sr-users November 2004

sr-users@lists.kamailio.org

203 participants
386 discussions

[Serusers] problems with ser+quintum gateway

by voip technocrat

Hi list, Iam useing the SER of version: ser 0.8.12-1rc6 (i386/linux) My aim is to place the calls on to the gateway iam useing the ser.cfg as follows. **************************************************** # ----------- global configuration parameters debug=8 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading # Uncomment this if you want to use SQL database loadmodule "/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" # ----------------- setting module-specific parameters # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; if (!uri=~"sip:9[1-9][0-9].*@.*") { if (method=="INVITE") { sl_send_reply("403", "Call cannot be served here"); } else if(method=="BYE") { forward(uri:host, uri:port); } else { forward(uri:host, uri:port); }; break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP rewritehostport("*.*.*.101:5060"); if (!t_relay()) { sl_reply_error(); }; } **************************************************** i have number like 214 which has been registerd with the fxs device to our ser proxy . so when i dial the numbers like 9198495305756 which will be pstn terminated with the help of quintum gateway. i observerd the request reaching to the quintum is with the address/domain/ ip of client Ip not the server domain / ip while in the gateway we have allowed all calls coming from the our ser domain so by result of this when an invite is coming with the client ip it could not able to acess the pstn number for the first few seconds we get the message like 183 processing the call but after some time it gives the message like 486 busy and call ends. but when i allow the client ip also in my quintum gateway the call is being sucessfull established i think ser is doing the state less forwarding this is main problem iam faceing. help of people like you in this will be highly appreciated with regards ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

20 years

RE: [Serusers] SER and IC-RADIUS

by Lucas Aimaretto

> > >try to include both dictionary.ser (from SER) and dictionary.sip > > >(from radiusclient) on ICRADIUS dictionary > > > I did, but nothing new happened ... > > I forget to say, add them to /etc/radiusclient/dictionary, > too. both ICRADIUS and radiusclient needs to point to the > same dictionary file content! Both of them have dictionary.sip and dictionary.ser (/etc/radiusclient/dictionary and my ICRadius dictionary). But, you know, ICRadius has its dictionary in a mysql database. Will it have something to do with everything ?? But, both dictionaries are just the same. > > It just makes no sense ... Why would some attributes be > > recognized and > > some others not. It is not logical, as they are all (the > > attributes) > > taken from the same files (dictionary.sip and dictionary.ser). > > check if /etc/radiusclient/radiusclient.conf points > to the correct dictionary you're using Yes it does. > > Is there any special configuration needed for the ser.cfg > > or for the > > radiusclient.conf ? The radius.conf file of my icradius has > > little to touch ... Any ideas ?? > > perhaps your SER is (like mine) pointing to > /usr/local/etc/radiusclient/ and you're tweaking /etc/radiusclient/ ? [root@radius]# ls -l /etc/radiusclient/dictionary lrwxrwxrwx 1 root root /etc/radiusclient/dictionary -> /usr/local/etc/radiusclient/dictionary Is this what you meant ??? By the way ... I mak a call but did not see the called-station-Id nor the calling-station-Id, any ideas ?? But the call could just be placed, cause, 1st authentication takes place ... I mean, the User making the call exists and so I get an access-accept from radius. Then ser places de call, but I do not see the 2nd authentication taking place ... Does this happen to you ?? > []s, !3runo Lucas --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.786 / Virus Database: 532 - Release Date: 29/10/2004

20 years

[Serusers] jiri away from mail

by jiri kuthan

I'm out of office in the IETF meeting (Washington DC) and elsewhere and cannot read my emails as frequently as I would wish. If you have any urgent questions to iptelorg, call me or contact my colleague, Dr. Sisalem, sisalem(a)iptel.org. -Jiri

20 years

[Serusers] rtpproxy monitoring / health checking

by Java Rockx

Hi All. Is there a way to monitor rtpproxy? Aside from the typical health checks on the process itself, is there a way to actually make sure that rtpproxy is processing calls and not simply a hung process? Regards, Paul __________________________________ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com

20 years

[Serusers] nathelper + squid

by Andrea Federico

Hi, can you tell me please how can I pass login and password to squid using nathelper ? Thank You, Andrea. -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

20 years

[Serusers] Using SER with UAs that do not register

by mark.roddy＠lakecommunications.com

Hi SER's, I want to set up a private SIP network over a VPN using SER but some of my UAs are only able to operate in peer to peer mode (ie they cannot register). My proposed network architecture is the following: UA1=192.168.0.4 UA2=192.168.1.4 UA3=192.168.2.4 SER=192.168.0.5 UA1 & UA2 can register with SER but UA3 cannot. I have already set this up and have successfully made calls between UA1 &UA2. How do I set up SER to make calls from either UA1 or UA2 to UA3, or calls from UA3 to either UA1 or UA2? Any help greatly appreciated! thx Mark ********************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify postmaster(a)lakecommunications.com This footnote also confirms that this email message has been scanned for the presence of computer viruses. www.lakecommunications.com **********************************************************************

