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37559 discussions

Re: [Users] OpenSER administrator issue
by Mike Williams 21 Aug '06

21 Aug '06

I don't think I'm using them, unless there's a bug in Kmail. Mike On Friday 11 August 2006 01:14, Louis Alexander wrote: > Btw, all emails I'm getting on this thread are blank. Are you guys using > PGP signatures? I think my outlook has an issue with that > > > > _____ > > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On > Behalf Of Mike Williams > Sent: Friday, August 11, 2006 12:35 PM > To: users(a)openser.org > Subject: Re: [Users] OpenSER administrator issue > > > > > > > > This e-mail and any attachment is for authorised use by the intended > recipient(s) only. It may contain proprietary material, confidential > information and/or be subject to legal privilege. It should not be copied, > disclosed to, retained or used by, any other party. If you are not an > intended recipient then please promptly delete this e-mail and any > attachment and all copies and inform the sender. Thank you.

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[Serusers] SER TCP Bug
by Curt Moore 21 Aug '06

21 Aug '06

Hello all. I've been doing some experimentation with both CVS HEAD and 0.9.6 of SER and version 1.1.0 of OpenSER. It appears that there is a TCP bug in SER which has been fixed in OpenSER. I've seen various posts in the past on the serdev and the serusers lists regarding this issue but with no definitive solution. Are there any plans to merge some of the great work of the OpenSER developers back into SER, such as this fix for TCP? Below is the logfile I receive when I try to place a call from a UDP to a TCP device using CVS HEAD of SER. The same routing logic in ser.cfg works perfectly in openser.cfg. Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: tcp_blocking_connect: poll error: flags 18 Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: tcp_blocking_connect: SO_ERROR (113) No route to host Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: tcpconn_connect: tcp_blocking_connect failed Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: tcp_send: connect failed Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: msg_send: ERROR: tcp_send failed Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: t_forward_nonack: sending request failed Aug 17 19:03:33 sg1test /usr/sbin/ser[32534]: ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Does anyone have any ideas on how to get this fixed in CVS HEAD of SER? I've tried some backporting of the changes to the tcp_* files from OpenSER to SER but have not had any luck as of yet as I'm not fully sure of how the code is supposed to be handling each case. This is to say that I'm just looking at code differences without really understanding how the code is supposed to work. I'm happy to do anything I can to help get this resolved in SER's CVS HEAD. Thanks! -Curt

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[Users] Relay and Redirect
by Andrew And 21 Aug '06

21 Aug '06

I receive "Redirect" with more than one contact destination, and I need to do fork (parallel). I am using "t_relay()" to do this, but I didn´t see correct relay because, now, the Openser sent Invite only one destination. Is there another command to do relay? Could some one help me?

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RE: [Users] Rewriting Contact HF port
by Justin Pearce 21 Aug '06

21 Aug '06

Hi Bogdan, I knew that Nathelper did some rewriting, but the circumstances required that the thing record the private IP and a certain known port instead of the public IP as a contact address (using fix_nated_register() to save the other address). I wound up borrowing the code from the Nathelper module and modifying it to accept a port argument and made it a separate module. This way, it only rewrote the port and not left everything else untouched. Thanks. Justin Pearce Information Technology/Programming Price Video Productions JustinP(a)PriceVideo.com 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Monday, August 21, 2006 6:19 AM To: Justin Pearce Cc: Klaus Darilion; users(a)openser.org Subject: Re: [Users] Rewriting Contact HF port Hi Justin, if you need save(location) to save as "received" a custom uri, you just need to set an avp with the uri you need to be saved. See: http://openser.org/docs/modules/1.1.x/registrar.html#AEN211 It's exactly what nathelper does. regards, bogdan Justin Pearce wrote: >Thanks, Klaus > >It looks like I'll have to make something similar to the nathelper >module to do what I need. > >Justin Pearce > >-----Original Message----- >From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >Sent: Monday, August 07, 2006 1:43 AM >To: Justin Pearce >Cc: users(a)openser.org >Subject: Re: [Users] Rewriting Contact HF port > >Justin Pearce wrote: > > >>I am wondering if there is a good way to rewrite the port in the >> >> >contact > > >>header in OpenSER 1.0.1 >> >> > >1. Why do you want to rewrite the port to fixed port? NAT problems? > > > >>before it is stored in the location table?(I'm using the MySQL >> >> >database > > >>backend). >> >>Thus far, I am using "subst(/^(Contact.+):[0-9]+(.+)$/\1:5060\2/ig);" >>just before I do >> >>"save("location");", but that does not seem to be working. >> >> > >subst will be applied before the message is forwarded. Thus, save() sees > >the original Contact header. You either loop the modified message to the > >SIP proxy again or you write your own function (similar to the function >in nathelper module) > >regards >klaus > > > >> >> >>Suggestions? >> >> >> >>Justin Pearce >> >>Information Technology/Programming >> >>Price Video Productions >> >>JustinP(a)PriceVideo.com <mailto:JustinP@PriceVideo.com> >> >>361-572-3810 >> >>800-733-3810 >> >>Fax: 361-572-3894 >> >>www.PriceVideo.com >> >> >> >> >> >> >> >----------------------------------------------------------------------- - > > >>_______________________________________________ >>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 > > >

