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37454 discussions

[Users] Problem with voice
by Andrew Nowrot 21 Feb '07

21 Feb '07

Hi I am new to all openser related things. I managed to install it, combine with postgres and do simple accounting, but now I am facing a slightly problem. The situation looks like this: I have two sip phones both on public IP and both registered to openser (on public IP). I am able to ring between this two phones (so I guess that the sip signalization works OK), but I can't hear any voice. For testing purposes I disabled the firewall on openser so I am sure that all ports are "free". My openser.cfg looks like this: #------------------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes # Set to no to enter debugging mode log_stderror=no # (cmd line: -E) Set to yes to enter debugging mode check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/build/buildd/openser-1.1.0 /debian/openser/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/build/buildd/openser-1.1.0 /debian/openser/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/build/buildd/openser-1.1.0 /debian/openser/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/postgres.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/dbtext.so" #loadmodule "/usr/lib/openser/modules/mediaproxy.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", no) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column", "ha1") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #usrloc|auth_db| modparam("usrloc|auth_db", "db_url", " postgres://openser:openserrw@XXX.XXX.XXX.XXX:5432/openser") modparam("acc", "db_url", "postgres://openser:openserrw@XXX.XXX.XXX.XXX :5432/openser") #modparam("acc", "db_url", "postgres://openser:openserrw@XXX.XXX.XXX.XXX :5432/openser") #modparam("acc", "failed_transaction_flag", 1) #modparam("acc", "report_cancels", 1) modparam("acc", "db_flag", 1) modparam("acc", "log_level", 1) #modparam("acc", "log_fmt", "mfs") #modparam("acc", "log_flag", 1 ) #modparam("acc", "log_fmt", "miocfs") #modparam("acc", "db_flag", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if(method=="INVITE") { # mark the INVITEs for accounting setflag(1); # enforce record-routing so the BYEs will come through this server record_route(); }; #if (!method=="REGISTER") # record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request if(method=="BYE") { setflag(1); } t_relay(); return; #append_hf("P-hint: rr-enforced\r\n"); #route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) } else { if (method=="REGISTER") { #Uncomment this if you want to use digest authentication if (!www_authorize("XXX.XXX.XXX.XXX", "subscriber")) { www_challenge("XXX.XXX.XXX.XXX", "0"); exit; } else { save("location"); exit; }; }; #lookup("aliases"); #if (!uri==myself) { # append_hf("P-hint: outbound alias\r\n"); # route(1); #}; # native SIP destinations are handled using our USRLOC DB if (uri=~"sip:003275566*") { if (!lookup("location")) { sl_send_reply("444", "Not Found"); exit; } append_hf("P-hint: usrloc applied\r\n"); } else { rewritehost("YYY.YYY.YYY.YYY"); #t_relay(); }; }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; #exit; } What could cause this silence? I was wondering maybe mediaproxy or rtpproxy could help, but there is no nat so I can find a reason why I should use them. Could you check my conf file and point me into appropriate direction. Any kind of help would be gladly expected. Best wishes to all. Andrew

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[Users] CDR extraction
by Nick De Cristofaro 21 Feb '07

21 Feb '07

Anyone can recommend a good CDR extraction module that would work with old ser-0.8.14 ? I want it to parse all our calls that are sitting in our old logs. Thanks Nick

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Re: [Users] Cisco 7960 phones
by Scott Yagel 21 Feb '07

21 Feb '07

Ok, I figured out how to unlock the config, using the new menu item called (doh!) "unlock config". One 7960 phone now a functioning SIP phone on the OpenSER. Last thing to figure out is why I cannot access it by browser. Thanks again for the help. Scott Yagel PacketCall, Inc. syagel(a)packetcall.net >-----Original Message----- >From: Stephen Varga [mailto:varga@zbzoom.net] >Sent: Wednesday, February 21, 2007 11:38 AM >To: syagel(a)packetcall.net >Cc: users(a)openser.org >Subject: Re: [Users] Cisco 7960 phones > >Scott, > >Put the attached files into your tftp server that the phones are pulling >their configs and image from. You should also have the following image >files in the same tftp server: > > P003-08-2-00.bin > P0S3-08-2-00.loads > P0S3-08-2-00.zip > P003-08-2-00.sbn > P0S3-08-2-00.sb2 > >I have used these files to successfully upgrade phones. > >If you have SIPxxxxxxxxxxxx.cnf file created for the phones, make sure >the image_version line is commented out. > >The biggest problem I have with this version is using that it reads the >dialplan.xml file but it does not work; I am guessing there is a syntax >change but can not been able to find SIP documentation related to the >8.2 code on Cisco's website. I have been staying with the 7.5 version >of the code because of this problem. > >I found watching which files are being requested by phone a big help in >getting the loads to work properly > >Let me know what happens. > >Steve >