20 years

[Serusers] problem with the ser+pstn gateway

by voip technocrat

hello friends, i was actuall trying for the ser+gateway but i was missing some where . my idea is that any number stat it gives message as seesion progress with the message type 183 the same is working in between the two sers i.e one user agent registered to the one ser server can be able to place call to another ser server user agent iam pasting my ser.cfg here please guide me where am i going wrong. # ----------- global configuration parameters ------------------------ debug=8 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=438 # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/group.so" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.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", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") modparam("usrloc","db_url","mysql://ser:heslo@localhost:3306/ser") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions" ,1) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper 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 # ------------------------- 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"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { log("***********************INVITE**************************"); fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; setflag(1); setflag(2); 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 append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; if (method=="INVITE") { log("**********************in invite******************************"); log(1, "INVITE\n"); setflag(1); /* set foraccounting (the same valueas in log_flag!) */ }; if (method=="MESSAGE") { log("**********************in message******************************"); log(1, "MESSAGE\n"); setflag(1); /* set foraccounting (the same valueas in log_flag!) */ }; if ( method=="BYE" || method=="CANCEL" ) { log("**********************in bye******************************"); log (1, "BYE or CANCEL\n"); setflag(1); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if(uri=~"^sip:[6-9].*@*.*.*.*") { log("**********************in forward*******************************"); rewritehostport("*.*.*.101:5060"); # forward(uri:host,uri:port); t_relay_to_udp("*.*.*.101","5060"); break; } else { log("**************************in user location********************"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; } }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } with regards tech_voip ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

20 years

[Serusers] [Fwd: Ver 0.8.14 on OpenBSD/sparc segfaults]

by msg

-------- Original Message -------- From: - Thu Nov 18 11:48:27 2004 X-Mozilla-Status: 0001 X-Mozilla-Status2: 00800000 Message-ID: <419CE06A.3020008(a)cybertheque.org> Date: Thu, 18 Nov 2004 11:48:26 -0600 From: msg <msg(a)cybertheque.org> Organization: Cybertheque Museum User-Agent: Mozilla/5.0 (Windows; U; Windows NT 5.0; en-US; rv:1.7.2) Gecko/20040804 Netscape/7.2 (ax) X-Accept-Language: en,ja MIME-Version: 1.0 To: serusers(a)lists.iptel.org Subject: Ver 0.8.14 on OpenBSD/sparc segfaults Content-Type: text/plain; charset=us-ascii; format=flowed Content-Transfer-Encoding: 7bit Greetings: With some tweaks (no pthreads, missing includes, reordering includes, etc) I was able to build Ver 0.8.14 on OpenBSD/sparc 3.2, gcc 2.95.3. When attempting to run the executable for the first few times, I got the unresolved symbol messages other OpenBSD users have posted to this list while 'ser' is loading modules; however this behavior vanished spontaneously and now ser simply segfaults and dumps core after about five seconds (Sun sparc IPX host). OpenBSD 3.2 doesn't have 'pthreads' but instead has a similar 'pth' package; 'ser' was built using SysV semaphores instead. `ser -V`: version: 0.8.14 (sparc/openbsd) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MM AP, PKG_MALLOC, F_MALLOC MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.168.4.3 2004/06/28 15:41:21 andrei Exp $ main.c compiled on 13:32:17 Nov 17 2004 with gcc 2.95 'ser -c' also segfaults but first reports: 0(26788) WARNING: could not read from /dev/random (5) 0(26788) ERROR: shm_mem_init: could not initialize lock Backtrace of the core file from running 'ser' with an empty 'ser.cfg': This GDB was configured as "sparc-unknown-openbsd3.2"... (no debugging symbols found)... Core was generated by `ser'. Program terminated with signal 11, Segmentation fault. Reading symbols from /usr/lib/libc.so.28.5...(no debugging symbols found)... done. Reading symbols from /usr/libexec/ld.so...(no debugging symbols found)...done. #0 0x9ee88 in lock_destroy () (gdb) bt #0 0x9ee88 in lock_destroy () #1 0x9ede3 in shm_mem_destroy () #2 0x9e80b in shm_mem_init_mallocs () #3 0x9ea13 in shm_mem_init () #4 0x9d783 in init_shm_mallocs () #5 0x3acb7 in main () #6 0x118df in ___start () Anyone care to help? All replies much appreciated. Michael Grigoni Cybertheque Museum

20 years

RE: [Serusers] NATHelper + usrloc (+ rtpproxy?)