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Re: [Serusers] Multihomed SER & NAT
by Sebastian Gabris 21 Aug '06

21 Aug '06

Hi all, finally I figured out how to change the sdp - part of the message by using force_rtpproxy. All works fine, I can make calls in both directions, but i still got two problems: 1. If someone from outside(VoIP - Provider) calls my UA, i have a ~10 sec. delay before i hear audio. - i checked my dumpfile, and saw that the delay is between the last 180 Ringing and the 200 OK. (- However RTP - Packets are sent before the 200 OK, after a 183-Message. Dont understand why but I dont think it matters that much.) 2. If the call is terminated from outside(VoIP - Provider) the BYE message is not handled correctly by SER. - the previous messages from the Voip provider have all a via HF with the private IP of my UA. The BYE message doesnt have such a Via. All other Via and RR fields are present. - also the BYE uri is somewaht strange. Instead of sip:123456@privateIPofUA it contains sip:123456@publicIPofSER. Thanks in advance for any help Smile Best regards Sebastian. > -----Ursprüngliche Nachricht----- > Von: "Greger V. Teigre" <greger(a)teigre.com> > Gesendet: 10.08.06 15:41:31 > An: Sebastian Gabris <Sebastian.Gabris(a)web.de> > CC: serusers(a)lists.iptel.org > Betreff: Re: [Serusers] Multihomed SER & NAT fix_nated_contact() is meant for changing the IP address of the Contact > header when SER has a public address and the UA has a private (i.e. > behind NAT). This means that it will rewrite Contact to contain the > source ip address:port found in the UDP/TCP header. > This is probably correct, because I assume the UA cannot be reached > from the outside anyway. If you record-route, subsequent requests > should go through your ser. > > If you have problems with responses not being returned, have a look at > Via headers as they control how answers to i.e an INVITE are routed > back to the UA. You may want to have a look at the rr module double > record route parameter to make sure that SER creates one record-route > for the internal and one for the external interface. > g-) > > Sebastian Gabris wrote: > After reading a bit more mails, I came to two possible problems: > 1. fix_nated_contact() needs a registrar to work properly > I dont use a registrar, since i register at the VoIP - Provider. > or > 2. fix_nated_contact() is not ment to be used on this scenario > Any ideas on this? > Sebastian > > > Hi everybody, > need some help on nattraversal. > i use a multihomed SER, one private IP & one public IP. > MyUA(privateIP) <-> (privateIP)SER(publicIP) <-> UA-(VoIP - Provider, pubicIP) > If i try to make a call from private net to public net, SER doesnt change the contact address. He changes the sdp - connection field but he, puts in the wrong IP - address. > Instead o writing down his public IP, SER writes his private IP. > Then SER forwards the call on his public interface to the provider, which will get wrong Contact & SDP information. > After reviewing my logs i saw that SER does the nat_uac_test, but the fix_nat_contact & fix_sdp dont work dont do what they should do :) > Here are the important bits of my config: > fork=yes > log_stderror=no mhomed=1 > listen=10.x.x.x > listen=193.x.x.x > loadmodule "/usr/lib/ser/modules/nathelper.so" > modparam("usrloc", "db_mode", 0) > modparam("registrar", "nat_flag", 6) > modparam("nathelper", "natping_interval", 30) # Ping interval 30 s > modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT > if (nat_uac_test("3")) { > if (method == "REGISTER" || ! search("^Record-Route:")) { > log(2,"LOG: Someone trying to register from private IP, rewriting\n"); fix_nated_contact(); # Rewrite contact with source IP of signalling > if (method == "INVITE") { > fix_nated_sdp("1"); # Add direction=active to SDP > }; > force_rport(); # Add rport parameter to topmost Via > setflag(6); # Mark as NATed > }; > }; > ... and so on. > ______________________________________________________________ > Verschicken Sie romantische, coole und witzige Bilder per SMS! > Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193 > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > _____________________________________________________________________ > Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! > http://smartsurfer.web.de/?mc=100071&distributionid=000000000071 > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

