql, it won't work.. > 3. Would you mind tell me if all the task that I have done correct or > not? I need some suggestion or adviseable. Please.... Install rails, then install mysql, then start the server, it should work fine... > > > I do hope anybody can help me. Please... > > > Regards, > > > > Ferianto > > ------------------------------------------------------------------------ > Yahoo! Messenger with Voice. Make PC-to-Phone Calls > > to the US (and 30+ countries) for 2¢/min or less. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out. --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

2 1

[Serusers] Help for sn2400 call
by pokerino76 04 Sep '06

04 Sep '06

The mediagateway it is installed in a webfarm of BT (British Telecom) , while the sn2400 it is a sipgateway. I do not succeed to carry out calls of the net outside, while I succeed to receive them. What I must make? thanks. ___________________________________________________________________ Prestiti da 15 istituti. Tan dal 5,90% per richieste entro il 31 Agosto. Servizio gratuito. http://click.libero.it/mutuionline1

1 0

[Serusers] Setting Accounting Flags:-(
by ravi reddy 04 Sep '06

04 Sep '06

Hi users , I am using ser.cfg from onsip.org and because i am not a programmer (means i dont understand the language)I am using the config file as i downloaded , before i gave a lot messages regarding this getting of multiple records in radius database as well as in SER mysql(ac) database also ; i am attaching the ser.cfg below please let me know where to set flag so that i can get single record per call ;-) thank you. debug=4 fork=yes log_stderror=yes listen=My Ip address # INSERT YOUR IP ADDRESS HERE port=5060 children=5 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=766 fifo_db_url="mysql://ser:heslo@localhost/ser" sip_warning=yes server_signature=no loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("auth_db|permissions|domain|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("xlog", "buf_size", 8192) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "radius_flag", 11 ) modparam("acc", "radius_missed_flag",12) modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "db_flag",13) modparam("acc", "db_table_acc", "acc") 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("81.21.33.35:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- setflag(11); if (loose_route() ){ ------------------------------------------------------>>>>>>>>>>here i set flag for radius accounting ; # mark routing logic in request #append_hf("P-hint: rr-enforced\r\n"); t_relay(); route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP 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()) { xlog("L_WARN", "[%Tf]: possible uri spoofing attempt!; method=%rm; IP=%is; from=%fu; to=%tu\n"); sl_send_reply("403", "Forbidden"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ---------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (uri=~"^sip:1[0-9]{8}@") { strip(1); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (uri=~"^sip:00[0-9]*@") { route(4); route(5); break; }; if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; if (avp_db_load("$ruri/username", "s:fwdbusy")) { if (!avp_check("s:fwdbusy", "eq/$ruri/i")) { setflag(26); }; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { setflag(27); }; }; t_on_failure("1"); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("pstn gateway ip"); avp_write("i:30", "inv_timeout"); t_on_failure("1"); route(4); route(1); } route[6] { # ---------------------------------------------------------- # Call Forwarding Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ---------------------------------------------------------- if (uri=~"^sip:1[0-9]{8}@") { strip(1); }; lookup("aliases"); if (uri!=myself) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); break; }; if (uri=~"^sip:00[0-9]*@") { route(4); route(5); break; } if (uri=~"^sip:+[0-9]{3}[0-9]*@") { route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { #local pstn route(4); route(1); break; }; sl_send_reply("404", "User Not Found"); }; route(4); 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(); }; } .........................)

1 0

[Users] Error when installing openseradmin.Please help..Please
by Ferianto siregar 04 Sep '06

04 Sep '06

Dear all, Thank you very much to all of you for your kind-hearted to reply my questions. Thank you very much. All, Now I try to build the openser admin in my openser server. I have tried to build it for many times, but never successfully. I know that my skill is too low. So I try to ask it to you. Please help me..Please. I have tried to build the openser admin packet by using README .But, I am confused. Maybe it is because of my lack skill to understand it. Here is the task that I have done: 1. I have download Ruby and Rails packet. The versions that I have got were rubygems-0.9.0.tar.gz and rails-1.1.6.tar.gz a. Untar the Rubygems packet # tar -zxvf rubygems-0.9.0.tar.gz # cd rubygems # ruby setup.rb b. Untar the Rails packet # tar -zxvf rails-1.1.6.tar.gz # cd rails # ruby script/server c. Untar the openseradmin packet # tar -xjvf openseradmin-0.3.tar.bz2 # cd openseradmin-0.3 d.Logging into the database server #mysql -u root -p > Connect openser; > GRANT ALL ON openser.* TO 'openser'@'localhost' IDENTIFIED BY 'openserrw'; > USE openser; > SOURCE /root/openseradmin-0.3/db/database_tables.sql; > EXIT Here is the confusing problem that I got : 1. When I try to run "gem install mysql", I got the error message. So, I can not execute it. What`s wrong? 2. When I try to access it by using web browser by typing "http://localhost:3000", I get nothing. What`s wrong? 3. Would you mind tell me if all the task that I have done correct or not? I need some suggestion or adviseable. Please.... I do hope anybody can help me. Please... Regards, Ferianto --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.