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[Serusers] Accounting with Stateless
by Gustavo Passos Tourinho 21 Feb '07

21 Feb '07

Hi All. Is it possible to use radius accounting with a stateless proxy? For example: if (uri=~"^sip:0001@test.org") { acc_rad_request ("Start"); rewritehostport("127.0.0.1"); forward (uri:host); } When I use this code, the radius request appears without Sip-Translated-Request-URI, why? Changing the above code for this, it works fine (include Sip-Translated-Request-URI): if (uri=~"^sip:0001@test.org") { rewritehostport("127.0.0.1"); t_relay(); } Thanks, Tourinho

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[Users] Re: Problem with voice
by vallimamod abdullah 21 Feb '07

21 Feb '07

Hi Andrew, On 2/20/07, "Andrew Nowrot" <andrew.nowrot(a)gmail.com> wrote: > > Hi > > [...] > I have two sip phones both on public IP and both registered to openser (on > public IP). I am able to ring between this two phones (so I guess that the > sip signalization works OK), but I can't hear any voice. For testing > purposes I disabled the firewall on openser so I am sure that all ports are > "free". My openser.cfg looks like this: > I faced a similar problem some time ago. Here are some tips based on what I have tried to solve this: - Capture the sip messages using ngrep: 'ngrep -qt -d eth0 -W byline port 5060' (of course, replace eth0 with your real iface) - Check the sdp part to see what media IP adress is advertised by each party ('c=IN IP4' line in the INVITE request and OK response.) You may have private IPs here if your sip phones are behind a NAT. - Sniff udp traffic in the port range where the RTP traffic is likely to be (I use '-s0 -X' options on tcpdump to get the payload in hex.) - If you don't see any incoming our outgoing packet then, there ay be a firewall drop issue. Hope this will help. -- Regards, -vma .

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[Users] Audio problem in NAT
by raviprakash sunkara 21 Feb '07

21 Feb '07

Hello Users, I posted so many mail to users but no one reply my issue please help me openSER proxy is mysipdomain.com , and its private_ ip is 192.168.2.60 and SIP server and Proxy is also in Behind UAC's are Behind the NATs --------------------- INVITE --------------- U 59.144.88.7:5060 -> 192.168.2.60:5060 INVITE sip:9002@mysipdomain.com;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKe2a540a8170eb12a. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>. Call-ID: 1685867393(a)192.168.1.2. CSeq: 1 INVITE. Contact: Indian-2 <sip:8002@192.168.1.2:5060;user=phone;transport=udp>. User-Agent: Cisco ATA 188 v3.2.1 atasip (050616A). Expires: 300. Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: 100rel,replaces. Content-Length: 245. Content-Type: application/sdp. v=0. o=8002 14279 14279 IN IP4 192.168.1.2. s=ATA186 Call. c=IN IP4 192.168.1.2. t=0 0. m=audio 16386 RTP/AVP 0 4 8 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.2.60:5060 -> 61.17.248.68:3186 INVITE sip:9002@192.168.2.7:5060;user=phone;transport=udp SIP/2.0. Max-Forwards: 10. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: Indian-2 <sip:8002@59.144.88.7:5060;user=phone;transport=udp>. User-Agent: Cisco ATA 188 v3.2.1 atasip (050616A). Expires: 300. Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: 100rel,replaces. Content-Length: 265. Content-Type: application/sdp. v=0. o=8002 14329 14329 IN IP4 192.168.1.2. s=ATA186 Call. c=IN IP4 192.168.1.2. t=0 0. m=audio 16386 RTP/AVP 0 4 8 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=direction:active. ------------------- RINGING------------ U 61.17.248.68:3186 -> 192.168.2.60:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Require: 100rel. RSeq: 1. Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Content-Length: 0. . # U 192.168.2.60:5060 -> 59.144.88.7:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Require: 100rel. RSeq: 1. Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Content-Length: 0. U 61.17.248.68:3186 -> 192.168.2.60:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. . v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.2.60:5060 -> 59.144.88.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. . v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 61.17.248.68:3186 -> 192.168.2.60:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. . v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.2.60:5060 -> 59.144.88.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