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[Users] acc db_localtime support missing
by Rodrigo P. Telles 21 Aug '06

21 Aug '06

Hi Folks, I tried to use param "db_localtime" on OpenSER 1.0.1 and 1.1.0 but openser complain with 0(28677) set_mod_param_regex: parameter <db_localtime> not found in module <acc> and does not start. openser.cfg looks like: ... loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("acc", "db_localtime", 1) # use GMT time - as OpenSER doc especify ... .. Is that a missing implementation or some thing wrong is happening? Thanks in advance, Telles

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RE: [Users] Rewriting Contact HF port
by Justin Pearce 21 Aug '06

21 Aug '06

Thanks, Klaus It looks like I'll have to make something similar to the nathelper module to do what I need. Justin Pearce -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Monday, August 07, 2006 1:43 AM To: Justin Pearce Cc: users(a)openser.org Subject: Re: [Users] Rewriting Contact HF port Justin Pearce wrote: > I am wondering if there is a good way to rewrite the port in the contact > header in OpenSER 1.0.1 1. Why do you want to rewrite the port to fixed port? NAT problems? > before it is stored in the location table?(I'm using the MySQL database > backend). > > Thus far, I am using "subst(/^(Contact.+):[0-9]+(.+)$/\1:5060\2/ig);" > just before I do > > "save("location");", but that does not seem to be working. subst will be applied before the message is forwarded. Thus, save() sees the original Contact header. You either loop the modified message to the SIP proxy again or you write your own function (similar to the function in nathelper module) regards klaus > > > > Suggestions? > > > > Justin Pearce > > Information Technology/Programming > > Price Video Productions > > JustinP(a)PriceVideo.com <mailto:JustinP@PriceVideo.com> > > 361-572-3810 > > 800-733-3810 > > Fax: 361-572-3894 > > www.PriceVideo.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] BCP: rewrite every header?
by Mark Kent 21 Aug '06

21 Aug '06

I just debugged the following: A PSTN gateway we use is completely non-responsive when the INVITE has this: From: Mark Kent (desk) <sip:markkent@sip.example.com>;tag=foo Contact: Mark Kent (desk) <sip:markkent@192.0.2.216:5060> but things work OK for this: From: "Mark Kent (desk)" <sip:markkent@sip.example.com>;tag=foo Contact: "Mark Kent (desk)" <sip:markkent@192.0.2.216:5060> and also for this: From: Mark Kent <sip:markkent@sip.example.com>;tag=foo Contact: Mark Kent <sip:markkent@192.0.2.216:5060> Now, I know that the EndUser box should put quotes in there to be safe. RFC3261 says "special characters MUST be in a quoted string to be used within a parameter value." But that's not the point of this post... the point is, should I be rewriting every header? Is that standard practice? BTW, I have the same hangup with the SJphone softphone, but there I think it's because of this: From: "Mark Kent(mac)"<sip:MarkKent@sip.example.com>;tag=foo See, no space between the double-quote and the left-angle-bracket. So, should I be re-writing headers to present a single face, independent of the EndUser boxes, to external SIP gateways? Thanks, -mark P.S. FYI, the box that is not putting in the quotes is a SPA3000 running 3.1.10(GWd). A SPA942 running 4.1.10(e) puts in the quotes.

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[Users] check displayname with proxy-authorizationname
by Hakan YASTI 21 Aug '06

