- sr-users - kamailio.org

[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--… [View More]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. [View Less]

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[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 - … [View More]

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[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 … [View More]

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[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.

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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",… [View More]

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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

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[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 ,… [View More]

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[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 … [View More]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. [View Less]

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[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 … [View More]

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[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

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[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 … [View More]

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[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

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[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

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[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 … [View More]

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[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=…

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[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 --- … [View More]

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[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…

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[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.

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[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 … [View More]

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[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 … [View More]

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[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:… [View More]

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[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

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[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

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[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.

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[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 … [View More]

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[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 … [View More]

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[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 … [View More]

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[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 … [View More]

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[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 … [View More]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 [View Less]

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[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 … [View More]

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@… [View More]

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=… [View More]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 [View Less]

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[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.… [View More]

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[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 … [View More]

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[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

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[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 … [View More]

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 … [View More] ===============> 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. [View Less]

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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 … [View More]

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[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 … [View More]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. [View Less]

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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 … [View More]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. [View Less]

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 … [View More] # 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(); }; } .........................) [View Less]

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[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 … [View More]

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: $… [View More]

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. … [View More]

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 … [View More]

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 … [View More]

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 … [View More]

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 /… [View More]

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 … [View More]

1 0

[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 … [View More]

2 3

[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 … [View More]

1 0

[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 … [View More]

2 1

[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 … [View More]

1 0

[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 "… [View More]

3 2

[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 … [View More]

2 2

[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.-

3 2

[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 … [View More]

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[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

2 1

[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 … [View More]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 [View Less]

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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 … [View More]

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[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 … [View More]

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[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

2 1

[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.… [View More]

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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 … [View More]

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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 … [View More]

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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

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[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 … [View More]

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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 ?................ … [View More]

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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 … [View More]

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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 … [View More]

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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-… [View More]

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[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 … [View More]

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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 … [View More]

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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 … [View More]

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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), … [View More]

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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 … [View More]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 > > > > > [View Less]

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[Serusers] Any plans to implement UAC module?
by Andy Thomas 31 Aug '06

31 Aug '06

Probably one for Greger- Does anyone know if there are plans to implement the UAC module (as seen in Openser) within the next version of Ser? I am currently running 0.9.6, and when I dial out to the PSTN, the CLI is shown as the Ser username, not the DID number All incoming DID numbers are specified as aliases, so when the PSTN dials in, it gets forward to the correct user. However I need to rewrite the outgoing SIP message, changing the the from: URI so that the alias is displayed, … [View More]

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Re: [Users] openser admin
by Mike Williams 30 Aug '06

30 Aug '06

On Monday 28 August 2006 03:15, raviprakash sunkara wrote: > Hi Mike , > Thanks for replying the mail , > openser administrator is wrriten in ruby on rails .. which is MVC structure Correct. > in database interface in web interface.. > If I commited in the subscriber in ur list... > In grp table also have to change from id to phplib_id.. No, that shouldn't be the case unless I'm mistaken. In the subscriber table, there was originally a phplib_id. I made the mistake of … [View More]replacing it with id, instead of using both. The grp table never had a phplib_id, so id wasn't replacing anything. > > Tell me is openser administrator in fully stable and compitable with > openser. I see no reason why it should not be. > > But i think it may be unstable or stable version .. > i'm sure openser administrator 0.2 is unstable version > I need to know about the 0.3 .... > please for give me , if any mistakens in my secenrio .... Hopefully, they'll all be stable versions. However, it is still a very new project, and I cannot say, that with so few regular users of it, all the bugs are out. Mike Williams > > On 8/26/06, Mike Williams <mike(a)mikebwilliams.com> wrote: > > Peter & Sunkara, > > > > Thanks for the good idea, I hadn't even thought of uninstall or upgrade > > situations. > > > > The phplib_id problem can be solved by readding that column. There is a > > commented-out SQL statement in the SQL script that you can use. It is > > listed > > below as well. > > > > ALTER TABLE subscriber ADD COLUMN phplib_id varchar(32) NOT NULL default > > '' > > FIRST; > > > > I will work on these things today. Also, please let me know of any > > feature requests you have, otherwise I will have nothing to work on. > > > > Thanks, > > > > Mike Williams > > > > On Friday 25 August 2006 06:58, peter.3.edwards(a)bt.com wrote: > > > Mike, > > > > > > What's the upgrade procedure to go from 0.2 -> 0.3? > > > > > > Presumably readd the phplib_id column as per your mail on 21 Aug ... > > > and, what - just replace the old openseradmin-0.2 directory with this > > > one (including creating a new database.yml file)? > > > > > > Also, is there a way to uninstall OpenSER Administrator without > > > effecting OpenSER ... Presumably, just delete the openseradmin-x > > > directory ... But is there an equivalent SQL script to uninstall it's > > > database component? > > > > > > I think it would be good if you could include upgrade and uninstall > > > instructions within the release notes / on the website. > > > > > > Cheers, > > > > > > Peter. > > > > > > > -----Original Message----- > > > > From: users-bounces(a)openser.org > > > > [mailto:users-bounces@openser.org] On Behalf Of Mike Williams > > > > Sent: 24 August 2006 21:02 > > > > To: users(a)openser.org > > > > Subject: [Users] openser admin > > > > > > > > All, > > > > > > > > OpenSER Admin version 0.3 has been released. This fixes the bug where > > > > compatibility with openserctl was broken, and adds the > > > > ability to add other > > > > management accounts to the interface. > > > > > > > > Let me know what you think. > > > > > > > > Thanks, > > > > > > > > Mike Williams > > > > > > > > _______________________________________________ > > > > Users mailing list > > > > Users(a)openser.org > > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users [View Less]

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RE: [Users] openser admin
by peter.3.edwards＠bt.com 30 Aug '06

30 Aug '06

Hi, Mike. I've noticed that you've added Upgrade and Uninstall options to the OpenSER Administrator website - great stuff. Although, I did notice that your upgrade instructions don't mention anything about upgrading the database tables - I presume if you need to do that at a later date, you'll supply an equivalent SQL script (like the database_tables.sql and delete_database_tables.sql scripts for install and uninstall, respectively)? Might be worth mentioning that. As for your request for … [View More]ideas for new features ... I think the documentation within OpenSER admin could do with a bit more content - I'm quite new to OpenSER and OpenSER Administrator so I'm not entirely clear what VoIP Accounts, Domains, Groups, Group Membership etc. specifically refer to (well, I get accounts and domains but it gets a bit fuzzier after that). Perhaps a little paragraph at the top of each page with a brief but clear description? On top of that, I guess I come less from a user-centric world and lean more towards the administrator side of things so I'm less interested in letting users login to change their profiles and more interested in seeing options that will allow easier adminstration - such as a web interface to configuring parts of the openser.cfg file - being able to toggle the debug level at will, stop/start/restart OpenSER from the web, change the port OpenSER listens on, enable/disable modules on startup - and, generally, anything that can be web-ified from the .cfg file. Obviously, I'm not talking about a replacement to the .cfg file - but something that can modify some of the various options on the fly would be useful ... Anyway, just thought I'd make some suggestions. I finally got the logging feature working today (oh, yes, when the log file doesn't exist, OpenSER doesn't handle the error very gracefully - so perhaps some work required on the error handling?) so I think I can safely say everything's working ... Many thanks, Peter. > -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Mike Williams > Sent: 26 August 2006 15:07 > To: users(a)openser.org > Subject: Re: [Users] openser admin > > > Peter & Sunkara, > > Thanks for the good idea, I hadn't even thought of uninstall > or upgrade > situations. > > The phplib_id problem can be solved by readding that column. > There is a > commented-out SQL statement in the SQL script that you can > use. It is listed > below as well. > > ALTER TABLE subscriber ADD COLUMN phplib_id varchar(32) NOT > NULL default '' > FIRST; > > I will work on these things today. Also, please let me know > of any feature > requests you have, otherwise I will have nothing to work on. > > Thanks, > > Mike Williams -- Peter Edwards 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. [View Less]

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Re: [Serusers] Serweb: HTTP 403 Forbidden error
by Roberto Lopes 30 Aug '06

30 Aug '06

Hi to all, My apache errror log just showed me this: [Tue Aug 29 15:45:57 2006] [error] [client 193.137.239.11] (13)Permission denied: access to /serweb/ denied Roberto > > On 8/29/06, Karel Kozlik <karel(a)iptel.org> wrote: > > > > hi, > > what your apache error log say? > > I think there are no read rights for apache on serweb files. > > > > Karel > > > > Roberto Lopes napsal(a): > > > > > > Dear SER Users, &… [View More]gt; > > > > > > > > I've a problem with Serweb and I hope somebody could help me out > > > I know this problem was already talked about but i never saw a > > solution > > > to it, so I'll post it again hoping somebody will know how to solve > > this > > > mistake. > > > So the problem is that when i try to put serweb working in my firefox > > > browser the following error is shown: > > > > > > > > > Forbidden > > > > > > You don't have permission to access /serweb/index.html on this server. > > > > > > Additionally, a 403 Forbidden error was encountered while trying to > > use > > > an ErrorDocument to handle the request. > > > > > > > > ------------------------------------------------------------------------ > > > Apache/2.2.0 (Fedora) Server at xxx.xxx.xxx.xxx Port 80 > > > > > > > > > I think i installed everything correctly. > > > > > > I'll show some of the configurations I've made > > > > > > #1 - In "/etc/php.ini" I changed the memory limit from 8 to 12M > > because > > > of the Serweb extension classkit.so > > > … > > > memory_limit = 12M > > > > > > #2 - I used pear to install classkit.so > > > > > > pear install <path to classkit-0.4.tgz> > > > > > > #3 - In "php.ini" > > > … > > > ;;;;;;;;;;;;;;;;;;;;;; > > > ; Dynamic Extensions ; > > > ;;;;;;;;;;;;;;;;;;;;;; > > > … > > > extension=classkit.so > > > > > > #4 - In "/serweb-0.4/config/config_data_layer.php": > > > … > > > $config->data_sql->host[$i]['host']="<IP MySQL Server>"; > > > … > > > $config->data_sql->host[$i][ > > > 'user']="<DB User>"; > > > $config->data_sql->host[$i]['pass']="<User password>"; > > > .. > > > > > > #5 - In "/root/instalacao/serweb-0.9.4/config/config_paths.php" > > > > > > $config->root_path="/root/instalacao/serweb- 0.9.4/html/"; > > > > > > #6 - I created a log file > > > touch /var/log/serweb > > > chown apache /var/log/serweb > > > > > > #7 - I created a folder for SIP domain > > > mkdir /root/installation/serweb-0.4/html/domains/xxx.xxx.xxx.xxx(IP of > > SER) > > > > > > #8 - I copied the files and folder from the default folder > > > ("/root/installation/serweb-0.4/html/domains/_default/") to the one i > > > created. > > > > > > #9 - I specified the SIP domain in the file > > > "/root/installation/serweb-0.4/config/set_domain.php": > > > … > > > $config->domain = "xxx.xxx.xxx.xxx(IP of SER)"; > > > … > > > > > > #10 - I enabled logging in file > > > "/root/installation/serweb-0.4/config/config.php": > > > … > > > $config->enable_loging = true; > > > $config->log_file = "/var/log/serweb; > > > > > > #11 - In /etc/httpd/conf/httpd.conf I added an alias > > > … > > > > > > <Directory "/var/www/html"> > > > Options Indexes FollowSymLinks > > > AllowOverride None > > > Order deny,allow > > > Allow from all > > > </Directory> > > > > > > <IfModule mod_userdir.c> > > > UserDir disable root > > > #UserDir public_html > > > </IfModule> > > > > > > Alias /serweb/ "/root/instalacao/serweb-0.9.4/html" > > > <Directory "/root/instalacao/serweb-0.9.4/html"> > > > Options Indexes MultiViews > > > Options +FollowSymlinks > > > AllowOverride None > > > Order Deny,Allow > > > Allow from all > > > </Directory> > > > ... > > > > > > #12 - I then put some commands for Apache to begin when computer would > > > > > start > > > > > > chkconfig httpd on > > > service httpd start > > > > > > #13 - I introduced the following URL > > > > > > http://<IP of SER>/<alias i wrote in httpd.conf>/ > > > http://xxx.xxx.xxx.xxx/serweb/ > > > > > > > > > And then that HTTP 403 error shows up > > > > > > I really hope somebody can help me because I'm doing a VoIP project > > for > > > my school and I'd really like to have everything done in time. > > > > > > Thank you for your attention, > > > Roberto Lopes > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > [View Less]

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[Serusers] FW: [Serdev] variable header field value comparation
by Luis Silva 30 Aug '06

30 Aug '06

>From: "Luis Silva" <lfs12(a)hotmail.com> >To: serdev(a)iptel.org >Subject: [Serdev] variable header field value comparation >Date: Fri, 25 Aug 2006 13:44:53 +0000 > >Hi all!! > >I need to compare an header field value with an avp. I tried to use the >hf_value_exists("header_name","$value") but the header that I need to >compare isn't static. I replaced the header_name with an avp but it didn't >work. then I tried to use avp_check("$value", "eq/$hdr($… [View More]

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[Serusers] Error with credentials
by Daniel Mirrha Santos 30 Aug '06

30 Aug '06

Does anybody know what is this error about "is_user_in(): No authorized credentials found (error in scripts)"? Thanks, Daniel

3 9

[Serusers] SIP soft client supporting G.728
by VIDOVIC Mato 30 Aug '06

30 Aug '06

Hi, I am looking for a Windows SIP soft client supporting G.728 codec for narrowband interworking with a PSTN gateway. Can someone help me? Best regards, Mato