2 1

[Serusers] Proxy and Presence Agent support
by Henry Duan 04 Sep '06

04 Sep '06

hi, I'm new to SER. I need to use SER as a proxy and a PA as well. I'm wondering if I could just download the package unstable version of SER with presence<http://ftp.iptel.org/pub/ser/presence/> from iptel.org website. BTW, I'm using Linux Fedora 5, not debian. I'm wondering if there is any problem for this. Thanks a lot! henry

3 2

[Users] Problem with ENUM
by Andrew And 03 Sep '06

03 Sep '06

Below I m showing my DNS configuration (Bind version 9): $ORIGIN 1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 8.8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server2.com!" . When I use DIG "1.8.8.5.2.1.2.3.3.e164.arpa" or " 1.8.8.5.3.1.2.3.3.e164.arpa", I don´t have problem. But When I try "1.8.8.3.2.1.2.3.3.e164.arpa" DIG don´t obtain resolution. I see that I need to add in DNS the tree between 1.2.3.3 and 8.5.2.1.2.3.3: $ORIGIN 2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@ server1.com!" . $ORIGIN 8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . I have a lot of servers and this is not scalable, then I can´t use the tree between 1.2.3.3 and 8.5.2.1.2.3.3. Is there another configuration that I don´t need to use the tree between 1.2.3.3 and 8.5.2.1.2.3.3?

1 0

[Serusers] SER+Asterisk
by Siqhamo Sifo 03 Sep '06

03 Sep '06

Where can I find docs on asterisk and ser integration.

1 0

[Users] Need help for openser admin installation
by Ferianto siregar 03 Sep '06

03 Sep '06

Dear all, Thanks to every body in this forum. Thank you very much I tried to build openser admin in my openser server. But, I am confused when i try to read the installation note. I do hope anybody can help me to understand it. These are my questions about it: 1. As I read, the openser admin need ruby packet. For installation of ruby, after we untar the packet, does we just type "gem mysql install" command?Because as i read from the installation openser admin, there are only this command. 2. When I try to build the openser admin, i got the error message. Here is the message : ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'VIEW usr_preferences_view AS SELECT id, username, domain, attribute, type AS att' at line 1 As I read from the message in this mailinglist before, this is because of the mysql version. I use mysql 3. Do i need use mysql 5? Where I can get the suitable varsion? That all my question. I do hope any body can help me. Thank you very much. Regards, Ferianto --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

1 0

[Serusers] Help for sn2400 call
by pokerino76 02 Sep '06

02 Sep '06

I have shaped the sn2400 for the ser, succeed to make telephone calls of the net inside ,the calls phone inside working, but I do not succeed to calls phone from the outside, which it must be the configuration of ser.cfg or the other? . Thaks for help.

2 1

[Serusers] SER+Asterisk
by Siqhamo Sifo 02 Sep '06

02 Sep '06

I have ser sitting on my iptables nat box and my asterisk box on the lan . Ser does forwarding so that any requests (register,invite,ack,...) to the nat box at 5060 r sent to my asterisk box on the lan .I can register from outside to my asterisk box but there is only one way audio , reason being that when the asterisk box send a sip packet whith session description the sdp part of the sip packet is not natted . I have tried : fix_nated_contact(); fix_nated_sdp("3");

3 2

[Serusers] Presence modules and LCS
by harrygaillac-sip＠yahoo.fr 02 Sep '06

02 Sep '06

Hello, Does ser presence modules are able to work with LCS micosoft ? http://www.microsoft.com/office/livecomm/prodinfo/compare.mspx#ECD anybody is experienced with LCS to answer ? Regards Harry p4.vert.ukl.yahoo.com uncompressed/chunked Fri Sep 1 12:13:36 GMT 2006 ___________________________________________________________________________ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com

1 0

[Users] RTPProxy non working?
by Federico Giannici 02 Sep '06

02 Sep '06

I'm trying to install rtpproxy. It usually works, but there are some cases that don't work. For example, when a Cisco PSTN gateway is called, RTP traffic is proxied in one direction only. I think the problem is in the handling of the reply received from the gateway: it have two "c" lines but only one is modified. Here it is the body of the original reply: v=0. o=CiscoSystemsSIP-GW-UserAgent 2199 2489 IN IP4 83.211.2.216. s=SIP Call. c=IN IP4 83.211.2.217. t=0 0. m=audio 18850 RTP/AVP 8 13 101. c=IN IP4 83.211.2.217. a=rtpmap:8 PCMA/8000. a=rtpmap:13 CN/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=direction:passive. And here it is the body modified by rtpproxy: v=0. o=CiscoSystemsSIP-GW-UserAgent 2199 2489 IN IP4 83.211.2.216. s=SIP Call. c=IN IP4 83.211.2.217. t=0 0. m=audio 35086 RTP/AVP 8 13 101. c=IN IP4 195.120.250.49. a=rtpmap:8 PCMA/8000. a=rtpmap:13 CN/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=direction:passive. a=nortpproxy:yes. Only the secondo "c" line contain the IP of our rtpproxy. I don't know RTP enough to understand if the problem is in rtpproxy or in the gateway. Thanks for any info. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 4

[Users] openser admin question..
by nick 01 Sep '06

01 Sep '06

Is an error like this normal when doing the db install? ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'VIEW usr_preferences_view AS SELECT id, username, domain, attribute, type AS att' at line 1 By the way, while ruby on rails might be a great development platform, it's a right pain in the behind to install.