21 Aug '06

Dear All, I want to check the "AUTHORIZATION_USERNAME" at proxy-authorization with the "DISPLAY_NAME" of the user agent.( I want them to be same ). How can I do this ? Regards, Hakan. U 2006/08/17 14:50:10.915465 10.0.0.10:9769 -> 10.0.0.15:5060 INVITE sip:DIALEDNUBER@10.0.0.15 SIP/2.0. To: <sip:DIALEDNUBER@10.0.0.15>. From: DISPLAY_NAME<sip:USERNAME@10.0.0.15>;tag=cb1e6d26. Via: SIP/2.0/UDP 10.0.0.10:9769;branch=z9hG4bK-d87543-880502875-1--d87543-;rport. Call-ID: a21d332a7f23d268. CSeq: 1 INVITE. Contact: <sip:USERNAME@10.0.0.10:9769>. 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: 229. . v=0. o=- 6672092 6672111 IN IP4 10.0.0.10. s=eyeBeam. c=IN IP4 10.0.0.10. t=0 0. m=audio 9764 RTP/AVP 18 101. a=alt:1 1 : 4F88BF12 49561D9D 10.0.0.10 9764. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 2006/08/17 14:50:10.917409 10.0.0.15:5060 -> 10.0.0.10:9769 SIP/2.0 407 Proxy Authentication Required. To: <sip:DIALEDNUBER@10.0.0.15>;tag=bf2a561eb1268c60203a720a0640ec5a.1fcb. From: DISPLAY_NAME<sip:USERNAME@10.0.0.15>;tag=cb1e6d26. Via: SIP/2.0/UDP 10.0.0.10:9769;branch=z9hG4bK-d87543-880502875-1--d87543-;rport=9769. Call-ID: a21d332a7f23d268. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="10.0.0.15", nonce="44e4591e0c9257971b55c6113dbcbbdb18552cd3". Server: OpenSer (1.0.1 (i386/freebsd)). Content-Length: 0. Warning: 392 10.0.0.15:5060 "Noisy feedback tells: pid=1378 req_src_ip=10.0.0.10 req_src_port=9769 in_uri=sip:DIALEDNUBER@10.0.0.15 out_uri=sip:DIALEDNUBER@10.0.0.15 via_cnt==1". . # U 2006/08/17 14:50:10.919273 10.0.0.10:9769 -> 10.0.0.15:5060 ACK sip:DIALEDNUBER@10.0.0.15 SIP/2.0. To: <sip:DIALEDNUBER@10.0.0.15>;tag=bf2a561eb1268c60203a720a0640ec5a.1fcb. From: DISPLAY_NAME<sip:USERNAME@10.0.0.15>;tag=cb1e6d26. Via: SIP/2.0/UDP 10.0.0.10:9769;branch=z9hG4bK-d87543-880502875-1--d87543-;rport. Call-ID: a21d332a7f23d268. CSeq: 1 ACK. Content-Length: 0. . # U 2006/08/17 14:50:10.920445 10.0.0.10:9769 -> 10.0.0.15:5060 INVITE sip:DIALEDNUBER@10.0.0.15 SIP/2.0. To: <sip:DIALEDNUBER@10.0.0.15>. From: DISPLAY_NAME<sip:USERNAME@10.0.0.15>;tag=cb1e6d26. Via: SIP/2.0/UDP 10.0.0.10:9769;branch=z9hG4bK-d87543-460538832-1--d87543-;rport. Call-ID: a21d332a7f23d268. CSeq: 2 INVITE. Contact: <sip:USERNAME@10.0.0.10:9769>. Max-Forwards: 70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="AUTHORIZATION_USERNAME",realm="10.0.0.15",nonce="44e4591e0c9257971b55c6113dbcbbdb18552cd3",uri="sip:DIALEDNUBER@10.0.0.15",response="93036675b43665e6c94523bf841b7678",algorithm=MD5. User-Agent: eyeBeam release 3004t stamp 16741. Content-Length: 229. . v=0. o=- 6672092 6672111 IN IP4 10.0.0.10. s=eyeBeam. c=IN IP4 10.0.0.10. t=0 0. m=audio 9764 RTP/AVP 18 101. a=alt:1 1 : 4F88BF12 49561D9D 10.0.0.10 9764. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv.

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[Users] doubt in Radius and openser fro accounting...
by raviprakash sunkara 21 Aug '06

21 Aug '06

Hi Users, For Me, in Openser + radius Authorize and Acthentication is Succeed Very Good and Nice, But When some to Account, I'm little sucking here When Uac1 dials to Uac2 , ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ radacctID | AcctSessionID | AcctuniqueID | UserName | AcctStartTime | ActStopTime | calledID | callingID | ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- 1 12h34yy 2334juuw45 Uac1 13:10:00 00:00:00 Uac2 uac1 2 12h34yy ko34ji899 Uac2 13:11:11 13:11:11 Uac1 Uac2 --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- In accountting with openser + radius in should insert only one insert for every call made . But I think it insert only two records only. and AcctuniqueD should me same for each call makes then only the AcctStopTime will properly ..... in openser... -------------------- loose record----------------------------------- if(method=="BYE" || method=="CANCEL") { acc_rad_request("Stop"); unforce_rtp_proxy(); } ---------------------------- record route section------------ if( method=="INVITE" || method=="ACK") { acc_rad_request("Start"); forcr_rtp_proxy(); } plz help me in that .. sorry in my english ....... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

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