by Matt Schulte

Ok, that's what I was stating actually. The fact that my invite's go below the t_on_reply, I'll paste my config update. Thanks for the info I'll search google for Java's implementation. ---SNIP--- ser.cfg t_on_reply("1"); if (method=="INVITE") { log(1, "LOG: Caught INVITE \n"); if (lookup("location")) { log (1, "LOG: Caught registered user, sending there\n"); # NOTE forcing rtp maybe bad idea for ALL users, this is # a quick fix #fix_nated_contact(); #force_rtp_proxy(); #forward(uri:host, uri:port); t_relay(); break; } else if (uri=~"^sip:[0-9]*@") { # ... forward to gateways then; forward(xxx.xxx.xxx.xxx, 5060); log(1, "LOG: Tapping rowlf\n"); break; }; }; if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? # if (isflagset(6)) { # log(1, "transaction was sent to a NATED client -> fix nated contact\n"); # fix_nated_contact(); # force_rtp_proxy(); # append_hf("P-hint: fixed NAT contact for response\r\n"); # } if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { log(1, "transaction was sent to a NATED client -> fix nated contact\n"); fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } -----Original Message----- From: Greger V. Teigre [mailto:greger@axxessanywhere.com] Sent: Friday, November 19, 2004 6:21 AM To: Matt Schulte; serusers(a)lists.iptel.org Subject: Re: [Serusers] NATHelper + usrloc (+ rtpproxy?) Hi Matt, When a non-NATed incoming call to a NATed client is processed (INVITE), you must make sure that you have a t_on_reply("1"); before you call t_relay (or forward). The INVITE will not be detected as behind a NAT, but the destination is (flag is set), and the reply will take care of the rewrite. In your config, it looks like you call t_relay before setting t_on_reply("1"); further down. A forward will only forward the SIP INVITE to another SIP proxy for processing. Paul (Java Rockx) just recently posted his config file with a working NAThelper/RTPproxy setup. I suggest you look at the call logic found there. His config is also easy to read with a lot of nice headers :-) I haven't tested RTP proxy between a client behind NAT and Asterisk, but I believe that as long as you record-route the INVITE (as you do) and handle the replies properly, it should work. g-) Matt Schulte wrote: > Another note to this, I moved my 'forward' and lookup statements down > below the t_onreply statement. I figured this should allow ser to see > that the client is in fact behind a NAT. It catches that now however I > see this in my debug (ser): > > ser[21770]: transaction was sent to a NATED client -> fix nated > contact ser[21770]: ERROR: on_reply processing failed > > Could the last error be a/the problem? Come on I know someone else has > had this problem. Please help! > NOTE: I just tested this out on Asterisk (as a client behind NAT) and > got the same results. It's simply not changing the RTP IP address.. > > --snippet-- > > onreply_route[1] { > # NATed transaction ? > if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { > log(1, "transaction was sent to a NATED client -> fix nated > contact\n"); > fix_nated_contact(); > force_rtp_proxy(); > # otherwise, is it a transaction behind a NAT and we did not > # know at time of request processing ? (RFC1918 contacts) > } else if (nat_uac_test("1")) { > fix_nated_contact(); > }; > > > -----Original Message----- > From: Matt Schulte > Sent: Thursday, November 18, 2004 8:09 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] NATHelper + usrloc (+ rtpproxy?) > > > All, > > This is my first post to this list so go easy on me. :-) I'm rather > new to Ser, in fact I just installed it for the first time early in > the week. I'm working on the NAThelper module to get traversal > working, I have outbound (sip phone -> NATout -> ser) working just > peachy, RTP works in both directions hooray. The question is I'm > having problems getting RTP inbound, the ring of course goes through, > and RTP from the NAT'd side of course works fine however getting back > through the NAT (from outside) for RTP in this sense fails. Let me > explain the setup: > > I'm using the registrar, NAThelper, usrloc, and of course (Portaone's) > RTPproxy modules. The current SIP phone is an SNOM (yes yes, I > know..). The "endpoint" is Asterisk. When I do a sip debug on > Asterisk, I see the RTP request however it's coming from the NAT'd > fake address: > > v=0 > o=root 780961119 780961119 IN IP4 192.168.1.101 > s=call > c=IN IP4 192.168.1.101 > t=0 0 > m=audio 10004 RTP/AVP 0 > a=rtpmap:0 pcmu/8000 > a=sendrecv > > > I have an idea of what to fix just not sure how to fix it. Obviously > we need it to goto RTPproxy, since this is "backwards" how would I get > it to recognize the correct IP? > > See my config below, most of it is ripped off of the NAThelper.cfg > example. :-) Thanks all.. > > NOTE: All calls are destined for ${SIPDOMAIN}, in this case, the > machines hostname. This is normal and intentional :-) > > # ---- SNIPPAGE ---- > modparam("rr", "enable_full_lr", 1) > > # !! Nathelper > 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 > # 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"); > break; > }; > if (msg:len >= max_len ) { > sl_send_reply("513", "Message too big"); > break; > }; > # !! Nathelper > # Special handling for NATed clients; first, NAT test is > # executed: it looks for via!