2 5

[Users] codecs problem...
by raviprakash sunkara 01 Sep '06

01 Sep '06

Hi Users...... A doubt For me in Codec.... I'm Using openSER and RTPProxy openser didn't take the Media tranmmision , it transmmit to RTPproxy or mediaproxy. Media(voice) taking place in RTP proxy... UA1 using the G729 codec -------> openser server -------------------> UA2 using G711u for above case it will not generate the porper handshanking............ But RTP Proxy may has transmiits the media , because it has codecs in ( media proxy or rtpproxy) For a realtime producttion, UAs may having the differnet codecs. plz resolve my doubt..... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Users] Integration of Openser with CDRTool: Installation errors
by Alberto Pacifico 01 Sep '06

01 Sep '06

Hi all, I'm trying to install CDRTool (http://www.ag-projects.com/CDRTool.html) and some errors happened. When I try to access the CDRTool webpage, I'm able log in the page, but when I try to enter in the "call search page" some warnings appears: "Warning: main(SOAP/Client.php): failed to open stream: Arquivo ou diretÃ³rio nÃ£o encontrado in /var/www/CDRTool/rating_lib.phtml on line 4" "Fatal error: main(): Failed opening required 'SOAP/Client.php' (include_path='.:/opt/lampp/lib/php') in /var/www/CDRTool/rating_lib.phtml on line 4" In the file "rating_lib.phtml", the line 4 is: "require_once("SOAP/Client.php");" The file Client.php does not exist... But a file named "client.php" exists..and into it, theres an instruction calling "Client.php": "include("SOAP/Client.php");" An additional information is that I'm using the XAMPP Server Apache. Please, Can you help me? Thanks in advance! Alberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] openser+asterisk+voicemail password
by ram 01 Sep '06

01 Sep '06

Hi iam integrating with asterisk voice mail with Openser when we are creating openser views with asterisk the secret going to be Binary(0);, how can i change to numeric ? why i need is, take example If the user Phone logged in and the user have person voicemessage box he only can access his voice messages, when he dial secret If not any one can come and access voice message by dialing access number. with the config seen in the examples, person can dial *98 and hear all the messages and delete too. so for the security reason, how can i change the secret, iam trying to alter the table, but its say;s in view mode. any suggestions. Ra,

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[Users] Question about voip-info.org's OpenSER/RTPProxy configuration example
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I was exploring the example configuration file in the following URL: http://voip-info.org/wiki/view/OpenSER+And+RTPProxy The following block bewilders me: % failure_route[1] { % if (isflagset(6) || isflagset(5)) { % unforce_rtp_proxy(); % } % } How a flag happens to be turned on there ? 6 is the registrar's nat_flag but no lookup() is performed. Furthermore, the tm module doesn't even depend on the registrar module. Flag 5 is set in route[2] which is obviously not called from failure_route[1]. Is there some black magic around here ? Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Users] RTPProxy failover... load-balancing ?
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I was wondering whether I could use multiple rtpproxy daemons along one OpenSER instance, in order to both provide a failover mechanism as well as load-balacing. Google immediately pointed me out the following thread: http://www.openser.org/pipermail/users/2005-October/001071.html It happens that openser/modules/nathelper/nathelper.c contains the following comment: % 2005-02-24 Added support for using more than one rtp proxy, in which % case traffic will be distributed evenly among them. In addition, % each such proxy can be assigned a weight, which will specify % which share of the traffic should be placed to this particular % proxy. % % Introduce failover mechanism, so that if SER detects that one % of many proxies is no longer available it temporarily decreases % its weight to 0, so that no traffic will be assigned to it. % Such "disabled" proxies are periodically checked to see if they % are back to normal in which case respective weight is restored % resulting in traffic being sent to that proxy again. % % Those features can be enabled by specifying more than one "URI" % in the rtpproxy_sock parameter, optionally followed by the weight, % which if absent is assumed to be 1, for example: % % rtpproxy_sock="unix:/foo/bar=4 udp:1.2.3.4:3456=3 udp:5.6.7.8:5432=1" At first glance, I would say the aforementioned comment states that nathelper achieves both failover and weighted load-balancing. Can anyone confirm this ? Ultimately, I would like to do use the following arch : ________+----------+ /________| RTPProxy | +---------+ // +----------+ | OpenSER |_____// +---------+ /\_________+----------+ / \ ______| RTPProxy | +---------+ / \/ +----------+ | OpenSER |__/_____/\ +---------+ \ \_____+----------+ \____________| RTPProxy | +----------+ If I understand correctly how it works, this should be possible to use N OpenSER instances along with M RTProxy instances. Am I right ? Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Users] Simple configuration
by Vladimir Orehov 01 Sep '06

01 Sep '06

Hello, I'm new to this server, and I ask for help! I don't undertand for now how to configure openSER. Can someone send me simple configuration file "openser.cfg" wich included: -- accounting loggoing to a specific file ( like "/home/agb/acc.log") -- authentication from simple text file (if possible), or using SQL -- routing numbers started with 00 to specific gateway with ip (123.123.123.123) Another question is: how to enable debugs of this server? Please reply ASAP and best regards, Vladimir Orehov SIA Velkoms Support engineer

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[Users] Question about voip-info.org's OpenSER/RTPProxy configuration example
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I was exploring the example configuration file in the following URL: http://voip-info.org/wiki/view/OpenSER+And+RTPProxy The following block bewilders me: % failure_route[1] { % if (isflagset(6) || isflagset(5)) { % unforce_rtp_proxy(); % } % } How a flag happens to be turned on there ? 6 is the registrar's nat_flag but no lookup() is performed. Furthermore, the tm module doesn't even depend on the registrar module. Flag 5 is set in route[2] which is obviously not called from failure_route[1]. Is there some black magic around here ? Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Users] Error in running serweb. Please help me..Please
by Ferianto siregar 01 Sep '06