2 1

[Serusers] caller=callee, Not giving me Busy
by ram 29 Aug '06

29 Aug '06

Hi all I have added the Follwing Line in my ser.cfg if (!uri==myself) { t_relay(); break; }; But iam not getting Busy Tone Ram

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[Serusers] t_was_cancelled() equivalent function for Ser?
by Ricardo Carvalho 29 Aug '06

29 Aug '06

Hi, I'm using Ser and I think I need to use the same functionality that the function t_was_cancelled() from openser does. As much as I know, this functions still doesn't exist in Ser. Is there any other that does the same? (I need it to be able to skip sending a call to Asterisk voicemail every time that a call is hangup by the caller when it is still ringing on the callee. If I don't such kind of function, if the caller hungs up when still ringing the callee UA, Ser sends the call to … [View More]

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[Serusers] Via Format?
by Ricardo Martinez 29 Aug '06

29 Aug '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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RE: [Users] Please help me in Openser Admin
by peter.3.edwards＠bt.com 29 Aug '06

29 Aug '06

> 1. Is openser adminstrator a web application? I mean, does it have > same function like "serweb" in ser? "OpenSER Administrator: Mission The mission is to provide a simple interface to OpenSER that can be quickly and easily modified and upgraded. It is not designed to provide an interface for the customer, but is to be used only for administration duties." http://openseradmin.sourceforge.net/ > 2. How to install the openser administrator in openser server? > I use openser1.1.0-… [View More]tls and what packets does it need? OpenSER Administrator: Installation http://openseradmin.sourceforge.net/docs.php > 3.Is the openser administrator just used by the administrator to manage > his/her server? or The openser administrator can be used like a > serweb?(the client also can use it) because as I read from the reply > message in this forum, the serweb can not use for openser1.1.0-tls See answer to question 1. > 4. If it is not like serweb, can anybody tell me what is the suitable > web application for openser ? As far as I'm aware from reading the website and mailing list archives, the only web application designed for use with OpenSER is OpenSER Administrator. It sounds like it might be possible to modify SERweb but that would require going "under the hood" ... Ferianto, all of this information is easily found from the website - rather copying your mail to three mailing lists which each have a slightly different remit, you may find you'd get a better response if you do some initial research first and then direct any further questions you still have, to the most appropriate list instead of sending to all of them at once. Even Microsoft know ... How to ask a question http://support.microsoft.com/?kbid=555375 HTH, Peter. -- Peter Edwards 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. [View Less]

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[Serusers] verify if an avp is null
by Joana Smith 29 Aug '06

29 Aug '06

Hi everyone!! How can I verify if an avp is null?? I tried if( $avp_name == '') {} but this doesn't work. any suggestions? tks a lot ps - I know that is possible to use is_sattr_set but I think avp module is deprecated in favor of avpops _______________________________________________________________________________________ Sabe em quanto tempo damos resposta para ter 4000Euros? Saber mais em http://www.iol.pt/correio/rodape.php?dst=0607071

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[Users] Voice-System SRL OpenSER Syslog Tutorial note suggestion
by peter.3.edwards＠bt.com 29 Aug '06

29 Aug '06

Hi, I've modified the Basic OpenSER Logging tutorial, to include a Solaris compatible entry for /etc/syslog.conf (note included below): http://www.openser.org/dokuwiki/doku.php?id=basic_openser_syslog_tutoria l The Voice-System SRL OpenSER Syslog Tutorial, could also do with having the same note added to this page: http://www.voice-system.ro/docs/ser-syslog/ar01s06.html Daniel, I've noted that you post here and that you're also the author of the above doc ... I thought you might want to … [View More]

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[Serusers] Ser x Postgres
by Daniel Mirrha Santos 29 Aug '06

29 Aug '06

Its possible to use postgres to authenticate ser users? What I have to do to make it work with postgres? Thanks, Daniel

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[Serusers] Build problem
by Greg Martin 29 Aug '06

29 Aug '06

I'm trying to build ser-0.9.6 on a sparc machine. 180> uname -rvmap SunOS tc90dv03 5.8 Generic_108528-13 sun4u sparc SUNW,UltraAX-i2 The errors I'm getting are: 172> make INSTALL=install include_modules="mysql" Makefile.rules:80: action.d: No such file or directory Makefile.rules:80: crc.d: No such file or directory Makefile.rules:80: daemonize.d: No such file or directory Makefile.rules:80: data_lump.d: No such file or directory Makefile.rules:80: data_lump_rpl.d: No such file or … [View More]directory Makefile.rules:80: dprint.d: No such file or directory Makefile.rules:80: dset.d: No such file or directory Makefile.rules:80: error.d: No such file or directory Makefile.rules:80: fifo_server.d: No such file or directory Makefile.rules:80: flags.d: No such file or directory Makefile.rules:80: forward.d: No such file or directory Makefile.rules:80: hash_func.d: No such file or directory Makefile.rules:80: ip_addr.d: No such file or directory Makefile.rules:80: main.d: No such file or directory Makefile.rules:80: md5.d: No such file or directory Makefile.rules:80: md5utils.d: No such file or directory Makefile.rules:80: modparam.d: No such file or directory Makefile.rules:80: msg_translator.d: No such file or directory Makefile.rules:80: pass_fd.d: No such file or directory Makefile.rules:80: proxy.d: No such file or directory Makefile.rules:80: qvalue.d: No such file or directory Makefile.rules:80: re.d: No such file or directory Makefile.rules:80: receive.d: No such file or directory Makefile.rules:80: resolve.d: No such file or directory Makefile.rules:80: route.d: No such file or directory Makefile.rules:80: route_struct.d: No such file or directory Makefile.rules:80: script_cb.d: No such file or directory Makefile.rules:80: socket_info.d: No such file or directory Makefile.rules:80: sr_module.d: No such file or directory Makefile.rules:80: stats.d: No such file or directory Makefile.rules:80: tcp_main.d: No such file or directory Makefile.rules:80: tcp_read.d: No such file or directory Makefile.rules:80: timer.d: No such file or directory Makefile.rules:80: tsend.d: No such file or directory Makefile.rules:80: udp_server.d: No such file or directory Makefile.rules:80: unixsock_server.d: No such file or directory Makefile.rules:80: usr_avp.d: No such file or directory Makefile.rules:80: mem/f_malloc.d: No such file or directory Makefile.rules:80: mem/mem.d: No such file or directory Makefile.rules:80: mem/memtest.d: No such file or directory Makefile.rules:80: mem/q_malloc.d: No such file or directory Makefile.rules:80: mem/shm_mem.d: No such file or directory Makefile.rules:80: mem/vq_malloc.d: No such file or directory Makefile.rules:80: parser/hf.d: No such file or directory Makefile.rules:80: parser/msg_parser.d: No such file or directory Makefile.rules:80: parser/parse_allow.d: No such file or directory Makefile.rules:80: parser/parse_content.d: No such file or directory Makefile.rules:80: parser/parse_cseq.d: No such file or directory Makefile.rules:80: parser/parse_disposition.d: No such file or directory Makefile.rules:80: parser/parse_diversion.d: No such file or directory Makefile.rules:80: parser/parse_event.d: No such file or directory Makefile.rules:80: parser/parse_expires.d: No such file or directory Makefile.rules:80: parser/parse_fline.d: No such file or directory Makefile.rules:80: parser/parse_from.d: No such file or directory Makefile.rules:80: parser/parse_hname2.d: No such file or directory Makefile.rules:80: parser/parse_hostport.d: No such file or directory Makefile.rules:80: parser/parse_methods.d: No such file or directory Makefile.rules:80: parser/parse_nameaddr.d: No such file or directory Makefile.rules:80: parser/parse_param.d: No such file or directory Makefile.rules:80: parser/parse_rpid.d: No such file or directory Makefile.rules:80: parser/parse_rr.d: No such file or directory Makefile.rules:80: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:80: parser/parse_to.d: No such file or directory Makefile.rules:80: parser/parse_uri.d: No such file or directory Makefile.rules:80: parser/parse_via.d: No such file or directory Makefile.rules:80: parser/parser_f.d: No such file or directory Makefile.rules:80: parser/digest/digest.d: No such file or directory Makefile.rules:80: parser/digest/digest_parser.d: No such file or directory Makefile.rules:80: parser/digest/param_parser.d: No such file or directory Makefile.rules:80: parser/contact/contact.d: No such file or directory Makefile.rules:80: parser/contact/parse_contact.d: No such file or directory Makefile.rules:80: db/db.d: No such file or directory Makefile.rules:80: db/db_fifo.d: No such file or directory Makefile.rules:80: lex.yy.d: No such file or directory Makefile.rules:80: cfg.tab.d: No such file or directory yacc -d -b cfg cfg.y conflicts: 1 shift/reduce make: *** [cfg.tab.c] Broken Pipe Any help would be appreciated, thanks. Greg Martin CTO Consumer Application Architecture TELUS Communications Company Phone: (780) 493-2786 Cell: (780) 718-4139 [View Less]

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[Users] Question about GFLAGS Module
by Michel Zenone 29 Aug '06

29 Aug '06

Hi! I wonder if there is only a unique bitmap of flags or if there is a bitmap of flags for each SIP dialog? Thanks! Michel

2 1

[Users] Using subst() of Textops module
by Leonid Fainshtein 29 Aug '06

29 Aug '06

Hi, In the textops module doc (http://openser.org/docs/modules/1.0.x/textops.html#AEN124) there is an example that demonstrates how to replace URI in the header "To" with the message URI: # replace the uri in to: with the message uri (just an example) if ( subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/t:\1\u\2/ig') ) {}; After I've fixed a small mistake (instead of /t: I put /To:) it worked. But I don't understand how does it work. It looks like "u" is substituted by URI from R-URI but I … [View More]

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[Serusers] Serweb: HTTP 403 Forbidden error
by Roberto Lopes 29 Aug '06

29 Aug '06

Dear SER Users, I've a problem with Serweb and I hope somebody could help me out I know this problem was already talked about but i never saw a solution to it, so I'll post it again hoping somebody will know how to solve this mistake. So the problem is that when i try to put serweb working in my firefox browser the following error is shown: Forbidden You don't have permission to access /serweb/index.html on this server. Additionally, a 403 Forbidden error was encountered while trying to use … [View More]an ErrorDocument to handle the request. ------------------------------ Apache/2.2.0 (Fedora) Server at xxx.xxx.xxx.xxx Port 80 I think i installed everything correctly. I'll show some of the configurations I've made #1 - In "/etc/php.ini" I changed the memory limit from 8 to 12M because of the Serweb extension classkit.so … memory_limit = 12M #2 - I used pear to install classkit.so pear install <path to classkit-0.4.tgz> #3 - In "php.ini" … ;;;;;;;;;;;;;;;;;;;;;; ; Dynamic Extensions ; ;;;;;;;;;;;;;;;;;;;;;; … extension=classkit.so #4 - In "/serweb-0.4/config/config_data_layer.php": … $config->data_sql->host[$i]['host']="<IP MySQL Server>"; … $config->data_sql->host[$i]['user']="<DB User>"; $config->data_sql->host[$i]['pass']="<User password>"; .. #5 - In "/root/instalacao/serweb-0.9.4/config/config_paths.php" $config->root_path="/root/instalacao/serweb- 0.9.4/html/"; #6 - I created a log file touch /var/log/serweb chown apache /var/log/serweb #7 - I created a folder for SIP domain mkdir /root/installation/serweb-0.4/html/domains/xxx.xxx.xxx.xxx(IP of SER) #8 - I copied the files and folder from the default folder ("/root/installation/serweb-0.4/html/domains/_default/") to the one i created. #9 - I specified the SIP domain in the file "/root/installation/serweb-0.4/config/set_domain.php": … $config->domain = "xxx.xxx.xxx.xxx(IP of SER)"; … #10 - I enabled logging in file "/root/installation/serweb-0.4 /config/config.php": … $config->enable_loging = true; $config->log_file = "/var/log/serweb; #11 - In /etc/httpd/conf/httpd.conf I added an alias … <Directory "/var/www/html"> Options Indexes FollowSymLinks AllowOverride None Order deny,allow Allow from all </Directory> <IfModule mod_userdir.c> UserDir disable root #UserDir public_html </IfModule> Alias /serweb/ "/root/instalacao/serweb-0.9.4/html" <Directory "/root/instalacao/serweb-0.9.4/html"> Options Indexes MultiViews Options +FollowSymlinks AllowOverride None Order Deny,Allow Allow from all </Directory> ... #12 - I then put some commands for Apache to begin when computer would start chkconfig httpd on service httpd start #13 - I introduced the following URL http://<IP of SER>/<alias i wrote in httpd.conf>/ http://xxx.xxx.xxx.xxx/serweb/ And then that HTTP 403 error shows up I really hope somebody can help me because I'm doing a VoIP project for my school and I'd really like to have everything done in time. Thank you for your attention, Roberto Lopes [View Less]

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Re: [Serusers] Proxy Authentication
by Andrey Kouprianov 29 Aug '06