=received and RFC1918 addresses > # in Contact (may fail if line-folding is used); also, > # the received test should, if completed, should check all > # vias for rpesence of received > if (nat_uac_test("3")) { > # Allow RR-ed requests, as these may indicate that > # a NAT-enabled proxy takes care of it; unless it is > # a REGISTER > log("LOG: Caught uac test 3 \n"); > if (method == "REGISTER" || ! > search("^Record-Route:")) { > log("LOG: Someone trying to register from private > IP, rewriting\n"); > > # This will work only for user agents that support > symmetric > # communication. We tested quite many of them and > majority is > # smart enough to be symmetric. In some phones it > takes a configuration > # option. With Cisco 7960, it is called > NAT_Enable=Yes, with kphone it is > # called "symmetric media" and "symmetric > signalling". > > fix_nated_contact(); # Rewrite contact with source > IP of signalling > if (method == "INVITE") { > log("LOG: fix nated sdp\n"); > fix_nated_sdp("1"); # Add direction=active to > SDP > }; > force_rport(); # Add rport parameter to topmost > Via setflag(6); # Mark as NATed > }; > }; > > # 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=="REGISTER") record_route(); > > # subsequent messages withing a dialog should take the > # path determined by record-routing > if (loose_route()) { > # mark routing logic in request > append_hf("P-hint: rr-enforced\r\n"); > route(1); > break; > }; > > if (!uri==myself) { > # mark routing logic in request > append_hf("P-hint: outbound\r\n"); > route(1); > break; > }; > > > if (uri==myself) { > > if (method=="REGISTER") { > log("LOG: Caught register, registering user in > local db\n"); > save("location"); > break; > }; > > lookup("aliases"); > if (!uri==myself) { > append_hf("P-hint: outbound alias\r\n"); > route(1); > break; > }; > log("LOG: Caught uri myself\n"); > # native SIP destinations are handled using our USRLOC > DB > #if (!lookup("location")) { > # sl_send_reply("404", "Do what now"); > # break; > #}; > }; > append_hf("P-hint: usrloc applied\r\n"); > route(1); > > } > > route[1] > { > # !! Nathelper > if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" > && !search("^Route:")){ > sl_send_reply("479", "We don't forward to private IP > addresses"); > break; > }; > # if client or server know to be behind a NAT, enable relay > if (isflagset(6)) { > log("LOG: Caught NAT flag 6 forcing rtp proxy\n"); > force_rtp_proxy(); > }; > if (method=="REGISTER") { > break; > log("LOG: Caught Register down in our call routing, > breaking\n"); > }; > > #### Below is mostly my own doing #### > if (method=="INVITE") { > log("LOG: Caught INVITE \n"); > if (lookup("location")) { > log ("LOG: Caught registered invite, sending > there\n"); > # NOTE forcing rtp maybe bad idea for ALL > users, this is > # a quick fix (which doesn't work anyway!) > #force_rtp_proxy(); > #forward(uri:host, uri:port); #nor does this > t_relay(); > break; > } else if (uri=~"^sip:[0-9]*@") { # ... forward to > asterisk; > forward(xxx.xxx.xxx.xxx, 5060); > log("LOG: Tapping rowlf\n"); > break; > }; > }; > #### #### > > t_on_reply("1"); > > if (!t_relay()) { > sl_reply_error(); > }; > } > > # !! Nathelper > onreply_route[1] { > # NATed transaction ? > if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { > fix_nated_contact(); > force_rtp_proxy(); > # otherwise, is it a transaction behind a NAT and we did not > # know at time of request processing ? (RFC1918 contacts) > } else if (nat_uac_test("1")) { > fix_nated_contact(); > }; > } > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

20 years

[Serusers] Cisco 79xx authentication (off-topic to some extent)

by Niall O'Reilly

Apologies for the extent to which this is off-topic. I've seen related mails in the serusers archive, and expect that this community can at worst tell me where better to post. I have a Cisco 7960 on my desk, and can't manage to make it authenticate to either the iptel.org SIP server or to an Asterisk system where I'm also made welcome. I'm not seeing problems with X-Lite. Between Google, serusers archives, and the Cisco docs, I can't see what it is that I need to do. I don't yet have my own SIP server running, so I'm not in a position to make server-side experiments. Any help will be greatly appreciated. Thanks in advance. Best regards, Niall O'Reilly PGP key ID: AE995ED9 (see www.pgp.net) Fingerprint: 23DC C6DE 8874 2432 2BE0 3905 7987 E48D AE99 5ED9

20 years

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sr-users November 2004