01 Sep '06

Dear all, First of all, I would like to say thanks to all of you for your help and suggestion to solve my problem untill now. Thank you very much. Now, I try to install serweb_2004-07-27 in my openser server. I have read the installation note that I got from the serweb packet carefully. So, I can build the serweb in my openser server successfully. It means that it can display a login screen in the browser. But, I have a problem when I try to login by using username "admin" with password "openserrw". (This is based on the installation note that I have read before) I got error message. This is the error message : Database error: Invalid SQL: select username, perms from subscriber where phplib_id = '721153ca8942ea1a08900f5b61b69a6c' MySQL Error: 1054 (Unknown column 'perms' in 'field list') Session halted. As I try to understand from the error message, it says that there is no "perms" column. So I try to check it in the subscriber table in mysql.But, I did not find "perms" column in the subscriber table. I just see these columns in the subscriber table: -phplib_id -last_name - sendnotification - rpid -username - phone - greeting -domn -domain - email_address - ha1 -uuid -password - datetime_created - hdb1 -first_name - datetime_modified - allow_find -confirmation - flag - timezone What should i do to solve this problem? What should I do if there is no "perms" column in the subscriber table? I do hope anybody can give me a suggestion because this is my first time to build the serweb and mysql. Please help me....Please... Note: I user openser1.1.0-tls and serweb_2004-07-27 As I see from the error message that the phplib_id is exactly matched with phplib_id where username "admin" with password "openserrw" (from the subscriber table). Thank you very much Regards, Ferianto --------------------------------- Get your email and more, right on the new Yahoo.com

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[Users] Migrate from SER to open SER
by ram 01 Sep '06

01 Sep '06

Hi all I have VoIP CDrouter running But i have seen lot of limitation How can i migrate from SER to Open Ser with out lossing any data of users information and CDR Help will be appriciated Ram

2 1

[Users] Incorrect Route header
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I am running a NATing proxy. Everything works fine, except that some of my neighbour proxies send me BYE requests with an incorrect Route header containing their own IP address. The result is naturally that my proxy sends back the request and the call is never ended. I know this is a wrong behaviour and I have contacted the remote administrator to have him fix his proxy. But in the meantime I would like to find a workaround for my users. I tried to do the following, before calling loose_routing(). I am not sure if it's relevant or not nevertheless: % if (method == "BYE") { % # XXX Remove buggy Route header from BYE packets. % xlog("L_ALERT", "Removing buggy Route header from BYE request"); % remove_hf("Route"); % lookup("location"); % } However, I raised the verbosity level and found that the Route header is read as soon as the packet is received, therefore the above block is not really useful. Thank you for your ideas. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Users] VIA Parameter?
by Ricardo Martinez 01 Sep '06

01 Sep '06

Hello list. Sorry to ask this question here, but i can't find the answer in the RFC3261 Is this "VIA" permited according to the RFC? Via: SIP/2.0/UDP 69.20.6.205:15060, SIP/2.0/UDP 69.1.229.69:4081;branch=z9hG4bK-58e9ccfcef0f4c99915b8007cb9713e1-1156793814 It seems to be two VIA's in ONE line... Thanks Regards, Ricardo.-

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[Users] SIP Redirection in Openser
by Padmaja RV 01 Sep '06

01 Sep '06

Hi all, I have tried implementing SIP redirection as mentioned in the example script below: route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); break; }; # rewrite current URI, which is always part of destination ser rewriteuri("sip:parallel@iptel.org:9"); # append one more URI to the destination ser append_branch("sip:redirect@iptel.org:9"); # redirect now sl_send_reply("300", "Redirect"); } But that does redirection only for that account as the account is included in the code. I need the code to be able to redirect any call. Can you tell me how I can generalize the code so that redirection works for any account I choose to redirect? Thanks in advance, Padmaja

2 1

[Users] Authentication in Trunk-Mode
by Martin Klisch 01 Sep '06

01 Sep '06

Hey, are there working ways for authentication with sip trunks? i've read cisco gateways (PGW) and cisco call manager can do this. anyone knows if openser+asterisk support it too?

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[Users] About 'location' table
by Michel Zenone 01 Sep '06

01 Sep '06

Hi! I have registered, thanks to openserctl, an user :'sip:testeur@192.168.0.102'. But when I tyr to reach him, the transaction isn't considered as local processed one. In location table there is: +username + domain + contact + |testeur | | sip:testeur@192.168.0.120:5080 | +---------+--------+--------------------------------+ -192.168.0.120 is the IP address of the user's phone. -192.168.0.102 is the openser IP address. The openser.cfg is as the initial openser.cfg. Thanks, Michel

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[Users] OpenSER AstLinux build posted for testing
by Kristian Kielhofner 01 Sep '06