29 Aug '06

Hi, Why are you selecting 313814? The problem is with 313812 isnt it? Your trace log shows the conflict between 313812 and 313813 (not 313814)... :) Here's the examle (from RFC document) of authorizationm response from the client back to the server with a 2nd INVITE... Authorization: Digest username="bob", realm="biloxi.com", nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093", uri="sip:bob@biloxi.com", qop=auth, … [View More] nc=00000001, cnonce="0a4f113b", response="6629fae49393a05397450978507c4ef1", opaque="5ccc069c403ebaf9f0171e9517f40e41" As you can see uri is "sip:bob@bioxy.com" as well as the username is "bob" as well (i.e. they are same)... That gives me an impression there's something wrong with ur client-side setup.. Regards, Andrey. On 8/29/06, Javier Oviedo <joviedo(a)plcendesa.com> wrote: > Andrey Kouprianov escribió: > > Hi, > > > > First of all it is GREAT that you've sent your logs and ser.cfg!! I > > wish everyone would do it (however, this is rarely the case). > > > > You SJPhone cannot call, because for some reason your URI and username > > differ. Take a closer look at this part: > > > > Proxy-Authorization: Digest > > username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 > > > > > > The username is 313813, but the URI is sip:313812@sipserver.es. Notice > > any difference? Plz, take a look at your account setup in SJPhone > > closely. I think the problem should be there. > > > > Regards, > > > > Andrey. > > > > On 8/28/06, Javier Oviedo <joviedo(a)plcendesa.com> wrote: > >> > >> Hi all > >> > >> In my network schema I have a Sip Express Router to authenticate my > >> outbound > >> calls to a PSTN. I add a new equipment to a ser with "serctl add user > >> pass > >> e-mail" command. My softphone is registered in proxy but when I make > >> a call > >> to a PSTN I obtain the following error: Call rejected: 407 Proxy > >> Authentication Rquired : > >> > >> I use the SJPhone software. > >> > >> Any idea about where is the problem > >> > >> Thanks in advance > >> Joe > >> > >> My ser configuration is: > >> > >> debug=3 > >> fork=no > >> log_stderror=yes > >> > >> port=5060 > >> children=4 > >> > >> dns=no > >> rev_dns=no > >> fifo="/tmp/ser_fifo" > >> fifo_db_url="mysql://ser:heslo@localhost/ser" > >> > >> loadmodule "/usr/local/lib/ser/modules/mysql.so" > >> loadmodule "/usr/local/lib/ser/modules/sl.so" > >> loadmodule "/usr/local/lib/ser/modules/tm.so" > >> loadmodule "/usr/local/lib/ser/modules/rr.so" > >> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > >> loadmodule "/usr/local/lib/ser/modules/usrloc.so" > >> loadmodule "/usr/local/lib/ser/modules/registrar.so" > >> loadmodule "/usr/local/lib/ser/modules/auth.so" > >> loadmodule "/usr/local/lib/ser/modules/auth_db.so" > >> loadmodule "/usr/local/lib/ser/modules/uri_db.so" > >> > >> modparam("auth_db|uri_db|usrloc", "db_url", > >> "mysql://ser:heslo@localhost/ser") > >> modparam("auth_db", "calculate_ha1", 1) > >> modparam("auth_db", "password_column", "password") > >> modparam("usrloc", "db_mode", 2) > >> modparam("rr", "enable_full_lr", 1) > >> > >> route { > >> > >> # > >> ----------------------------------------------------------------- > >> # Sanity Check Section > >> # > >> ----------------------------------------------------------------- > >> if (!mf_process_maxfwd_header("10")) { > >> sl_send_reply("483", "Too Many Hops"); > >> break; > >> }; > >> > >> if (msg:len > max_len) { > >> sl_send_reply("513", "Message Overflow"); > >> break; > >> }; > >> > >> # > >> ----------------------------------------------------------------- > >> # Record Route Section > >> # > >> ----------------------------------------------------------------- > >> if (method!="REGISTER") { > >> record_route(); > >> }; > >> > >> # > >> ----------------------------------------------------------------- > >> # Loose Route Section > >> # > >> ----------------------------------------------------------------- > >> if (loose_route()) { > >> route(1); > >> break; > >> }; > >> > >> # > >> ----------------------------------------------------------------- > >> # Call Type Processing Section > >> # > >> ----------------------------------------------------------------- > >> if (uri!=myself) { > >> route(1); > >> break; > >> }; > >> > >> if (method=="ACK") { > >> route(1); > >> break; > >> } if (method=="INVITE") { > >> route(3); > >> break; > >> } else if (method=="REGISTER") { > >> route(2); > >> break; > >> }; > >> > >> lookup("aliases"); > >> > >> if (uri!=myself) { > >> route(1); > >> break; > >> }; > >> > >> if (!lookup("location")) { > >> sl_send_reply("404", "User Not Found"); > >> break; > >> }; > >> > >> route(1); > >> } > >> > >> route[1] { > >> > >> # > >> ----------------------------------------------------------------- > >> # Default Message Handler > >> # > >> ----------------------------------------------------------------- > >> if (!t_relay()) { > >> sl_reply_error(); > >> }; > >> } > >> > >> route[2] { > >> > >> # > >> ----------------------------------------------------------------- > >> # REGISTER Message Handler > >> # > >> ---------------------------------------------------------------- > >> sl_send_reply("100", "Trying"); > >> > >> if (!www_authorize("localhost","subscriber")) { > >> www_challenge("localhost","0"); > >> break; > >> }; > >> > >> if (!check_to()) { > >> sl_send_reply("401", "Unauthorized"); > >> break; > >> }; > >> > >> consume_credentials(); > >> > >> if (!save("location")) { > >> sl_reply_error(); > >> }; > >> } > >> > >> route[3] { > >> # > >> ----------------------------------------------------------------- > >> # INVITE Message Handler > >> # > >> ----------------------------------------------------------------- > >> if (!proxy_authorize("localhost","subscriber")) { > >> proxy_challenge("localhost","0"); > >> break; > >> } else if (!check_from()) { > >> sl_send_reply("403", "Use From=ID"); > >> break; > >> }; > >> > >> consume_credentials(); > >> > >> lookup("aliases"); > >> if (uri!=myself) { > >> route(1); > >> break; > >> }; > >> > >> #if (!lookup("location")) { > >> # sl_send_reply("404", "User Not Found"); > >> # break; > >> #}; > >> > >> route(1); > >> } > >> > >> > >> > >> SoftPhone LOG: > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> INVITE sip:313812@sipserver.es SIP/2.0 > >> To: <sip:313812@sipserver.es> > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 1 INVITE > >> Contact: <sip:313813@172.25.97.138:9528> > >> Max-Forwards: 70 > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > >> SUBSCRIBE, > >> INFO > >> Content-Type: application/sdp > >> User-Agent: eyeBeam release 3004t stamp 16741 > >> Content-Length: 274 > >> > >> v=0 > >> o=- 32759240 32759303 IN IP4 172.25.97.138 > >> s=eyeBeam > >> c=IN IP4 172.25.97.138 > >> t=0 0 > >> m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 > >> a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 > >> a=fmtp:101 0-15 > >> a=rtpmap:100 speex/16000 > >> a=rtpmap:101 telephone-event/8000 > >> a=sendrecv > >> > >> 17:04:43.1 > >> RECEIVING FROM: 10.110.0.127:5060 > >> SIP/2.0 407 Proxy Authentication Required > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport=9528 > >> Call-ID: fc7b726c6545d126 > >> CSeq: 1 INVITE > >> Proxy-Authenticate: Digest realm="localhost", > >> nonce="44f307373cf9fb29da3028442e23f976513fc91d" > >> Server: Sip EXpress router (0.9.6 (i386/linux)) > >> Content-Length: 0 > >> Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 > >> req_src_ip=172.25.97.138 req_src_port=9528 > >> in_uri=sip:313812@sipserver.es > >> out_uri=sip:313812@sipserver.es via_cnt==1" > >> > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> ACK sip:313812@sipserver.es SIP/2.0 > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 1 ACK > >> Content-Length: 0 > >> > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> INVITE sip:313812@sipserver.es SIP/2.0 > >> To: <sip:313812@sipserver.es> > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 INVITE > >> Contact: <sip:313813@172.25.97.138:9528> > >> Max-Forwards: 70 > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > >> SUBSCRIBE, > >> INFO > >> Content-Type: application/sdp > >> Proxy-Authorization: Digest > >> username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 > >> > >> User-Agent: eyeBeam release 3004t stamp 16741 > >> Content-Length: 274 > >> > >> v=0 > >> o=- 32759240 32759303 IN IP4 172.25.97.138 > >> s=eyeBeam > >> c=IN IP4 172.25.97.138 > >> t=0 0 > >> m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 > >> a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 > >> a=fmtp:101 0-15 > >> a=rtpmap:100 speex/16000 > >> a=rtpmap:101 telephone-event/8000 > >> a=sendrecv > >> > >> 17:04:43.1 > >> RECEIVING FROM: 10.110.0.127:5060 > >> SIP/2.0 100 trying -- your call is important to us > >> To: <sip:313812@sipserver.es> > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 INVITE > >> Server: Sip EXpress router (0.9.6 (i386/linux)) > >> Content-Length: 0 > >> Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 > >> req_src_ip=172.25.97.138 req_src_port=9528 > >> in_uri=sip:313812@sipserver.es > >> out_uri=sip:313812@sipserver.es via_cnt==1" > >> > >> > >> 17:04:43.1 > >> RECEIVING FROM: 10.110.0.127:5060 > >> SIP/2.0 407 Proxy Authentication Required > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 INVITE > >> Proxy-Authenticate: Digest realm="localhost", > >> nonce="44f307373cf9fb29da3028442e23f976513fc91d" > >> Server: Sip EXpress router (0.9.6 (i386/linux)) > >> Content-Length: 0 > >> Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 > >> req_src_ip=10.110.0.127 req_src_port=5060 in_uri=sip:313812@sipserver.es > >> out_uri=sip:313812@sipserver.es via_cnt==2" > >> > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> ACK sip:313812@sipserver.es SIP/2.0 > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 ACK > >> Content-Length: 0 > >> > >> > >> 17:04:43.2 Call (l:'Joe' r:'sip:313812@sipserver.es') - Call being > >> terminated. Reasons: "Proxy Authentication Required", (code: 407) > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >> > >> > >> > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > Hi Andrey > > I think tha tihs part of log is correct: > > Proxy-Authorization: Digest > username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 > > > The username is the caller id and the uri is de destination number. My > SJphone configuration is simply: The user, pass and account name are > 313813. I attach my mysql subscriber table: > > > mysql> select * from subscriber where username=313814; > +----------------------------------+----------+----------------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+--------------+------+------------------+----------+----------------------------------+----------------------------------+------------+----------+------+------+------+ > | phplib_id | username | domain | > password | first_name | last_name | phone | email_address | > datetime_created | datetime_modified | confirmation | flag | > sendnotification | greeting | ha1 | > ha1b | allow_find | timezone | rpid | domn | > uuid | > +----------------------------------+----------+----------------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+--------------+------+------------------+----------+----------------------------------+----------------------------------+------------+----------+------+------+------+ > | 3fbf3938ba07fa0f280ffe44926e4cd3 | 313814 | sipserver.es | 313814 > | | | | joe(a)plcendesa.com | 2006-08-28 > 18:23:22 | 0000-00-00 00:00:00 | | o | > | | 3fbf3938ba07fa0f280ffe44926e4cd3 | > 147098ce3cd77298d89982cd37b5813a | 0 | NULL | NULL | NULL | > NULL | > +----------------------------------+----------+----------------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+--------------+------+------------------+----------+----------------------------------+----------------------------------+------------+----------+------+------+------+ > 1 row in set (0.00 sec) > > > and my uri table: > > mysql> select * from uri; > +----------+----------------------+----------+---------------------+ > | username | domain | uri_user | last_modified | > +----------+----------------------+----------+---------------------+ > | 313813 | sipserver.createc.es | 313813 | 2006-08-28 17:52:48 | > | 313814 | sipserver.createc.es | 313814 | 2006-08-28 18:23:22 | > +----------+----------------------+----------+---------------------+ > 2 rows in set (0.00 sec) > > > With the X-Lite softphone the same thing happens. > > Thanks in advace! > Regards > Joe > [View Less]

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[Users] Understanding the Call Flow
by Max Clark 29 Aug '06

29 Aug '06

Hi all, I am trying to get a better understanding of the call flow - specifically for distant users. In our current configuration we have all of the UAs registering to our OpenSER box which manages the authentication, and passes the call forward if allowed. Like so: UA -> SER -> PSTN GW -> PSTN I have been experimenting with using a mediaproxy/rtp proxy to improve the quality of service to users beyond the network edge - effectively: UA -> Proxy -> SER -> PSTN GW -> … [View More]

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[Users] Please help me in openser administrator...Please
by Ferianto siregar 29 Aug '06

29 Aug '06

Dear all, Thanks to all of you for your kind-hearted to give me solution in solving my problem. Thank you very much. Thank you very much to Andrey Kouprianov for replying my message. Thanks I have some questions about the openser admin. I do anybody can help me to understand what the openser admin use for and how to install it. These are my question: 1. Is openser adminstrator a web application? I mean, does it have same function like "serweb" in ser? 2. How to install the … [View More]

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R: R: [Users] using AVPs to write into database
by D'Addelfio Davide 28 Aug '06