2 1

[Users] SDP Parser
by Michel Bensoussan 21 Feb '07

21 Feb '07

Hello For each voice session I need to know the used codec (for bandwith calculation). For that I need to parse the SIP message body. I didn't find in OpenSER such a functionality. Is there a module that doing that? Or maybe someone is working on it? A suggestion for an open source? Thanks. Regards, Michel.

6 15

[Serusers] SER 2.0 TLS module documentation
by Greger V. Teigre 21 Feb '07

21 Feb '07

FYI. This doc is available at http://www.iptel.org/ser/doc/modules/tls It is auto-updated every night from the latest CVS. See http://www.iptel.org/views/tableofmodules for all available module docs. g-) Andrei Pelinescu-Onciul wrote: > andrei 2007/02/21 01:54:45 CET > > SER CVS Repository > > Modified files: > modules/tls README.TLS > Added files: > modules/tls README > modules/tls/doc Makefile certs_howto.xml functions.xml > history.xml params.xml tls.xml > Log: > - added tls module documentation (not yet complete, still missing select, rpc and better tls.cfg description). > > Test case: try to read the generated README and see if it makes sense and you uderstand at least 88,73% from it. Prerequisites: well rested and a blood alcohol level within legal driving limits. WARNING: known to induce strong drowsiness. > > Revision Changes Path > 1.1 +483 -0 sip_router/modules/tls/README (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/README… > 1.2 +2 -0 sip_router/modules/tls/README.TLS > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/README… > 1.1 +29 -0 sip_router/modules/tls/doc/Makefile (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/Ma… > 1.1 +94 -0 sip_router/modules/tls/doc/certs_howto.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/ce… > 1.1 +17 -0 sip_router/modules/tls/doc/functions.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/fu… > 1.1 +22 -0 sip_router/modules/tls/doc/history.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/hi… > 1.1 +395 -0 sip_router/modules/tls/doc/params.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/pa… > 1.1 +103 -0 sip_router/modules/tls/doc/tls.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/tl… > _______________________________________________ > Serdev mailing list > Serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev > > >

1 0

[Users] "noisy feedback"
by Peter Gradwell 21 Feb '07

21 Feb '07

Hi What does "noisy feedback" mean? Quite often we see it in sip traces: SIP/2.0 200 OK To: "Andy Abraham"<sip:2036367@lon-sip-1.gradwell.net>;tag=20c56ba31bbe85b9bc9d0e0 9f876aff7.93cb From: "Andy Abraham"<sip:2036367@lon-sip-1.gradwell.net>;tag=wgu8z0auky Via: SIP/2.0/UDP 192.168.153.100:2085;branch=z9hG4bK-shuf9rjbthup Call-ID: 3c26700aecd1-45fzt2ryqvy3@snom190 CSeq: 2 REGISTER Contact: <sip:2036367@192.168.153.100:2085;line=oru1hcdy>;q=1;expires=25 Server: OpenSer (1.2.0-pre5-notls (i386/linux)) Warning: 392 193.111.201.74:5060 "Noisy feedback tells: pid=6112 req_src_ip=193.111.200.11 req_src_port=5082 in_uri=sip:lon-sip-1.gradwell.net out_uri=sip:lon-sip-1.gradwell.net via_cnt==1" Content-Length: 0 Cheers peter -- peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/ -- engineering & hosting services for email, web and voip -- -- http://www.peter.me.uk/ -- http://www.voip.org.uk/ --

2 1

[Serusers] Radius + Route
by Zeeshan Sajid 21 Feb '07

21 Feb '07

i am trying to configure radius with SER and I have succeeded in doing authentication and accounting for SER using Radius. Now my question is there any way that we can get/load the address of the next gateway and route our call to that. i mean i dont want to use the LCR module instead i would like to use some other LCR code that communicates via radius. thanks in advance regards Zeeshan Awan --------------------------------- Looking for earth-friendly autos? Browse Top Cars by "Green Rating" at Yahoo! Autos' Green Center.

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