01 Sep '06

Hello OpenSer Users, It seems that many of you are also Asterisk users or at least subscribe to asterisk-users. Some of you might recognize AstLinux... Just in case you don't: http://www.astlinux.org ---and--- http://www.sf.net/projects/astlinux [in transition] Anyways, I have long hoped to build a version of AstLinux that includes OpenSer. Basically, what I have done is taken astlinux-trunk, add openser, add rtpproxy, de-select the Asterisk packages, and build! What I got was 11mb (gzipped) disk images ready to be written to CF cards/hard disks. As usual, I have created two images: *net4801.img.gz is for the Soekris net4801 SBC *i586.img.gz is for just about any i586 and higher machine There is also an ISO image ready to boot on or install to most i586 or higher machines. You guessed it, it is the one with .iso in the filename. It includes both the net4801 and i586 images. These are 48mb gzipped disk images, with GRUB bootloader and ext2 root filesystem. If you want to write them to disk, try this: gunzip [image.gz] dd if=[output file.img] of=/dev/sdb # <--- VERIFY THIS DEVICE Where /dev/sdb is where your USB CF flash adapter is. If you want access to the ext2 filesystem, try this: gunzip [image.gz] mkdir /mnt/image mount -t ext2 -o loop,offset=32256 [image.gz] /mnt/image If you are curious about how these images were created, take a look at SVN on the astlinux project page. The OpenSer version used for these builds was openser-1.1.0-notls. I built every module that did not depend on an external library that I don't have yet (sorry, no mysql, jabber, or radius yet). I need help!!! For the time being, I don't have mysql. Also, because AstLinux is targeted for compact flash, mysql isn't a good idea anyways. I was not able to get a working configuration in the time alloted using DBTEXT. Does anyone have a sample config that I can put in this build? As of now, openser, rtpproxy, openser modules, and all of the astlinux-style init scripts are done. All I need is a good sample config file to make it "plug and play". I also need testing! My current to-do list: - add mysql (and the openser modules, of course) - add radius support (freeradius?) - jabber/xmpp (depending on size, complexity, etc) - fix astshape (iproute2-based traffic shaper) RTP TOS values to be the same that rtpproxy uses, and standardize this across all astlinux builds - COME UP WITH A NAME. I have no idea what to call this thing. Any suggestions? SERlinux (or something like it) is too obvious and boring! You can grab the goodies here: http://www.krisk.org/openser/ Regardless, I hope that someone finds this interesting or useful. Let me know what you think! -- Kristian Kielhofner

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[Users] icmp port unreachable
by Helge Waastad 01 Sep '06

01 Sep '06

Hi, (just a little bit on the side-bar) I've had some discussion with the developer of twinkle (which is, BTW, a very nice linux softphone) regarding handeling icmp port unreachable and switching to a lower priority SIP router (by DNS SRV). It seems like the dist I'm using (Dapper, kernel 2.6.15-26) will not easily allow the applications to receive the icmp on an unconnected udp socket, hence failover will not work. However on an 2.6.11.4-21.11 kernel it works. Does anyone know if there are any kernel opts which should be added for having this kind of support? Just my thoughts this morning......

2 1

[Users] Openser + Asterisk for P2P calls
by Lucio Maciel 01 Sep '06

01 Sep '06

Hi, Hi, im using an Openser integrated with Asterisk to route PSTN calls, this is working great, but i also want to route P2P calls to Asterisk like this UA1 -> Openser -> Asterisk (calls DIAL(SIP/xxxxx@openser) -> Openser -> UA2, to use its mediaproxy, and account all calls. When UA1 calls UA2, everything goes fine, UA2 receive the INVITE and sends the OK to UA1, the only thing is that UA2 waits for an ACK from the OK that never comes. Have someone tried to do this? How can i configure openser and asterisk for this? Thanks Lucio Maciel

2 1

[Users] IP authentication
by Brian Fertig 01 Sep '06

01 Sep '06

greetings, I have digest and basic auth setup. I am trying to figure out how to setup IP auth. I have read a lot of documentation but I can't seem to find anything to help me understand how to accomplish this. Can someone point me in a direction to some information on how to set this up? Brian

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[Users] 1.1.x/permissions.html is not up to date
by Juha Heinanen 01 Sep '06

01 Sep '06

for some reason web page http://www.openser.org/docs/modules/1.1.x/permissions.html is not up to date. for example, 1.3.12. tag_col (string) 1.3.13. peer_tag_avp (string) are missing. they do exist in rel_1_1_0 source. -- juha

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[Serusers] extra things confusing auth_radius?
by Natambu Obleton 01 Sep '06

01 Sep '06

Ok. I have two clients setup. One is X-Lite and the other is a polycom 601. I am doing radius authentication, but for some reason it doesn't work with the polycom 601 phone. In the debug the only thing that is different about them is something in the uri. Aug 30 09:16:51 laplata ser[25968]: DEBUG: get_hdr_field: <To> [72]; uri=[sip:311@voip.fasttrackcomm.net] Works, but.. Aug 30 09:17:53 laplata ser[25970]: DEBUG: get_hdr_field: <To> [45]; uri=[sip:311@voip.fasttrackcomm.net;user=phone] <mailto:%5bsip:311@voip.fasttrackcomm.net;user=phone%5d> Doesn't. Like it doesn't even send an authentication request to my radius server. I think it is the ';user=phone' part that is confusing it? So what I want to do is remove that part from the URI as a test. I just added a subst to remove it, but still isn't working with radius. It acts like the realm is no longer local. Here is my config file: http://www.natambu.com/serconf.html Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

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[Serusers] SIP Contact header field issue incase of NAT
by Abdul Qadir 01 Sep '06

01 Sep '06

Hi All, I have problem connecting Nokia Phone with SER. The problem is When SER receives the registration request from the Nokia phone. The registration request contain the phone's private IP (192.xxx.xx.xx), since we use wifi for connection. Our proxy detects that since this is a private IP handles the NAT translation and determines the phone public IP address (e.g., 202.xxx.xx.xx, port xx). Our SIP proxy informs the phone that registration is successful but the contact information is changed from userid(a)192.xxx.xxx.xx to userid(a)202.xxx.xxx.xx, port xx. On recieving this changed contact info the Phone rejects 200 Ok. We tried to contact nokia people and they said that its part of RFC3261 to reject such contacts. After this I have also gone through RFC 3261 section "10.2.4" and it seems that phone is doing rite. Can any body tell me why SER is changing the contact address in case of NAT. It would be higly appreciated if you can give me some reference from some RFC or other document which I can show to nokia people. Best Regards, Abdul Qadir --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