28 Aug '06

Thanks, I've looked into but it works only with 1.1.x version...!!! What's the different with avp_db_store command, may I use it to store something in my SQL DB? I'll try to do but I have some problems, he doesn't register... I have another question: I need to work over sdp messages, is possible to do a parsing, always using the scripts language...? Any thought? Thanks Davide -----Messaggio originale----- Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Inviato: mercoledì 5 … [View More]aprile 2006 19.15 A: D'Addelfio Davide Cc: Jayesh Nambiar; Bogdan-Andrei Iancu; openser Oggetto: Re: R: [Users] using AVPs to write into database Hello, On 04/05/06 18:01, D'Addelfio Davide wrote: > > Hi guys, I have a similar problem, maybe easier. I'm new in openser so > I'd like to have some help. > > I'd like openser create a table in MySQL db when receive an INVITE, a > row with from uri, method and something else, I've look into modules > docs but I've not understanding at all...could you help me, please? > if you are using openser development version, then take a look at avp_db_query() in avpops module http://openser.org/docs/modules/1.1.x/avpops.html#AEN239 Cheers, Daniel > Thanks for reply > > Davide > > ------------------------------------------------------------------------ > > *Da:* users-bounces(a)openser.org [mailto:users-bounces@openser.org] > *Per conto di *Jayesh Nambiar > *Inviato:* mercoledì 5 aprile 2006 15.53 > *A:* Bogdan-Andrei Iancu > *Cc:* openser > *Oggetto:* Re: [Users] using AVPs to write into database > > Hi Bogdan, > > Thanks for the reply and sorry for sending the earlier mail 4 times as > yahoo was not responding peoperly. > > When I do an avp_write(), why does the avp name get stored in the uuid > column? > > I am not using any uuid as of now. I also tried to add the username of > the person who is calling but it again creates two rows, one with the > value as > > username and other row with the value as forwarded number. > > I thought if there is a way to insert the username also, it wud be > easier to check, if the incoming call for that user is to be forwarded > or not. > > I get the following in my usr_preferences table after I dial 86 > followed by any number. > > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > | uuid | username | domain | attribute | type | value | modified | > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > | s:callfwd | | | callfwd | 0 | sip:5515551478@202.80.61.10 | > 2006-04-05 19:10:49 | > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > > The script snippet is as follows: > > if(uri=~"^sip:86[0-9]*@") { > > > strip(2); > avp_write("$ruri", "s:callfwd"); > #avp_write("$from/username", "i:999"); > #avp_db_store("i:999", "i:/usr_preferences"); > avp_db_store("s:callfwd", "s:callfwd/usr_preferences"); > sl_send_reply("200", "OK"); > exit; > }; > > > */Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>/* wrote: > > Hi, > > you need two steps: > 1) write the ruri (after strip) into an AVP - use avp_write() > 2) write the avp into the db - use avp_db_store() > > regards, > bogdan > > Jayesh Nambiar wrote: > > > Hi all, > > I am trying to implement the call forwarding feature in openser. The > > forwarding part is fine. But I have the follwing scenario: > > Whenever a user needs to change his forwarding number, he should be > > able to do so from his end device only. > > For eg: He first presses 86 and then the 10 digit number to be > > forwarded. Here the 86 has to be stripped and the 10 digit number > > should be inserted into the call forward value column of the > > preferences table. > > Also if he wants to remove the forwarded number, he can do so by > > pressing say for eg: 87. on receiving this number openser should > > delete that value from the > > table. > > Is this possible. I tried to do the following, but somehow it > does not > > change the > > value. > > if(uri=~"^sip:86[0-9]*@") { > > if(avp_db_load("$from/username", "s:callfwd")) { #check > > if call-fwd feature is enabled for the user > > log(1,"AVP condition returned true"); > > strip(2); > > avp_write("$ruri", "s:callfwd"); > > avp_print(); > > log(1,"AVP written"); > > sl_send_reply("200", "OK"); > > exit; > > }; > > }; > > Is avp_write the proper method or I guess avp_db_store can also help > > me. The avp_print() function also does not show me anything in the > > log. Are there any > > logical mistakes or I have mis-interpreted the syntax of avpops > functions. > > Please help me in thsi regard. > > Thanks a lot in advance. > > Jayesh. > > > > ------------------------------------------------------------------------ > > Jiyo cricket on Yahoo! India cricket > > > > Yahoo! Messenger Mobile > > > > Stay in touch with your buddies all the time. > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > ------------------------------------------------------------------------ > > Jiyo cricket on Yahoo! India cricket > <http://us.rd.yahoo.com/mail/in/mailcricket/*http:/in.sports.yahoo.com/crick…> > Yahoo! Messenger Mobile > <http://us.rd.yahoo.com/mail/in/mailmobilemessenger/*http:/in.mobile.yahoo.c…> > Stay in touch with your buddies all the time. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > [View Less]

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[Serusers] SER with VoiceMail Problem
by ram 28 Aug '06

28 Aug '06

Hi i am new to ser, iam going through the list, i found some of the similar queries http://lists.iptel.org/pipermail/serusers/2003-October/003132.html I have used the config file changed according to my setup and iam able to pick up messages from Asterisk also But i have some problem here is 1. i have registered user 8888 in my SER, when i dial 8888 its going to voice mail, how can i set the caller==calee Busy even i have added this if (!uri==myself) { t_relay(); … [View More]

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[Serusers] Proxy Authentication
by Javier Oviedo 28 Aug '06

28 Aug '06

Hi all In my network schema I have a Sip Express Router to authenticate my outbound calls to a PSTN. I add a new equipment to a ser with "serctl add user pass e-mail" command. My softphone is registered in proxy but when I make a call to a PSTN I obtain the following error: *Call rejected: 407 Proxy Authentication Rquired *: I use the SJPhone software. Any idea about where is the problem Thanks in advance Joe My ser configuration is: debug=3 fork=no log_stderror=yes port=5060 children=4 … [View More] dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("localhost","subscriber")) { www_challenge("localhost","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; #if (!lookup("location")) { # sl_send_reply("404", "User Not Found"); # break; #}; route(1); } SoftPhone LOG: 17:04:43.1 SENDING TO: 10.110.0.127:5060 INVITE sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 1 INVITE Contact: <sip:313813@172.25.97.138:9528> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 274 v=0 o=- 32759240 32759303 IN IP4 172.25.97.138 s=eyeBeam c=IN IP4 172.25.97.138 t=0 0 m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 407 Proxy Authentication Required To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="localhost", nonce="44f307373cf9fb29da3028442e23f976513fc91d" Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=172.25.97.138 req_src_port=9528 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==1" 17:04:43.1 SENDING TO: 10.110.0.127:5060 ACK sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 1 ACK Content-Length: 0 17:04:43.1 SENDING TO: 10.110.0.127:5060 INVITE sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Contact: <sip:313813@172.25.97.138:9528> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 274 v=0 o=- 32759240 32759303 IN IP4 172.25.97.138 s=eyeBeam c=IN IP4 172.25.97.138 t=0 0 m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 100 trying -- your call is important to us To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=172.25.97.138 req_src_port=9528 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==1" 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 407 Proxy Authentication Required To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Proxy-Authenticate: Digest realm="localhost", nonce="44f307373cf9fb29da3028442e23f976513fc91d" Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=10.110.0.127 req_src_port=5060 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==2" 17:04:43.1 SENDING TO: 10.110.0.127:5060 ACK sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 2 ACK Content-Length: 0 17:04:43.2 Call (l:'Joe' r:'sip:313812@sipserver.es') - Call being terminated. Reasons: "Proxy Authentication Required", (code: 407) [View Less]

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Re: [Serusers] rewrite the From part of SIP INVITE messages
by rcarvalho＠iric.up.pt 28 Aug '06

28 Aug '06

I'm sceptic about the use of functions that replace parts of SIP messages, because that way, will Ser still keep record of transactions? Wont those replacements affect Ser to recognise to whom route subsequent messages? Regards, Ricardo. Alexandr Dubovikov wrote: > On Fri, Aug 25, 2006 at 05:33:52PM +0100, Ricardo Carvalho wrote: >> I need to rewrite the From part of SIP INVITE messages before sending >> them to my IP Telco, or else they wont accept my calls! >> … [View More]

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[Users] An error with exec_msg
by Zenone 28 Aug '06

28 Aug '06

Hi! I have an error when I try to run this in my openser.cfg: exec_msg("TAG=$(echo $SIP_HF_FROM|cut -f2 -d=);FROM=$(echo $SIP_HF_FROM|cut -f2 -d\<|cut -f1 -d\>|cut -f1-2 -d:);/usr/local/openser-1.1.0-tls/uri-folder/record $FROM $SIP_USER $TAG"); The obtained error is: Aug 28 15:11:21 localhost /usr/local/sbin/openser[32748]: ERROR:xl_parse_spec: bad pvar name "echo $SIP_HF_FROM|cut -f2 -d=);FROM=$(echo $SIP_HF_FROM|cut -f2 -d\<|cut -f1 -d\>|cut -f1-2 -d:);/usr/local/openser-1.1.0-… [View More]

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Re: [Serusers] serweb gives a blank page
by Andy Thomas 28 Aug '06

28 Aug '06

Andrey, Thanks for the pointer - I turned on PHP logging and found the problem- PHP Fatal error: Smarty error: unable to write to $compile_dir '/tmp/serweb'. Be sure $compile_dir is writable by the web server user. in /home/serweb/smarty/Smarty.class.php on line 1102 Down to permissions on the /tmp/serweb directory - all now working, thanks!!

2 1

[Users] Please help me in Openser Admin
by Ferianto siregar 28 Aug '06

28 Aug '06

Dear all, Thanks to all of you for your kind-hearted to give me solution in solving my problem. Thank you very much. Thanks to openser developer team, who has released Openser Administrator in sourceforge.net I have some questions about the openser admin. I do anybody can help me to undestand what the openser admin use for and how to install it. These are my question: 1. Is openser adminstrator a web application? I mean, does it have same function like "serweb" in ser? 2. How to install the … [View More]

2 1

[Serusers] REGISTRATION high pace & build_contact() error
by Kostas Marneris 28 Aug '06

28 Aug '06

Hi all, using sipsak in usrloc mode to test the SER sip REGISTRAR with the following process : loop: ./sipsak -U -p host.foo.com -x 3600 -a passwd -s sip:user@foo.com goto loop I get the following mesg in LOG file after 2 mins : build_contact(): No memory left ... and after a while : ERROR: build_res_buf_from_sip_req: out of memory ; needs 28132 send_reply(): Error while sending 200 OK ... CRITICAL ERROR : Can't save Contact to MySQL Location DB (this is a log mesg in my ser.cfg in case : … [View More]

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[Serusers] serweb gives a blank page
by Andy Thomas 28 Aug '06

28 Aug '06

I have installed Serweb, but when I run it I just get a blank page, with just the serweb title at the top. Nothing else appears, no login box or anything! I have set PEAR LOG to debug mode and the log only shows this- Aug 28 12:04:37 serweb [info] Useing file from default domain for filename: separator.html Aug 28 12:04:40 serweb [info] Useing file from default domain for filename: styles.css Aug 28 12:04:40 serweb [info] Useing file from default domain for filename: prolog.html Aug … [View More]

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[Users] Problem with SQL queries in 1.1.0?
by Federico Giannici 28 Aug '06

28 Aug '06

We recently upgraded from version 1.0.0 to 1.1.0 from CVS of a few days ago. After a couple of days of uptime, some SQL errors began to appear. First there was this one: Aug 21 10:07:07 eowyn OpenSER[19895]: ERROR: warning_builder: buffer size exceeded Aug 21 10:07:07 eowyn OpenSER[19895]: WARNING: warning skipped -- too big Aug 21 10:07:07 eowyn OpenSER[19895]: submit_query: Aug 21 10:07:07 eowyn OpenSER[19895]: db_insert: Error while submitting query Aug 21 10:07:07 eowyn OpenSER[19895]: … [View More]

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[Serusers] Multi process RTP Proxy
by Ramu Yadav 28 Aug '06

28 Aug '06

Hi All, With single process RTP Proxy we are facing CPU issues. When we made around 80 or above odd calls, rtpproxy is taking CPU(50%), and flow of RTP packets delivering some what late to the UA. Because of this we are facing some sound quality issues. To get rid of this CPU usage, we are planning to write rtpproxy as a multi process. Is this a good idea to write rtpproxy as multi process or is there any work around to improve the performance of the rtpproxy. Help and Ideas are highly … [View More]

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[Serusers] Granstream AtA g729 codec --- help please
by ravi reddy 28 Aug '06

28 Aug '06

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 … [View More]

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[Serusers] ivr play when the Account Disabled
by ram 27 Aug '06

27 Aug '06

Hi all I am using SER, and testing in my setup i have few issues , and looking forward for the suggestion iam running over night script, check the balance if the balance go to Nagetive, iam deactivating the user Like removing the Grants of LD and INT they can make local call. How can i run IVR for the negative balance user, when they dial to Long Distance Balance not there message. and how can i setup IVR program to user to check the balance of the account. any suggestions. ram

3 2

[Serusers] multidomain with loose routing
by Weiter Leiter 27 Aug '06

27 Aug '06

Some UAs, when not explicitly configured with an outbound proxy (some even in this), follow rfc's recommendation of sending a request with a preloaded route set containing one entry, that of what they think it is their outbound proxy, determined through various means. This is a problem when ser as a serving proxy for multiple domains does the loose_route'ing: if the route entry in the message is an ip address, ser will realise that it is him that must serve the request and everything goes … [View More]

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[Serusers] forward_tcp reliability
by Nhadie 27 Aug '06

27 Aug '06

Hi All, How reliable is forward_tcp? I have 3 SIP servers, 1 acting as the database of the three, I used forward tcp to replicate registration, unfortunately there are some instance that a local extension cannot contact another extension. Here's a part of my config if ((src_ip == 10.10.10.149) || (src_ip == 10.10.10.151)) { save_memory("location"); }; if (!www_authorize("sipdomain.com", "subscriber")) { www_challenge("sipdomain.com", "0"); break; }; if (!save("location")) { sl_reply_error();… [View More]

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RE: [Serusers] How to fix IP address stored in location db
by İlker Aktuna (Koç.net) 27 Aug '06