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[Serusers] Bandwidth parameter modification (b:) from ser.cfg
by "M. Victoria Peláez" 01 Sep '06

01 Sep '06

Hi! I am a beginner with SIP/SER systems. I have been having a look into ser.cfg, in order to modify some SDP parameters, for instance, fix_nated_sdp to change connection IP, and I have found enough information to do it. But, now, I am traying to change b parameter, but cannot find where to do it (Which module manage it?). I am also looking for a document with a description of all bandwidth types (CT, AS). Could you tell me something about this stuff, please? Thank you very much in advance for your help. I am very sorry if this is a very silly question! Best regards, Victoria

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[Serusers] mediaproxy
by Javier Ramirez 01 Sep '06

01 Sep '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

2 1

[Users] openser routing lookup question
by Dave Logan 01 Sep '06

01 Sep '06

Hey all, I have what may or may not be a simple situation. I've read through dozens of forum entries, dozens more list mails, and a handful of "tutorials" for openser, and I'm still missing something I think. I'm not afraid of the documentation, but neither the "core cookbook" nor the modules documentation had enough detail that I could catch on to what's going on. Let me lay out my scenario, and maybe somebody can point to a particular document or solution I can follow up on. I have multiple asterisk boxes: four as media gateways with TDM trunks, one feature server (VM), and three registrars. So needless to say, I have a lot of dialplan in there just for calls that stay in my network. What I'd like is to introduce a single openser box in there, and for any call that any of those asterisk machines gets, it sends it to openser, which then just calls a simple script (which I've already more or less written) to know what to do. So for example, I own the 555-1000(-1099) DID block DID 555-1000 is registered to machine1 DID 555-1001 is registered to machine2 In openser, I'm envisioning a call like such (php style notation): my $number = 5551000 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551000@machine1 or my $number = 5551001 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551001@machine2 Then I can just route it in the normal openser way. So there, my lookupdestination script knows all the numbers in my central database and where they're located, and on an unknown number assumes it's outside our network so just sends it to the gateway as is (5552000@gw1) or so. Thanks for any pointers at all, dave -- Dave Logan http://www.digitalcoven.com/ "Do let's pretend that I'm a hungry hyena, and you're a bone!" - Alice

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[Users] Help in dailplan in asterisk
by raviprakash sunkara 01 Sep '06

01 Sep '06

Hi Users, I'm new to Asterisk, and I'm working with openSER , For Call Routing I'm using the OpenSER and for PBX, Voicemail and Conferencing I"m using the Asterisk. Now i'm planing for Voicemail, openser listen on 192.168.2.75:5060 Asterisk listen on 192.168.2.76:5060. in Extension.conf [from-sip[ exten => 9001,1,Ringing exten => 9001,2,Voicemail(9001) Just I wnat to leave the voicemail in voice mail box, Can anybody help me......... below message What it is ?................ Executing Ringing("SIP/9001-0984c1e8", "") in new stack -- Executing VoiceMail("SIP/9001-0984c1e8", "9001") in new stack Aug 31 19:31:10 ERROR[7869]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. Aug 31 19:31:10 WARNING[7869]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '9001' Aug 31 19:31:10 WARNING[7869]: pbx.c:1700 pbx_extension_helper: No application 'Hungup' for extension (from-sip, 9001, 3) == Spawn extension (from-sip, 9001, 3) exited non-zero on 'SIP/9001-0984c1e8' -- Incoming call: Got SIP response 479 "Regretfully, we were not able to process the URI (479/SL)" back from 192.168.2.75 -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

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[Users] avp_pushto() changed???
by Greg Fausak 31 Aug '06

31 Aug '06

This doesn't work for me any more: avp_pushto("$P-Asserted-Identity", "$avp(s:tmp_uri)"); I get this: Aug 31 13:42:00 www openser[21792]: ERROR:avpops:fixup_pushto_avp: unsupported destination "$P-Asserted-Identity"; expected $ru,$du,$br Has anything changed? -g -- Greg Fausak greg(a)thursday.com

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[Users] mediaproxy
by Javier Ramirez 31 Aug '06

31 Aug '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

1 0

[Users] mediaproxy
by Javier Ramirez 31 Aug '06

31 Aug '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

1 0

[Serusers] mediaproxy
by Javier Ramirez 31 Aug '06

31 Aug '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

1 0

[Users] OpenSER Summit
by Bogdan-Andrei Iancu 31 Aug '06

31 Aug '06

Hello everybody, It's more than one year since the OpenSER project started. A large community of users and developers has been formed around OpenSER. It's a very active community and one of the most important goals of the OpenSER project is to keep the users community as synchronized as possible with the developers and management groups. During past time we met a lot of people that were happy to see us or we were happy to see them. So, we came to the idea of organizing a OpenSER Summit where OpenSER fans (all types of fans) should meet to share their experience, to discuss about future ways of developing the project or just to personally meet people from the mailing lists. The idea is to have people talking about how they are using OpenSER in different contexts and which were/might be their needs, development people talking about possible solutions and about their ideas of enhancing OpenSER, etc... We were thinking of taking advantage of the upcoming VoN in Berlin in the beginning of November and to organize the Summit within the same time frame - it will be more convenient for people to travel for two events at the same time. We would like to have some feedback from the community - how many people are interested in such a meeting, what will be their primary interests. Any other OpenSER-related ideas are most welcome! Best regards, Bogdan

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[Serusers] pre_auth(): Credentials with given realm not found
by ram 31 Aug '06

31 Aug '06

Hi all iam testing Voicemessages with Asterisk in the log i get frequently pre_auth(): Invalid nonce value received, very suspicious ! when i debug, i get the following message i have posted my config couple of day back 5(2213) get_hdr_field: cseq <CSeq>: <6> <REGISTER> 5(2213) DEBUG: is_maxfwd_present: value = 70 5(2213) parse_headers: flags=256 5(2213) DEBUG: get_hdr_body : content_length=0 5(2213) found end of header 5(2213) find_first_route(): No Route headers found 5(2213) loose_route(): There is no Route HF 5(2213) check_self - checking if host==us: 14==13 && [mydomain.com] == [myipaddress] 5(2213) check_self - checking if port 5060 matches port 5060 5(2213) parse_headers: flags=-1 5(2213) parse_headers: flags=-1 5(2213) check_via_address(some ip address, some ip address, 0) 5(2213) check_self - checking if host==us: 14==13 && [mydomain.com] == [myipaddress] 5(2213) check_self - checking if port 5060 matches port 5060 5(2213) parse_headers: flags=4096 5(2213) pre_auth(): Credentials with given realm not found 5(2213) build_auth_hf(): 'WWW-Authenticate: Digest realm="mydomain.com", nonce="44f6ee623e15edb81a22dcec25ebe8c58b6e056d" '

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RE: [Users] openser admin question..
by peter.3.edwards＠bt.com 31 Aug '06

31 Aug '06

> mysql-4.1.20-1 > > There's no 5 built for centos yet, but this should be fairly > up to date... Well, there's always the option to compile from source ... which isn't quite as painful as it sounds. > The program works.. but I'm not sure what that error might > cause in way > of bugs, or whatever. me either ... sorry! > I had no end of problems installing mysql-ruby in the end, I ended up > running the extern.rb with some command line options > --with-mysql-config > and that got the makefile built. Yeah, same here ... it was quite tedious trying to get it to compile correctly. Ended up having to reinstall MySQL to get the libmysqlclient.so library which isn't bundled with the Solaris package. *sigh* > BTW, what version of mysql-ruby should I be running, I'm > using 2.5.1 at > the moment, 2.6 and 2.7 were also available I went for 2.7 - again, the latest version available - when I just built my Solaris 9 OpenSER Administrator installation. HTH, Peter. -- Peter Edwards tel +44 (0) 1473 606492 This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

1 0

[Serusers] LCR module documentation
by Graham Lewis 31 Aug '06

31 Aug '06

2 1

[Serusers] caller=callee, Not giving me Busy
by ram 31 Aug '06

31 Aug '06

Hi iam using old SER 0.814, since its deployed already in my Office we are in process of migrationg to openser soon mean time i need to fix some problems with existing server only that is the reason iam trying to modify the code and achive the following things 1. Voice mail 2. Caller =callee 3. if the user not in voicemail group ( get busy), if the user off line, Not found I have modified some code after reading all news group here iam attaching the config file which iam trying iam also getting the some errors in Log Aug 30 17:58:20 router ser[609]: ERROR: sl_reply_error used: I'm terribly sorry, server error occured (1/SL) Aug 30 17:58:27 router ser[604]: ERROR: sl_reply_error used: I'm terribly sorry, server error occured (1/SL) Aug 30 18:00:05 router ser[1564]: ERROR: tcp_init: bind(6, 0x80bd7c0, 16) on 127.0.0.1: Address already in use Aug 30 18:00:06 router ser[1571]: ERROR: tcp_init: bind(6, 0x80bd7c0, 16) on 127.0.0.1: Address already in use iam running a cron job to check the services #!/bin/sh mysqlcheck --password=cdrouter --auto-repair --all-databases sleep 4 ser sleep 4 chmod 777 /tmp/ser_fifo help will be appriciated Ram

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[Serusers] How to Integrate SER Server with Jabber IM.
by srinivas sriram 31 Aug '06

31 Aug '06

Hello, I am look to send Missed Call Alert Instant Message(IM) to Jabber Client from SER Server. 1) I have Installed and Configured SER on IP(xyz.xyz.xyz.xx), and Added two(2) user in mysql i.e 1000, 2000 and also Configured SER Client(Kphone) with SER Server. 2) I have Installed and Configured Jabber Server i.e WildFire on IP(xyz.xyz.xyz.yy), and Added two(2) user, i.e 1000, 2000 ans also configured Jabber Client(Exodus) with Jabber Server. 3)My aim is to send Missed call alert, when a call is made from SER Client(K phone) User name:1000 to SER Client(Kphone) User name:2000, and User 2000 is not replyed. SER Server should send Instant Message(IM) to Jabber Client( User name: 2000) saying one missed call from user 1000. 4)Ser Server is running on xyz.xyz.xyz.xx Jabber Server is running on xyz.xyz.xyz.yy Came you please send me ser.cfg with give me this feature ( SER integration with Jabber IM ). And step to Integrate with Jabber IM. My configuration is attached as ser.cfg. Thanks & Regards ************************************ Sriram Srinivas ************************************

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[Serusers] How to Integrate SER Server with Jabber IM.
by srinivas sriram 31 Aug '06