27 Aug '06

Hi, Thanks for your email. I get a "page not found" error when I try http://iptel.org/ser/gettingstarted <http://iptel.org/ser/gettingstarted> Actually at which URL is the example configuration script for using fix_nated_register() ? Also, which script should I edit to be able to use ser_ctl ? When I edit the serctl.conf file no change happens... Regards, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Friday, August 25, 2006 5:28 … [View More]PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to fix IP address stored in location db http://iptel.org/ser/gettingstarted Download the document and scripts for how to handle NATed clients. If you change default settings, you need to edit the script. g-) İlker Aktuna (Koç.net) wrote: Hi, Thanks for your reply. However, I couldn't understand it. What does exactly fix_nat_register do ? I have a client which does not have STUN feature and thus I can't make a call to this client. What should I do to make a call to this client ? Btw, I get the following errors when I use ser_ctl command: ser_ctl: (111, 'Connection refused'): error And WARNING: ListMethods on 'http://localhost:5060/' failed: socket.error My ser is running on 5061 port. Thanks, ilker -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Friday, August 25, 2006 12:10 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to fix IP address stored in location db The stored contact should be as received. You need to have the correct request URI when forwarding to a UA, even if the ip address is a private one. If not, the UA will most likely respond with 404 User not found. fix_nated_register() will put received ip:port in an avp and save will store this avp as received (you will see it with serctl ul show) g-) İlker Aktuna (Koç.net) wrote: Hi, I am trying to change stored uri in the location db. I assume it could be done with the help of nathelper module and I use fix_nated_register() function just before save_noreply("location") command. However I still see the internal IP address of the client in the location db. What is the easiest way to change the stored contact address ? Thanks, ilker __________ _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________ [View Less]

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[Serusers] (no subject)
by rcarvalho＠iric.up.pt 26 Aug '06

26 Aug '06

I'm sceptic about the use of functions that replace parts of SIP messages, because that way, will Ser still keep record of transactions? Wont those replacements affect Ser to recognise to whom route subsequent messages? Regards, Ricardo. Alexandr Dubovikov wrote: > On Fri, Aug 25, 2006 at 05:33:52PM +0100, Ricardo Carvalho wrote: >> I need to rewrite the From part of SIP INVITE messages before sending >> them to my IP Telco, or else they wont accept my calls! >> … [View More]

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RE: [Users] openser admin
by peter.3.edwards＠bt.com 26 Aug '06

26 Aug '06

Mike, What's the upgrade procedure to go from 0.2 -> 0.3? Presumably readd the phplib_id column as per your mail on 21 Aug ... and, what - just replace the old openseradmin-0.2 directory with this one (including creating a new database.yml file)? Also, is there a way to uninstall OpenSER Administrator without effecting OpenSER ... Presumably, just delete the openseradmin-x directory ... But is there an equivalent SQL script to uninstall it's database component? I think it would be good … [View More]

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[Users] Error in running serweb. Please help me..Please
by Ferianto siregar 26 Aug '06

26 Aug '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 "… [View More]

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[Serusers] mysql modules load problems
by ram 26 Aug '06

26 Aug '06

Hi all iam install in new server RHEL 4.0 with Mysql 5.0 ser-0.97-pre compiled perfectly and installed perfectly when i enable mysql modules in ser.cfg and try to run [root@vmail ser]# ser -c -f ser.cfg 0(10560) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log 0(10560) parse error (28,13-14): failed to load module ERROR: bad config file (1 errors) what is the problem here Ram

2 1

[Serusers] rewrite the From part of SIP INVITE messages
by Ricardo Carvalho 25 Aug '06

25 Aug '06

I need to rewrite the From part of SIP INVITE messages before sending them to my IP Telco, or else they wont accept my calls! Anyone knows how can I do that? Thanks, Ricardo.

2 1

RE: [Serusers] How to fix IP address stored in location db
by İlker Aktuna (Koç.net) 25 Aug '06

25 Aug '06

Hi, Thanks for your reply. However, I couldn't understand it. What does exactly fix_nat_register do ? I have a client which does not have STUN feature and thus I can't make a call to this client. What should I do to make a call to this client ? Btw, I get the following errors when I use ser_ctl command: ser_ctl: (111, 'Connection refused'): error And WARNING: ListMethods on 'http://localhost:5060/' failed: socket.error My ser is running on 5061 port. Thanks, ilker -----Original Message-----… [View More] From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Friday, August 25, 2006 12:10 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to fix IP address stored in location db The stored contact should be as received. You need to have the correct request URI when forwarding to a UA, even if the ip address is a private one. If not, the UA will most likely respond with 404 User not found. fix_nated_register() will put received ip:port in an avp and save will store this avp as received (you will see it with serctl ul show) g-) İlker Aktuna (Koç.net) wrote: > Hi, > > I am trying to change stored uri in the location db. > I assume it could be done with the help of nathelper module and I use fix_nated_register() function just before save_noreply("location") command. > However I still see the internal IP address of the client in the location db. > > What is the easiest way to change the stored contact address ? > > Thanks, > ilker > __________ _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________ [View Less]

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[Users] Openser + Radius + Mysql: Acconting - AcctStopTime doesn't appear in the table radacct (radius DB)
by Alberto Pacifico 25 Aug '06

25 Aug '06

Hi all, I have problem with Accounting in radius database. The parameter AcctStopTime of the table named radacct isn't filled by Openser. The parameter AcctStartTime is okay. In my code of Openser I'm using: ... modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) # set the reporting log level modparam("acc", "log_level", 1) # number of the flag which will be used to mark messages for accounting modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) … [View More]

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Re: [Users] force_rtp_proxy() and m=video
by Atle Samuelsen 25 Aug '06

25 Aug '06

HI Adam, The reason for this is 2 reasons.. the nathelper module does not understand the port/1 parameter (wich says "use 2 ports, port 5010 and 5011) and it does also not understand the RTP/SAVPF. If you fix these 2 things, this should work :-) (Is this a Ojo?) -A In-Reply-To: <029301c6c82a$68a8c370$1601a8c0@oralnetws2> * Adam Linford <adam.linford(a)oralnet.co.uk> [060825 11:40]: > I've been happily using the force_rtp_proxy() function, exported from the > nathelper … [View More]

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[Serusers] About Cisco 7960 an option Call Forwarding All
by Javier Oviedo 25 Aug '06

25 Aug '06

Hi all In my SIP environment I have several Cisco 7960. These telephones have the option "Call Forwarding All", when I use this option with ser, the telephone generete a 302 SIP Code response, but the sip proxy does not forward the call to the new destiny. Any Idea to solve this problems Thanks in advance Joe

1 0

[Users] need to find out OUTGOING IF / SOCKET
by Gerry 25 Aug '06

25 Aug '06

Dear openser-users, I am trying to set-up a configuration where openser + RTPproxy are relaying messages between disconnected networks (i.e. openser + RTPproxy having more than one network-interface). With this method, I could additionally bridge IPv4-only and IPv6-only clients. Therefore I need to find out, on which interface (or socket, respectively) openser will send the message out because RTPproxy needs this info and is too dumb to find out by itself. I currently have a _very_ … [View More]

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[Users] force_rtp_proxy() and m=video
by Adam Linford 25 Aug '06

25 Aug '06

I've been happily using the force_rtp_proxy() function, exported from the nathelper module for some time now. Recently, I've added some videophone UA's to our system, and we are finding that rtpproxy is only relaying the audio for sessions between video phones. Having ran some SIP traces I've noticed that the SDP is not being correctly rewritten. Whilst the contact header for the audio is being allocated a port from rtpproxy's range correctly, the corresponding m= field for the video is not … [View More]

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[Serusers] Asterisk t38passthrough
by Ricardo Carvalho 25 Aug '06

25 Aug '06

Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax machine. Every time I send a fax, it gets sent using codec G711, and never T.38. I added the following parameters in the [general] section as well as in device configurations: t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes I think that's the only thing that is needed to do to enable T.38 pass through... Why does Asterisk keeps sending in G711? Any help? Regards, Ricardo.

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[Serusers] How to fix IP address stored in location db
by İlker Aktuna (Koç.net) 25 Aug '06

25 Aug '06

Hi, I am trying to change stored uri in the location db. I assume it could be done with the help of nathelper module and I use fix_nated_register() function just before save_noreply("location") command. However I still see the internal IP address of the client in the location db. What is the easiest way to change the stored contact address ? Thanks, ilker … [View More]

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Re: [Serusers] Voice Mail problem
by Greger V. Teigre 25 Aug '06

25 Aug '06

Please copy serusers. It's your failure route you should look at! g-) ram wrote: > Hi > > iam following the same document only to make my voice mail work > below is my config, its works if the user is registered locally > and if not available it will moving to voicemail > > But when iam dialing PSTN still its going to voicemail > > Aug 25 11:05:19 cdrouter ser[9337]: **************** vm start - begin > ****************** > Aug 25 11:05:19 cdrouter Sems[… [View More]802]: Error: 404 voicemail: no email > address for user <1XXXXXX360> > Aug 25 11:05:19 cdrouter ser[9348]: ACC: call missed: from=99999 < > sip:99999@domain.com <mailto:sip:99999@domain.com>>;tag=2858293102, > i-uri=sip:1XXXXXXX360@domain.com <mailto:sip:1XXXXXXX360@domain.com>, > method=INVITE, o-uri= sip:1XXXXXX360@domain.com > <mailto:sip:1XXXXXX360@domain.com>, code=404 Not Found > Aug 25 11:05:19 cdrouter ser[9337]: **************** vm start - end > ****************** > > > ------------ > if (!lookup("location")) { > > # Voicemail specific configuration - begin > > if(method=="ACK" || method=="INVITE" || > method=="BYE"){ > > setflag(2); > if(t_newtran()){ > > t_reply("100","Trying -- just wait a > minute !"); > > if(method=="INVITE"){ > log(1,"**************** vm > start - begin ******************\n"); > > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm start - end > ******************\n"); > break; > }; > > if(method=="BYE"){ > log(1,"**************** vm end - begin > ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm end - end > ******************\n"); > break; > }; > } > else { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new > transaction"); > }; > > > /* Not found*/ > sl_send_reply("404","Not Found"); > }; > # Voicemail specific configuration - end > }; > > > > > On 8/24/06, *Greger V. Teigre* <greger(a)teigre.com > <mailto:greger@teigre.com>> wrote: > > Look at the features-callfwd.cfg example from onsip.org > <http://onsip.org/>, now available here: > http://www.iptel.org/ser/doc/gettingstarted > g-) > > ram wrote: > Hi > > I have existing SER Running > > i have local users ( extensions) and DID ( all other countries) > > iam able to make calls in and out , there is no problem > > but when the user not available, it has to send to Voice mail, > but its only sending the users > who are locally start with 9*, > > how can i send the user start with 9 or any DID not availble > send to voicemail > > here is my config > > > if (!lookup("location") && > ( uri=~"^sip:9.*@domain.com <mailto:9.*@domain.com>" ) > ) { > > # Voicemail specific configuration - begin > > if(method=="ACK" || method=="INVITE" || > method=="BYE"){ > > setflag(2); > if(t_newtran()){ > > t_reply("100","Trying -- just wait > a minute !"); > > if(method=="INVITE"){ > log(1,"**************** vm > start - begin ******************\n"); > > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm start - > end ******************\n"); > break; > }; > > if(method=="BYE"){ > log(1,"**************** vm end - > begin ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm end - > end ******************\n"); > break; > }; > } > else { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new > transaction"); > }; > > break; > }; > break; > }; > > > > any help will be great > > > > Ram > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > > [View Less]

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[Users] Internet Providers changing the SIP headers
by Bruno Machado 25 Aug '06

25 Aug '06

Hi fellows. We are in trouble with this "deceitful" error. I mean, errors. Look at RURI. Our proxy receive this "thing". We have a number there! Sometimes, this number is shown within the word REGISTER, like that: "REGI:34846:STER". We can rewrite the URI, but Openser doesnt understand the method "REGI:34846:STER". Second error: Via header. The REGISTER is showing the NAT_DEVICE_IP instead UA_IP. And this VoIP isnt behind any VoIP device! So, "who" are changing it? "Who" can does it? Is … [View More]

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[Users] How to configure openser as a redirect server
by Padmaja RV 25 Aug '06

25 Aug '06

Hi all, I need to implement Openser as a redirect server for all the 3XX class of responses. I have checked the redirect.cfg example in the cvs and it includes only for the 300 response. The other two like 302 moved temporarily or 301 moved permanently are not included anywhere. Can anybody send me the script for the same? Thank You, Padmaja

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[Serusers] maximum number of routes?
by Jean-Michel Hiver 25 Aug '06

25 Aug '06

Hi, I have found that ser yields an error when using up to 20 routes (i.e. route[20] block) and openser 40 routes. Is there an option to increase the number of routes ser can handle? Cheers, Jean-Michel.