31 Aug '06

Hello, I am look to send Missed Call Alert Instant Message(IM) to Jabber Client from SER Server. 1) I have Installed and Configured SER on IP(xyz.xyz.xyz.xx), and Added two(2) user in mysql i.e 1000, 2000 and also Configured SER Client(Kphone) with SER Server. 2) I have Installed and Configured Jabber Server i.e WildFire on IP(xyz.xyz.xyz.yy), and Added two(2) user, i.e 1000, 2000 ans also configured Jabber Client(Exodus) with Jabber Server. 3)My aim is to send Missed call alert, when a call is made from SER Client(K phone) User name:1000 to SER Client(Kphone) User name:2000, and User 2000 is not replyed. SER Server should send Instant Message(IM) to Jabber Client( User name: 2000) saying one missed call from user 1000. 4)Ser Server is running on xyz.xyz.xyz.xx Jabber Server is running on xyz.xyz.xyz.yy Came you please send me ser.cfg with give me this feature ( SER integration with Jabber IM ). And step to Integrate with Jabber IM. My configuration is attached as ser.cfg. Thanks & Regards ************************************ Sriram Srinivas ************************************

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RE: [Users] openser admin question..
by peter.3.edwards＠bt.com 31 Aug '06

31 Aug '06

> Is an error like this normal when doing the db install? > > ERROR 1064 (42000): You have an error in your SQL syntax; check the > manual that corresponds to your MySQL server version for the right > syntax to use near 'VIEW usr_preferences_view AS SELECT id, username, > domain, attribute, type AS att' at line 1 Hi, Nick. Not that it helps you much - but, no, I don't think that's normal - I didn't get this when installing on MySQL 5.0.24 (latest stable version), compiled from source. You might get compatibility issues if your version of MySQL is out of date ... what are you running? I'm sure Mike Williams will be along to help you debug it ... > By the way, while ruby on rails might be a great development > platform, > it's a right pain in the behind to install. Completely agree - especially when you're in a network which requires the use of a proxy to reach the web ... Bleugh. Still, once it's up and running it should be easy to maintain, or so I'm led to believe! Peter. -- Peter Edwards tel +44 (0) 1473 606492 This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

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[Serusers] Granstream AtA g729 codec --- help please
by ravi reddy 31 Aug '06

31 Aug '06

---------- Forwarded message ---------- From: ravi reddy <mravikreddy(a)gmail.com> Date: Aug 29, 2006 9:56 AM Subject: Re: [Serusers] Granstream AtA g729 codec --- help please To: "Greger V. Teigre" <greger(a)teigre.com> Thanks Greger for asking me, I want a small information regarding billing of sip calls (even though it is not related SER) please make a note to me :-) here in radacct i am getting more than one record for every call ; some folks told that i need some perl script to format all the database entries and write a fresh copy so that i can get one record for one call which is easy for billing. So, what i need a suggestion from you is 1) Do i need to learn perl language and write the script ?. 2) or is there any other way to control the overflow of start stop messages in to SER ...?. If you know please tell me . Thank You. Regards, Ravi. On 8/28/06, Greger V. Teigre <greger(a)teigre.com> wrote: > > Good to hear! > g-) > PS! No need to "sir"-me. Keep it informal, I'm Greger ;-) > > > ravi reddy wrote: > > Dear Sir, > > Thanks for your reply. i found the problem :-) > > the problem is itself in my pstn gateway that they have to configure to > allow my SER sending g729 codecs , > > i came to know from the folling grep message. > > v=0. > o=32331001 8000 8001 IN IP4 192.168.0.74. > s=SIP Call. > c=IN IP4 81.21.33.35. > t=0 0. > m=audio 60040 RTP/AVP 18 4 99 2. > a=sendrecv. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:99 iLBC/8000. > a=fmtp:99 mode=20. > a=rtpmap:2 G726-32/8000. > a=ptime:20. > > > # > U 81.21.33.35:5060 -> 81.21.34.34:5068 > SIP/2.0 501 Not Implemented. > Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received= 81.21.34.34 > ;branch=z9hG4bK8bcaffffe66dffff. > From: "ravi" <sip:32331001@81.21.33.35>;tag=502effff5ddeffff. > To: <sip:99106883@81.21.33.35>;tag=E067A27C-3ED. > Date: Mon, 28 Aug 2006 08:24:22 GMT. > Call-ID: 61d40000ddc8ffff(a)192.168.0.74. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 47822 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > > > because this 501 is internal server or gateway error so iam working on > that now. > > Thankyou. > > Regards, > Ravi. > > > > > On 8/28/06, Greger V. Teigre <greger(a)teigre.com > wrote: > > > > My gut feeling would tell me that the codec has nothing to do with it > > unless you find error messages in your ser log or var/log/messages. Sure > > nothing else has changed? > > g-) > > > > ravi reddy wrote: > > Hi SER users, > > > > Iam using SER-0.9.6 with mediaproxy-0.5 and every think > > works fine except with the codecs . G711 a & g711 ulaw works fine but when i > > tuned grandstream settings to use g729 codec for pstn calls the call is not > > done by the Mediaproxy server . > > > > but when i > > created a g729 fake rtp generator it looks fine in sessions.py . so in > > order to forward the g729 phone call in to the pstn world what i have to do > > ?. > > > > Some folks told that "integrate SER with Asterisk Works!" is it really > > works for me? > > > > my pstn provider supports all codecs . ofcourse iam working in that > > company itself. > > > > please suggest some thing , so that I can bye pass this problem. > > > > > > Thank You. > > > > > > Regards, > > Ravi. > > > > ------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > > > > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

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