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[Serusers] avpops & usr_preferences -- multiple username/domain/attr sets
by O'Shaughnessy Evans 25 Aug '06

25 Aug '06

Hello. I'm trying to implement serial forking with the avpops module, but I've run into a small stumbling point. If I want to store more than 1 forwarding contact in the usr_preferences table, I can't because the table as originally defined from the ser codebase sets up (username, domain, attribute) as a primary key. To get around this, is it normal to use a different table that doesn't require that tuple to be unique, or is it recommended to change the definition of the usr_preferences … [View More]

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[Users] NAT traversal with mediaproxy on natted private address
by Simon Morvan 25 Aug '06

25 Aug '06

+----------+ +-----+ +------+ |OpenSER | +------+ |Phone| |Router| |MediaProxy| |Router| +-----+ +------+ +----------+ +------+ | | | | | | --------------------- -------------------------- -------------- Internet Private LAN A Private LAN B Here is the situation : SIP phones are in a private LAN A. They can speak with the server hosting openser and mediaproxy which is located on … [View More]

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[Users] Call a stored procedure
by Miroslaw Nawrot 24 Aug '06

24 Aug '06

Hi, Does someone have an axample of using avp_db_query() with a mysql stored procedure? I keep getting an error "submit_query: PROCEDURE openser.sp_getrouteinfo can't return a result set in the given context" whenever I pass a stored procedure call to that functions. Thanks. Mirek N Globalive Communications Corp. The Future of Voice T:416-640-1088 F:416-640-1089 60 Adelaide St E, 6th Floor Toronto, ON, M5C 3E4 www.globalive.com

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[Users] openser admin
by Mike Williams 24 Aug '06

24 Aug '06

All, OpenSER Admin version 0.3 has been released. This fixes the bug where compatibility with openserctl was broken, and adds the ability to add other management accounts to the interface. Let me know what you think. Thanks, Mike Williams

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[Users] How to change domain when REGISTERing UA?
by Dmitry Lyubimkov 24 Aug '06

24 Aug '06

Hi! Description of the problem: The some UA can send only IP addresses as domains. Some subscribers incorrectly configure their UA and as a result as the domain IP of SIP Proxy comes. In result in the table location the part of subscribers has the symbolical domain (voapp.ru) and part - IP address of SIP Proxy (62.33.22.14) It leads to to the certain complexities in switching incming calls. The problem disappears if to tell: modparam("usrloc", "use_domain", 0) modparam("registrar", "… [View More]use_domain", 0) And use only names without domains. But in our case it does not approach since we have some independent domains. I tried to rewrite fields From and To route(8); save("location"); route[8] { #correct From and To in REGISTER messages with IP address instead of domain name if (search("From:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+.*") or search("To:.*@[0-9+\.[0-9]+\.[0-9]+\.[0-9]+.*")) { subst("/^From:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/Fr om:\1sip:\2@voapp.ru\7/i"); subst("/^To:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/To:\ 1sip:\2@voapp.ru\7/i"); xlog("L_ERR", "REGISTER From To done\n"); return(-1); }; return(1); } Fields varied but function save took their not changed values from initial message. After that I have tried to send REGISTER message once again to myself for processing: if (!route(8)) { t_relay(); } else { save("location"); }; In this case in the location table there was a record with the corrected name of the domain. But in Contact field there was an address of SIP Proxy server instead of address of UA. Can anybody prompt idea how to to change domain in From To fields at reception of REGISTER message? Tnx. How to change domain when REGISTERing UA? Hi! Description of the problem: The some UA can send only IP addresses as domains. Some subscribers incorrectly configure their UA and as a result as the domain IP of SIP Proxy comes. In result in the table location the part of subscribers has the symbolical domain (voapp.ru) and part - IP address of SIP Proxy (62.33.22.14) It leads to to the certain complexities in switching incming calls. The problem disappears if to tell: modparam("usrloc", "use_domain", 0) modparam("registrar", "use_domain", 0) And use only names without domains. But in our case it does not approach since we have some independent domains. I tried to rewrite fields From and To route(8); save("location"); route[8] { #correct From and To in REGISTER messages with IP address instead of domain name if (search("From:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+.*") or search("To:.*@[0-9+\.[0-9]+\.[0-9]+\.[0-9]+.*")) { subst("/^From:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/Fr om:\1sip:\2@voapp.ru\7/i"); subst("/^To:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/To:\ 1sip:\2@voapp.ru\7/i"); xlog("L_ERR", "REGISTER From To done\n"); return(-1); }; return(1); } Fields varied but function save took their not changed values from initial message. After that I have tried to send REGISTER message once again to myself for processing: if (!route(8)) { t_relay(); } else { save("location"); }; In this case in the location table there was a record with the corrected name of the domain. But in Contact field there was an address of SIP Proxy server instead of address of UA. Can anybody prompt idea how to to change domain in From To fields at reception of REGISTER message? Tnx. Dmitry [View Less]

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RE: [Serusers] Alternate port for rewritehost
by Stephen Helfen 24 Aug '06

24 Aug '06

Thanks...found it 2 seconds after sending the email! :PDate: Thu, 24 Aug 2006 21:21:10 +0200From: greger(a)teigre.comTo: steve_helfen(a)hotmail.comCC: serusers(a)iptel.orgSubject: Re: [Serusers] Alternate port for rewritehost Use rewritehostport("domain:5060") g-) Stephen Helfen wrote: All, I am receiving SIP traffic on a non-5060 port but need to change the port of the rewritehost to 5060 as the carrier is dropping my INVITE messages with the non-5060 port. Any suggestions? … [View More]

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[Serusers] SER and RTPproxy?!
by Caxton 24 Aug '06

24 Aug '06

Hi all: The document metion that "RTPproxy will not be installed on the SER server, but on a remote machine." If I install SER on "192.168.1.3", and install RTPproxy on "192.168.1.5". How do they work? Now, I install they on a same machine, and I run SER first, then usr command "rtpproxy -l xxx.xxx.xxx.xxx" to force rtpproxy. The SIP client can register behind NAT, but there are some problem with RTP stream. I found that RTP strem is able to relay to this machine, but this machine … [View More]

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[Serusers] Alternate port for rewritehost
by Stephen Helfen 24 Aug '06

24 Aug '06

All,I am receiving SIP traffic on a non-5060 port but need to change the port of the rewritehost to 5060 as the carrier is dropping my INVITE messages with the non-5060 port. Any suggestions?Thanks,Steve _________________________________________________________________ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://get.live.com/toolbar/overview

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[Users] Please help for guiding serweb installation in openser server
by Ferianto siregar 24 Aug '06

24 Aug '06

Dear all, Thank you very much to all of you for help me to answer my question about serweb before. So, based on your suggestion, I tried to install serweb 0.9.3 for my openser server. But, when I tried to install it, I am confused. I don`t understand about the installation note that I have got from untar the serweb 0.9.3 packet. These are some questions that I need to ask: 1. Where should we untar the serweb 0.9.3 packet? Should we untar it in root or anywhere we like? Because when we untar … [View More]

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[Users] NAT (again)
by Mark Kent 24 Aug '06

24 Aug '06

I'm testing my rtpproxy config with a linksys phone and a grandstream phone, either not NAT'ed, or behind a cisco (full cone), or behind a Netgear (symmetric). Things generally work, but I have a problem with the grandstream behind the symmetric nat. This is the register: U NAT-IP:3284 -> PROXY-IP:5060 REGISTER sip:sip.example.com SIP/2.0 Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK0103d0ff79cc9c02 From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=… [View More]743335557afcade4 To: <sip:frick@sip.example.com;user=phone> Contact: <sip:frick@NAT-IP:3280;user=phone> Authorization: Digest username="frick", realm="sip.example.com", algorithm=MD5, uri="sip:sip.example.com", nonce="foo", response="bar" Call-ID: 11f6defb0857db17(a)192.168.0.100 CSeq: 116 REGISTER Expires: 60 User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 This is the resulting contact: ...Record(0x2864b680).. domain: 'location' aor : 'frick(a)sip.example.com' ~~~Contact(0x2864d4e0)~~~ domain : 'location' aor : 'frick(a)sip.example.com' Contact : 'sip:frick@NAT-IP:3280;user=phone' Expires : 34 q : Call-ID : '11f6defb0857db17(a)192.168.0.100' CSeq : 119 User-Agent: 'Grandstream BT100 1.0.6.7' received : 'sip:NAT-IP:3284' Path : '' State : CS_SYNC Flags : 1 Sock : PROXY-IP:5060 (0x8122f50) Methods : 5695 next : 0x0 prev : 0x0 ~~~/Contact~~~~ .../Record.. Now, what I don't understand is the two ports used above: 3280 and 3284. It looks like the phone thinks it is being natted to port 3280, but it's really getting port 3284 When I make a call I get U NAT-IP:3284 -> PROXY-IP:5060 INVITE sip:16509415674@sip.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK4a4c039fbcb5bef8 From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=2dc2fc3c94e93951 To: <sip:16509415674@sip.example.com;user=phone> Contact: <sip:frick@NAT-IP:3280;user=phone> Supported: replaces Call-ID: 1f04f9c6ec474f70(a)192.168.0.100 CSeq: 26381 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 419 v=0 o=frick 8000 8002 IN IP4 NAT-IP s=SIP Call c=IN IP4 NAT-IP t=0 0 m=audio 3295 RTP/AVP 0 8 4 18 2 15 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 and my side sends back the Proxy Authentication Required, but it sends it to port 3280. Now, is that based on the Via: header or the Contact: header, or (unlikely) the stored contact info from the REGISTER? The phone acts like it never got it (and I'm guessing it didn't) and sends the INVITE again without any auth data. The thing that bugs me is that this is an outbound call, from the phone out through the NAT box, I figured that would be doable. The Linksys942 works, so I'm hoping this is a Grandstream bug, or something I can fix in my openser config. Thanks, -mark [View Less]

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[Users] Limiting Call Volume and CPS
by Brian Jones 24 Aug '06

24 Aug '06

Hi, I was wondering if anyone had any suggestions on how I could use OpenSER to limit call volume and call per second on a peer basis. Thanks! Brian.

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[Users] Weird forwarding with Route header
by Jeremie Le Hen 24 Aug '06

24 Aug '06

Hi list, my proxy (yy.yy.yy.yy) receives a message like this: BYE sip:12345678@xx.xx.xx.xx SIP/2.0 Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on> My ruleset is something like this (with the REGISTER logic pulled out) : %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% route { if (!mf_process_maxfwd_header("10")) { … [View More]

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[Users] dropping a 200
by Simon Morvan 24 Aug '06

24 Aug '06

Just to be sure : there's no way to call drop() in an onreply_route block when it's not a provisionnal reply ? -- Simon.

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[Users] how to correctly handle NAT on inbound call from another SIP-proxy?
by Gerd Feiner 24 Aug '06

24 Aug '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi list, we have some trouble handling NAT correctly when receiving a call from another SIP-proxy. the mediaproxy-module does not seem to rewrite SDP when it finds another proxy's record-route header, though it invokes mediaproxy and sends uncorrected SDP to the other SIP-server resulting in one-way-audio as the provided c=-field includes the RFC1918-address of the client. when receiving calls through our Cisco AS5350 everything is fine ('cause … [View More]

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[Users] Q:ABOUT:MODULE:auth_radius
by Alejandro Sanchez 24 Aug '06

24 Aug '06

Hello. openser 1.1 radiusclient-ng 5 freeradius 1.1.1 I need to enable the modules *_radius for my openser instalation a follow the guide for enable radius then when i try to start openser i get the following error. 0(0) ERROR: auth_radius: can't get code for the Digest-Response attribute Anyone knows what is the cause for this error? Thank's Alex. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b… [View More]

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[Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

Hi, Is there a way to coexist multiple domains in Ser, distinguishing users even if they have the same username in subscriber table but different domains? I'll explain better, I have subscriber table with: | username | domain | e-mail | ------------------------------------------------ | user_A | domain_A | user_A@domain_A | ------------------------------------------------ | user_A | domain_B | user_A@domain_B | and in domain table there are both … [View More]

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Re: [Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

It's true Asterisk can use different contexts, but that doesn't solve my problem because I use Ser + Asterisk and Ser is used for routing SIP messages between SIP UAs, not Asterisk. In my architecture, Asterisk is only used for advanced services like voicemail or call conferencing and for gateway, not to route messages between UAs. Ricardo. openser(a)comcast.net wrote: > The way to do that is intergrate Ser with Asterisk and in Asterisk you > can build context and keep the same … [View More]

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[Serusers] Voice Mail problem
by ram 24 Aug '06

24 Aug '06

Hi I have existing SER Running i have local users ( extensions) and DID ( all other countries) iam able to make calls in and out , there is no problem but when the user not available, it has to send to Voice mail, but its only sending the users who are locally start with 9*, how can i send the user start with 9 or any DID not availble send to voicemail here is my config if (!lookup("location") && ( uri=~"^sip:9.*@domain.com" ) ) { … [View More]

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[Serusers] Broken ser.cfg
by Jean-Michel Hiver 23 Aug '06

23 Aug '06

Can somebody please take a look at the routing logic below... I am trying to use exec_dset along with another program to do some LCR with SER but it doesn't work properly (here is a capture of what it produces: http://pastebin.ca/146771 Any ideas of what's doing wrong? Code: # -------------------------------------- route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { … [View More]

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[Serusers] Need config help for PSTN call routing blocks to service provider.
by Chahn John Kim 23 Aug '06

23 Aug '06

Hi, I am new to SER and need experts' help in PSTN call routing block to call termination service provider in a scenario below: We have our own sip realm/server that will handle all VoIP calls among our own subscribers; sip.ours.com. This will forward all PSTN calls, authenticated and authorized by it, to sip.theirs.com, which will authorize usning our IP address and forward to pstngateway.theirs.com. What I have been given by SP is: sip.theirs.com, IP, myuserID and password. What they … [View More]

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Re: [Users] options for rtpproxy.
by Bogdan-Andrei Iancu 23 Aug '06

23 Aug '06

Hi Bill, Bill Zhang wrote: >Anyone can explain following options: > >1. -2fv > > v - print the version f - stay in foreground, don't become a daemon 2 - duplicate mode >2. -s path > > command socket path >3. -L nfiles > > maximum file descriptor number that can be opened regards, bogdan >The rest of them seems to be straight forward. This rtproxy documentation >is HARD to find:(. > >usage: rtpproxy [-2fv] [-l addr1[/addr2]] [-6 addr1[/… [View More]

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Re: [Users] rtpproxy or RTPProxy, where to download or checkout--found it:). Re: Users Digest, Vol 15, Issue 61
by Mark Kent 23 Aug '06

23 Aug '06

In the RTPproxy source code, if you look at rtpp_log.h you really get the impression that syslog() functionality can be dropped in with ease. I still believe that, and believe it can be elegant, despite the fact that my solution is fairly crappy. But, it's done... so here it is (it relies on dprint.[ch] from the openser1.1 distribution): After ./configure, do this: In Makefile, change CFLAGS line like this: CFLAGS = -g -O2 -DUSESYSLOG -I../openser-1.1.0-tls (point to wherever you build … [View More]

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[Serusers] SER / Missing call records
by Darren Sessions 23 Aug '06

23 Aug '06

All, We're experiencing a problem with missing call records. There is nothing in the log files that indicate a problem either with SER or our database server and likewise there is nothing in the database server's logs to indicate a problem. System loads are within reason on both servers, network connectivity has been tested and verified, and a through hardware shakedown on both boxes has been completed without any errors. We first noticed this problem when our underlying vendor's bill (… [View More]based on minutes of course) did not correspond with our own. Upon doing a CDR comparison (based on what was in both the acc and calls tables), we were able to verify that the calls in dispute were indeed ours (based on the sip call-id's IP addresses). We're processing roughly 1.4 million calls per day on this SER server in question. Out of those calls, 508 calls (give or take a couple hundred) with a total duration of over 5,300 minutes - can not be accounted for in any of the SER database tables. We're running 0.9.5-pre1 with Suse Enterprise Server 9.0 (2.6.11 Kernel). Server is an HP DL380 with dual 3.2GHz processors with 6GB physical ram (only 1.4GB in use system wide); disk space is not an issue as well (only 10% in use). We've scoured the mailing lists and change logs for any hint of an issue similar to ours without any results. The closest we've been able to come that's anywhere near along the lines of our issue in any way that made any sense were these posts: http://lists.iptel.org/pipermail/serdev/2003-August/000602.html http://lists.iptel.org/pipermail/serusers/2005-February/015102.html At this point, we're not sure if this issue has something to do with the FIFO and SER making bulk dump uploads with call data to the database, or some other issue along those lines - or if this is a new issue all together. Any help would be greatly appreciated. Thanks, - Darren Fifo settings: fifo="/tmp/ser_fifo" fifo_db_url="mysql://xxx:xxx@xxx/serdb" Here are our module specific parameters: # -- timers -- modparam("tm", "wt_timer", 10) modparam("tm", "retr_timer1p1", 2) # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- database URL -- modparam("usrloc|acc|auth_db|group","db_url","mysql://xxx:xxx@xxx/serdb" ) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth","secret","xxx") # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- Nathelper -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) # -- acc params -- modparam("acc", "db_flag", 3) [View Less]

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[Serusers] Numbering Plan
by Daniel Mirrha Santos 23 Aug '06

23 Aug '06

Does anybody knows if I can create a numbering plan using a database? How is the easiest mode to do a numbering plan on ser? Thanks, Daniel

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Re: [Users] 403 Forbidden-----onsip configs testing. user not able to register
by Ferianto siregar 23 Aug '06

23 Aug '06

Hi, Ram its Not looking up in aliases table , it look ups only on location table. the each request and responses . So, You have to add your domain in the domain table by using : insert into domain values('urs_domain', ' date) ; in openser databases tables. I do hope it can help you. Thanks. On 8/22/06, ram wrote: > > Hi all > > iam trying to read the getting started document from onsip > when iam configuring pstn-gw.cfg > with the below config > … [View More]> when the X-lite user try to loging , SIP/2.0 403 Forbidden, in the Log of > X-lite > > what is wrong , even i have changed AVP after reading module configs in > openser. > > any one suggest me what is wrong iam doing here > > Ram > > ----------------- > openser.cfg > # > # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ > # > # simple quick-start config script > # > > # ----------- global configuration parameters ------------------------ > > debug=9 # debug level (cmd line: -dddddddddd) > fork=yes > log_stderror=no # (cmd line: -E) > > /* Uncomment these lines to enter debugging mode > #fork=no > #log_stderror=yes > */ > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > listen=myip > port=5060 > children=4 > fifo="/tmp/openser_fifo" > fifo_db_url="mysql://openser:openserrw@localhost /openser" > > > # ------------------ module loading > ---------------------------------- > > > loadmodule "/usr/local/lib/openser/modules/mysql.so" > loadmodule "/usr/local/lib/openser/modules/sl.so" > loadmodule "/usr/local/lib/openser/modules/tm.so" > loadmodule "/usr/local/lib/openser/modules/rr.so" > loadmodule "/usr/local/lib/openser/modules/maxfwd.so" > loadmodule "/usr/local/lib/openser/modules/usrloc.so" > loadmodule "/usr/local/lib/openser/modules/registrar.so" > loadmodule "/usr/local/lib/openser/modules/textops.so" > loadmodule "/usr/local/lib/openser/modules/auth.so" > loadmodule "/usr/local/lib/openser/modules/auth_db.so" > loadmodule "/usr/local/lib/openser/modules/uri.so" > loadmodule "/usr/local/lib/openser/modules/uri_db.so" > loadmodule "/usr/local/lib/openser/modules/domain.so" > loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/openser/modules/nathelper.so" > loadmodule "/usr/local/lib/openser/modules/avpops.so" > loadmodule "/usr/local/lib/openser/modules/permissions.so" > > modparam("auth_db|permissions|uri_db|usrloc","db_url", > "mysql://openser:openserrw@localhost/openser") > 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("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > > modparam("registrar", "nat_flag", 6) > > modparam("rr", "enable_full_lr", 1) > > modparam("tm", "fr_timer_avp", "i:25") > modparam("tm", "fr_inv_timer_avp", "i:25") > > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > > route { > > # > ----------------------------------------------------------------- > # Sanity Check Section > # > ----------------------------------------------------------------- > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483", "Too Many Hops"); > return; > }; > > if (msg:len > max_len) { > sl_send_reply("513", "Message Overflow"); > return; > }; > > # > ----------------------------------------------------------------- > # Record Route Section > # > ----------------------------------------------------------------- > if (method=="INVITE" && client_nat_test("3")) { > # INSERT YOUR IP ADDRESS HERE > > record_route_preset("myproxyip:5060;nat=yes"); > } else if (method!="REGISTER") { > record_route(); > }; > > # > ----------------------------------------------------------------- > # Call Tear Down Section > # > ----------------------------------------------------------------- > if (method=="BYE" || method=="CANCEL") { > end_media_session(); > }; > > # > ----------------------------------------------------------------- > # Loose Route Section > # > ----------------------------------------------------------------- > if (loose_route()) { > > if (has_totag() && (method=="INVITE" || method=="ACK")) { > > if > (client_nat_test("3")||search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > return; > }; > > # > ----------------------------------------------------------------- > # 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"); > }; > return; > }; > > if (method=="CANCEL") { > route(1); > return; > } else if (method=="INVITE") { > route(3); > return; > } else if (method=="REGISTER") { > route(2); > return; > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; > }; > > if (!lookup("location")) { > sl_send_reply("404", "User Not Found"); > return; > }; > > route(1); > } > > route[1] { > > # > ----------------------------------------------------------------- > # Default Message Handler > # > ----------------------------------------------------------------- > > t_on_reply("1"); > > if (!t_relay()) { > > if (method=="INVITE" || method=="ACK") { > end_media_session(); > }; > > sl_reply_error(); > }; > } > > route[2] { > > # > ----------------------------------------------------------------- > # REGISTER Message Handler > # > ----------------------------------------------------------------- > > sl_send_reply("100", "Trying"); > > if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { > setflag(6); > fix_nated_register(); > force_rport(); > }; > > if (!www_authorize("","subscriber")) { > www_challenge("","0"); > return; > }; > > if (!check_to()) { > sl_send_reply("401", "Unauthorized"); > return; > }; > > consume_credentials(); > > if (!save("location")) { > sl_reply_error(); > }; > } > > route[3] { > > # > ----------------------------------------------------------------- > # INVITE Message Handler > # > ----------------------------------------------------------------- > > if (!allow_trusted()) { > > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > return; > } else if (!check_from()) { > sl_send_reply("403", "Use From=ID"); > return; > }; > > consume_credentials(); > }; > > if (client_nat_test("3")) { > setflag(7); > force_rport(); > fix_nated_contact(); > }; > > if (uri=~"^sip:1[0-9]{10}@") { > strip(1); > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; === message truncated === --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. [View Less]

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[Users] Problem with forking
by Claudio Miceli 23 Aug '06

23 Aug '06

I am in a big trouble. I am using openser 1.0.1 and I am still having some problems with redirect. I get some numbers in myredirect and I try to pass it with t_relay(), but at this moment everything gets confused. The calls do not happen sequentially or in any logical order. Is there another way of doing the forking ? Am i doing something terribly wrong ? And sometimes when I make a call my openser sends messages to just one client and not to the other, or sends to both but with a great time … [View More]

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RE: [Users] registered("location")
by Helge Waastad 23 Aug '06

23 Aug '06

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[Users] Please help me for SMS module (SIP to GSM)
by ade darmawan 23 Aug '06

23 Aug '06

Please help me mr... i have read this module...but i don't understand with 4 question : 1. I found that in the SMS Module (SIP to GSM) don't explained the software which used by the client/user agent or where we should to do testing in client? 2. I wanna know the other facilities should be used, beside openser-1.0.0.tar.gz,mysql,DNS? 3. For the final result, is this module could send SMS by 2 directions? 4. Could you please the step to do the testing? Thank's for all -----------… [View More]

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[Users] Byes not accounted
by Jayesh Nambiar 23 Aug '06

23 Aug '06

Hi all, I have got a strange problem with my accounting. This is my scenario: I have a registered user dialling a pstn number. The call is connected and then the registered UA switches off the IP device and there is no BYE from the registered UA. Now the pstn gateway sends a BYE after certain period and Openser tries to send that BYE to the registered UA, but the registered UA is powered off and it never replies. So Openser retransmits the BYE messages and finally generates a 408 response … [View More]

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[Users] libmysqlclient15-dev Download Fail
by openser＠comcast.net 23 Aug '06

23 Aug '06

I am downloading the prerequisite for Openser and one of the items keeps failing me. This the error below. Any help will be appreciated. [root@localhost openser]# yum install libmysqlclient15-dev Setting up Install Process Setting up repositories updates-released 100% |=========================| 951 B 00:00 extras 100% |=========================| 1.1 kB 00:00 base 100% |=========================| 1.1 kB 00:00 Reading repository … [View More]

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Re: [Users] rtpproxy or RTPProxy, where to download or checkout
by Klaus Darilion 23 Aug '06

23 Aug '06

Bill Zhang wrote: > This could be a simple question but I went through the whole user archive > and yet to find out where to download it or cvs checkout, I do see > mediaproxy as part of OPENSER module, wonder why rtpproxy or RTPProxy is > not:(. Hi Bill! There are 2 RTP proxies available. 1. Mediaproxy 2. rtpproxy To communicate with Mediaproxy, openser uses the mediaproxy module. To communicate with rtpproxy, openser uses the nathelper module. The modules are part of openser, … [View More]

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[Users] Redirect
by Andrew And 23 Aug '06

23 Aug '06

I am using redirect with ENUM, and am I wanting to know if my code is correct, then I write the code below. Could someone help me please? . . . } else if (method=="INVITE") { # If Invite is to ENUM. # Block ENUM prefix("+"); enum_query("e164.arpa."); #Redirect sl_send_reply("300","Redirect"); return; };

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Re: [Users] 403 Forbidden-----onsip configs testing. user not able to register
by Ferianto siregar 23 Aug '06

23 Aug '06

Hi, Ram its Not looking up in aliases table , it look ups only on location table. the each request and responses . So, You have to add your domain in the domain table by using : insert into domain values('urs_domain', ' date) ; in openser databases tables. I do hope it can help you. Thanks. On 8/22/06, ram wrote: > > Hi all > > iam trying to read the getting started document from onsip > when iam configuring pstn-gw.cfg > with the below config > … [View More]> when the X-lite user try to loging , SIP/2.0 403 Forbidden, in the Log of > X-lite > > what is wrong , even i have changed AVP after reading module configs in > openser. > > any one suggest me what is wrong iam doing here > > Ram > > ----------------- > openser.cfg > # > # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ > # > # simple quick-start config script > # > > # ----------- global configuration parameters ------------------------ > > debug=9 # debug level (cmd line: -dddddddddd) > fork=yes > log_stderror=no # (cmd line: -E) > > /* Uncomment these lines to enter debugging mode > #fork=no > #log_stderror=yes > */ > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > listen=myip > port=5060 > children=4 > fifo="/tmp/openser_fifo" > fifo_db_url="mysql://openser:openserrw@localhost /openser" > > > # ------------------ module loading > ---------------------------------- > > > loadmodule "/usr/local/lib/openser/modules/mysql.so" > loadmodule "/usr/local/lib/openser/modules/sl.so" > loadmodule "/usr/local/lib/openser/modules/tm.so" > loadmodule "/usr/local/lib/openser/modules/rr.so" > loadmodule "/usr/local/lib/openser/modules/maxfwd.so" > loadmodule "/usr/local/lib/openser/modules/usrloc.so" > loadmodule "/usr/local/lib/openser/modules/registrar.so" > loadmodule "/usr/local/lib/openser/modules/textops.so" > loadmodule "/usr/local/lib/openser/modules/auth.so" > loadmodule "/usr/local/lib/openser/modules/auth_db.so" > loadmodule "/usr/local/lib/openser/modules/uri.so" > loadmodule "/usr/local/lib/openser/modules/uri_db.so" > loadmodule "/usr/local/lib/openser/modules/domain.so" > loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/openser/modules/nathelper.so" > loadmodule "/usr/local/lib/openser/modules/avpops.so" > loadmodule "/usr/local/lib/openser/modules/permissions.so" > > modparam("auth_db|permissions|uri_db|usrloc","db_url", > "mysql://openser:openserrw@localhost/openser") > 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("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > > modparam("registrar", "nat_flag", 6) > > modparam("rr", "enable_full_lr", 1) > > modparam("tm", "fr_timer_avp", "i:25") > modparam("tm", "fr_inv_timer_avp", "i:25") > > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > > route { > > # > ----------------------------------------------------------------- > # Sanity Check Section > # > ----------------------------------------------------------------- > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483", "Too Many Hops"); > return; > }; > > if (msg:len > max_len) { > sl_send_reply("513", "Message Overflow"); > return; > }; > > # > ----------------------------------------------------------------- > # Record Route Section > # > ----------------------------------------------------------------- > if (method=="INVITE" && client_nat_test("3")) { > # INSERT YOUR IP ADDRESS HERE > > record_route_preset("myproxyip:5060;nat=yes"); > } else if (method!="REGISTER") { > record_route(); > }; > > # > ----------------------------------------------------------------- > # Call Tear Down Section > # > ----------------------------------------------------------------- > if (method=="BYE" || method=="CANCEL") { > end_media_session(); > }; > > # > ----------------------------------------------------------------- > # Loose Route Section > # > ----------------------------------------------------------------- > if (loose_route()) { > > if (has_totag() && (method=="INVITE" || method=="ACK")) { > > if > (client_nat_test("3")||search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > return; > }; > > # > ----------------------------------------------------------------- > # 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"); > }; > return; > }; > > if (method=="CANCEL") { > route(1); > return; > } else if (method=="INVITE") { > route(3); > return; > } else if (method=="REGISTER") { > route(2); > return; > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; > }; > > if (!lookup("location")) { > sl_send_reply("404", "User Not Found"); > return; > }; > > route(1); > } > > route[1] { > > # > ----------------------------------------------------------------- > # Default Message Handler > # > ----------------------------------------------------------------- > > t_on_reply("1"); > > if (!t_relay()) { > > if (method=="INVITE" || method=="ACK") { > end_media_session(); > }; > > sl_reply_error(); > }; > } > > route[2] { > > # > ----------------------------------------------------------------- > # REGISTER Message Handler > # > ----------------------------------------------------------------- > > sl_send_reply("100", "Trying"); > > if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { > setflag(6); > fix_nated_register(); > force_rport(); > }; > > if (!www_authorize("","subscriber")) { > www_challenge("","0"); > return; > }; > > if (!check_to()) { > sl_send_reply("401", "Unauthorized"); > return; > }; > > consume_credentials(); > > if (!save("location")) { > sl_reply_error(); > }; > } > > route[3] { > > # > ----------------------------------------------------------------- > # INVITE Message Handler > # > ----------------------------------------------------------------- > > if (!allow_trusted()) { > > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > return; > } else if (!check_from()) { > sl_send_reply("403", "Use From=ID"); > return; > }; > > consume_credentials(); > }; > > if (client_nat_test("3")) { > setflag(7); > force_rport(); > fix_nated_contact(); > }; > > if (uri=~"^sip:1[0-9]{10}@") { > strip(1); > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; === message truncated === --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. [View Less]

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[Users] Create 302 message and modify contact header
by Christophe Irles 23 Aug '06

23 Aug '06

Hi, I checked archive without any result... From an openSER rule, I want to create a 302 message if a specific RURI is used. First I need to change the contact header to specify the URI of the new callee. I tried something like that: avp_printf("$avp(s:100)", "sip:test@christophe.fr;target=$ruri;cause=486;"); avp_pushto("$ct", "$avp(s:100)"); # => error generated on this line ERROR:avpops:fixup_pushto_avp: unsupported destination "$ct"; expected $ru,$du,$br So how can I modify the … [View More]

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RE: [Users] rtpproxy or RTPProxy, where to download or checkout--found it:). Re: Users Digest, Vol 15, Issue 61
by Mark Kent 23 Aug '06

23 Aug '06

>> RE: [Users] rtpproxy or RTPProxy, where to download or checkout--found it:) And no one else might have the same hesitation as you? (i.e., why not post the answer to your own question). This is actually semi-interesting, because if you go here http://ftp.iptel.org/pub/rtpproxy/ you get the impression that rtpproxy-0.3 is the one to get. There is a file there named: LATEST-IS-0.3 But here: http://www.voip-info.org/wiki-Portaone+rtpproxy it gives instructions for how to pull … [View More]

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[Users] SMS module (SIP to GSM) question?
by ade darmawan 23 Aug '06

23 Aug '06

ade darmawan <generasi84(a)yahoo.com> wrote: Please help me mr... i have read this module...but i don't understand with 4 question : 1. I found that in the SMS Module (SIP to GSM) don't explained the software which used by the client/user agent or where we should to do testing in client? 2. I wanna know the other facilities should be used, beside openser-1.0.0.tar.gz,mysql,DNS? 3. For the final result, is this module could send SMS by 2 directions? 4. Could you please the … [View More]

1 0

[Users] 403 Forbidden-----onsip configs testing. user not able to register
by ram 23 Aug '06

23 Aug '06

Hi all iam trying to read the getting started document from onsip when iam configuring pstn-gw.cfg with the below config when the X-lite user try to loging , SIP/2.0 403 Forbidden, in the Log of X-lite what is wrong , even i have changed AVP after reading module configs in openser. any one suggest me what is wrong iam doing here Ram ----------------- openser.cfg # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global … [View More]configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=myip port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", " mysql://openser:openserrw@localhost/openser") 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("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_timer_avp", "i:25") modparam("tm", "fr_inv_timer_avp", "i:25") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("myproxyip:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # 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"); }; return; }; if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (uri=~"^sip:1[0-9]{10}@") { strip(1); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); route(5); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("mygateway:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS # avp_write("i:45", "inv_timeout"); 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(); }; } [View Less]

2 1

[Users] options for rtpproxy.
by Bill Zhang 23 Aug '06

23 Aug '06

Anyone can explain following options: 1. -2fv 2. -s path 3. -L nfiles The rest of them seems to be straight forward. This rtproxy documentation is HARD to find:(. usage: rtpproxy [-2fv] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path] [-t tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles] Thanks in advance. Best Regards, Bill -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bill Zhang Sent: Tuesday, August 22, 2006 6:34 PM To: users(a)… [View More]

1 0

RE: [Users] rtpproxy or RTPProxy, where to download or checkout--found it:).
by Bill Zhang 23 Aug '06

23 Aug '06

Thanks. Best Regards, Bill -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bill Zhang Sent: Tuesday, August 22, 2006 5:55 PM To: users(a)openser.org Subject: [Users] rtpproxy or RTPProxy, where to download or checkout This could be a simple question but I went through the whole user archive and yet to find out where to download it or cvs checkout, I do see mediaproxy as part of OPENSER module, wonder why rtpproxy or RTPProxy is not:… [View More]

1 0

[Users] mediaproxy vs. rtpproxy
by Ever Zalazar 23 Aug '06

23 Aug '06

Hi people! Just to know, which is much better for openser mediaproxy or rtpproxy? Best Regards Ever

3 2

[Users] Transaction matching issue?
by H Quintana 22 Aug '06

22 Aug '06

Hello, I'm running OpenSER 1.1.0 in the server xxx.yyy.91.231, I'm also testing a program as 3rd party call control at xxx.yyy.91.230 that is sending "simultaneous" reINVITEs to the 2 parties involved in the call (10252052 and 10092010) through OpenSER. Both messages are accepted by the loose_route function. OpenSER relays the 1st reINVITE (to 10092010) but the second reINVITE (to 10252052) is not relayed. Is openSER considering the 2nd message as a retransmission of the first reINVITE ? … [View More]

2 2

[Serusers] avps question
by Luis Silva 22 Aug '06

22 Aug '06

Hi all! Why isn't possible to do "$avp_name[0] = 0;" ? is there another way of doing it? tks in advance

1 0

[Serusers] preferred order of codecs?
by Ricardo Carvalho 22 Aug '06

22 Aug '06

Hi, Is there a way in Ser to stablish a preferred order of codecs to be used by the phones? Thanks, Ricardo.

6 11

[Users] $ru, $rU, $ru/username, $ruri/username, usage
by Greg Fausak 22 Aug '06

22 Aug '06

I want to use the RURI username in an avp query. I've seen a bunch of different ways to select it. Which way is correct? I am currently using $ruri/username, because I got an error with $rU. http://openser.org/docs/pseudo-variables.html shows '$rU', which complains when I try to reference that. Aug 21 06:55:58 www openser[89119]: ERROR:avpops:fixup_pushto_avp: unsupported destination "$rU"; expected $ru,$du,$br Aug 21 06:55:58 www openser[89119]: ERROR: fix_actions: fixing failed (… [View More]

2 6

[Users] registered("location")
by Helge Waastad 22 Aug '06

22 Aug '06

Hi, I'm just wondering if the 'registered("location")' works when using db only mode for usrloc? br hw

2 5

[Serusers] Modifying the To Field and From field of an INVITE message
by Sammy Touati 22 Aug '06

22 Aug '06

Hi, Can the To and From fields in a SIP INVITE message be modified in SER? I know that the Request URI can be modified but i also need to modify the To field. Thanks. sato.

4 4

[Users] Please help for guiding serweb installation in openser
by Ferianto siregar 22 Aug '06

22 Aug '06

Dear all, Thank you very much to all of you for help me to answer my question about serweb before. So, based on your suggestion, I tried to install serweb 0.9.3 for my openser server. But, when I tried to install it, I am confused. I don`t understand about the installation note that I have got from untar the serweb 0.9.3 packet. These are some questions that I need to ask: 1. Where should we untar the serweb 0.9.3 packet? Should we untar it in root or anywhere we like? Because when we untar … [View More]

1 0

[Users] t_on_failure doesn't
by Nathan Hawkins 22 Aug '06

22 Aug '06

I'm trying to use the LCR module to do a failover between two gateways. For testing purposes, I have a gateway record pointed at an IP that isn't used. The problem is, I can't seem to get OpenSER to fail over to the other gateway. I want to use TCP (with TLS when I'm done), but I get this: Jul 20 15:36:57 asiago /usr/sbin/openser[11896]: route[6]: routing to INVITE sip:MYDID@MYIPADDRESS:5060;transport=tcp Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: ERROR: tcp_blocking_connect: … [View More]timeout (10) Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: ERROR: tcpconn_connect: tcp_blocking_connect failed Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: ERROR: tcp_send: connect failed Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: msg_send: ERROR: tcp_send failed Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: ERROR:tm:t_forward_nonack: sending request failed (IP and username censored.) OpenSER sends this to the caller: SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM) Why can't I catch this with either the return code from t_relay or the failure route? It will go to the failure route if I get a reply back from the next hop. But I need to fail to the other gateway. The relevent parts of the configs are shown below. (Addresses censored.) Any help would greatly appreciated. Thanks, ---Nathan Config snippet: # ok, so maybe try a gateway... if( load_gws() ) { xlog("L_INFO","route[3]: gateway routing $rm $ru"); t_on_failure("1"); route(6); return; } } route[6] { xlog("L_INFO","route[6]: $rm $ru"); t_on_failure("1"); t_on_reply("1"); if (!next_gw()) { xlog("L_INFO","route[6]: no more gateways for $ru"); exit; } xlog("L_INFO","route[6]: routing to $rm $ru"); if(!t_relay()) { xlog("L_INFO","route[6]: t_relay returned $rc"); sl_reply_error(); } return; } failure_route[1] { xlog("L_INFO","failure_route[1]: $rm $ru"); t_on_failure("1"); if(t_check_status("404")) { xlog("L_INFO","failure_route[1]: failing call with 404 for $ru"); return; } if (!next_gw()) { xlog("L_INFO","route[6]: no more gateways for $ru"); exit; } append_branch(); xlog("L_INFO","failure_route[1]: sending $rm $ru"); if(!t_relay()) { xlog("L_INFO","route[6]: t_relay returned $rc"); } return; } Gateways: mysql> select * from gw where grp_id=4; +---------+--------+-----------+------+------------+-----------+--------+ | gw_name | grp_id | ip_addr | port | uri_scheme | transport | prefix | +---------+--------+-----------+------+------------+-----------+--------+ | ithaka | 4 | CENSORED | 5060 | NULL | 2 | NULL | | larink | 4 | BAD_ADDRESS | 5060 | NULL | 2 | NULL | +---------+--------+-----------+------+------------+-----------+--------+ 2 rows in set (0.00 sec) ithaka is a working OpenSER proxy. larink is an empty IP address. (There used to be a server named larink.) [View Less]

3 6

[Users] SER + phone provisioning server - what are the options?
by Mahatma Sarasvati 22 Aug '06

22 Aug '06

Hi All, I'm tired of manually editing text files. What solutions are available for avoiding manual editing and saving of Cisco .cnf files for individual phones while using SER or OpenSER? I've got 80+ phones in my organization and I'm tired of manually managing the phone .cnf files I'm imagining some piece of code that auto-generates the files on command or on a schedule based on entries in the MySQL database. Maybe a column can be added to the Subscriber table or somewhere that holds the … [View More]

1 0

[Users] Bug in nathelper module or in the replace() function
by Federico Giannici 22 Aug '06

22 Aug '06

If a message body is modified by both force_rtp_proxy() and replace( "^Content-Length:", "l:" ) then the resulting message is corrupted. The message size is updated but the header field name appear in both the short and long form. Here it is an example: l:Content-Length: 403 The result is the same if I reverse the order of the two operations too. BTW, I use that replace call (and others) when the message is large to prevent it to fragment. Bye. -- … [View More]

3 6

[Serusers] SER + SEMS in TCP case
by Pat wang 22 Aug '06

22 Aug '06

Hi all, Has anyone tried SER + SEMS in TCP mode yet? I mean the UAs use TCP as transport protocol. I am able to setup SEMS as a conference server. I found "transport=TCP" is missing in the Contact header in the 200ok/SDP after SEMS answers the call. Eyebeam seems not like it and does not response to that 200ok. Result is SER keeps sending that 200ok/SDP till timeout. I tried to use textop module to catch that 200ok then replace the Contact header with a correct one but SER and SEMS … [View More]

1 0

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