[Serusers] One failure_route problem remaining
Greger V. Teigre
greger at teigre.com
Thu Sep 8 10:11:45 CEST 2005
Hi Corey,
Good to hear that the rewrite helped you out. I'll comment inline.
Corey S. McFadden wrote:
> Guys,
>
> First, thanks again to those who have offered advice. I have
> re-started
> building from the ONsip.org rtpproxy example config and have
> everything
> working properly except one thing.
>
> In the interest of being thorough, I kept a log of what I changed and
> why:
>
> - Tested REGISTER, works OK.
>
> - Calls between UAs initially failed when proxy_authorize in route[3]
> was called. Disabled proxy_authorize checks. Works OK. (Need to
> revisit this one for security. Can anyone point out the problem?)
If your UACs can successfully register, proxy_authorize should also work.
The INVITE should be authenticated to avoid non-registered users to make
calls.
> - Implemented failure_route for voicemail on timeout. Timeout calls
> from UAs work fine to VM via Asterisk.
>
> - Implemented 7 digit dial perl script and added second lookup. No
> problems. Added NPANXX route lookup script in route[3]. No
> problems.
>
> - Implemented "Your call cannot be completed as dialed" via Asterisk
> server by replacing sl_send_reply 404 in route[3] with route(7). No
> problems.
>
> - Initially was unable to dial voicemail access number from UA. It
> would connect to Asterisk server but not have audio. Call would
> drop. Fixed by adding this to route[1]:
>
> # NAT Fix
> if (method=="INVITE" && nat_uac_test("19")) {
> force_rport();
> fix_nated_contact();
> force_rtp_proxy();
> };
You will be better off putting the NAT code in a route (without a break) and
then call the route as a function.
If you look at the nat-rtpproxy config, you will see that route[3] has logic
that should be able to handle this.
> - Voicemail access is now OK.
>
> - UA timeout calls to voicemail now fail. Debug on Asterisk looks
> like the contact is being fixed twice. Added flag(9) to make sure
> NAT fix only happens once. UA timeout calls now go to voicemail
> properly.
This has probably something to do with your added NAT test.
>
> - I notice a lot of "Warning: sl_send_reply: I won't send a reply for
> ACK!!" but don't know if this is significant or not. From what I've
> read it sounds like ACKs are getting an sl_reply rather than being
> t_relayed but I didn't really modify anything related ... ?
Yes, this error tells you that ACKs end in an sl_reply, which they
shouldn't. You need to identify the type of ACKs (probably related to your
GW, as a guess), so you can make sure the ACKs are handled correctly.
>
> Now, the biggest problem seems to be calls coming from the inbound SIP
> gateway aren't being handled properly on timeout. (Calls from the
> inbound
> SIP gateway work fine to UAs.) This very well may be the fault of the
> other equipment, but it's nothing we can debug directly so I want to
> make
> sure there's nothing else wrong.
Could have something to do with the ACKs. But you don't describe very well
what "aren't being handled properly on timeout" means. I would advice to
get rid of all error messages before trying to figure out what is happening
here. Maybe the error disappears...
g-)
> Keep in mind, the inbound SIP gateway CAN communicate with the
> Asterisk
> server by dialing a DID that is directed to voicemail. I thought it
> might
> be some kind of routing problem between those boxes, but this would
> seem
> to disprove that.
>
> I'm going to include the abbreviated ngrep dump from the SER machine
> and
> then the ser.cfg file.
>
> Thanks again for the help thus far and in advance for any assistance
> on
> this.
>
> -Corey
>
>
>
> Here is an IP legend:
> xxx.xxx.xxx.7 - Inbound SIP Gateway
> xxx.xxx.xxx.36 - SER Server
> xxx.xxx.xxx.80 - Asterisk Server
> 216.xxx.ua.207 - NATed UA public IP
>
>
> ** NGREP DUMP FROM SER SERVER
> #
> U xxx.xxx.xxx.7:5060 -> xxx.xxx.xxx.36:5060
> INVITE sip:5414362504 at xxx.xxx.xxx.36:5060 SIP/2.0..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B #
> U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
> SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41 #
> U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:7565
> INVITE sip:5414362504 at 216.xxx.ua.207:5060 SIP/2.0..Record-Route:
> <sip:xxx.xxx.xxx.36;ftag=Xvn755Ml3U1j41B-IPTrunk-67-17-23at #
> U 216.xxx.ua.207:7565 -> xxx.xxx.xxx.36:5060
> SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0,SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;rport=5060 #
> U 216.xxx.ua.207:7565 -> xxx.xxx.xxx.36:5060
> SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0,SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;rport=506 #
> U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
> SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;rport=5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..From:
> sip:61087 # # *** Call is diverted to Asterisk server here.
> #
> U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.80:5060
> INVITE sip:5414362504 at xxx.xxx.xxx.80:5060 SIP/2.0..Record-Route:
> <sip:xxx.xxx.xxx.36;ftag=Xvn755Ml3U1j41B-IPTrunk-67-17-23a #
> U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:7565
> CANCEL sip:5414362504 at 216.xxx.ua.207:5060 SIP/2.0..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0..From: sip: #
> U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
> SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;bran #
> U 216.xxx.ua.207:7565 -> xxx.xxx.xxx.36:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0..From:
> sip:6108712381 at xxx.xxx.xxx.7:5060;tag= #
> U 216.xxx.ua.207:7565 -> xxx.xxx.xxx.36:5060
> SIP/2.0 487 Request Cancelled..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0,SIP/2.0/UDP
> xxx.xxx.xxx.7:5060 #
> U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:7565
> ACK sip:5414362504 at 216.xxx.ua.207:5060 SIP/2.0..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.0..From: sip:610 #
> U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:5060
> ....
> #
> U xxx.xxx.xxx.36:5060 -> 216.xxx.ua.207:7565
> ....
> #
> U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z #
> U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..Record-Route:
> <sip:209.216.160. #
> U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z #
> U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..Record-Route:
> <sip:209.216.160. # # Several OKs seem to be going through here...
> U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z #
> U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..Record-Route:
> <sip:209.216.160. #
> U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> xxx.xxx.xxx.36;branch=z9hG4bKe432.bd7822d3.1..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z #
> U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-Xvn755Ml3U1j41B-0..Record-Route:
> <sip:209.216.160. # # Call is dropped by now.
>
>
>
>
> ** Asterisk console sip debug
>
> <-- SIP read from xxx.xxx.xxx.36:5060:
> INVITE sip:5414362504 at xxx.xxx.xxx.80:5060 SIP/2.0
> Record-Route:
> <sip:xxx.xxx.xxx.36;ftag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7;lr=on>
> Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKef17.c97ef7d2.1
> Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;rport=5060;branch=z9hG4bK-00101-AJn351hl3U1H58O-0
> Max-Forwards: 5
> From:
> sip:6108712381 at xxx.xxx.xxx.7:5060;tag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7
> To: sip:5414362504 at xxx.xxx.xxx.36:5060
> Call-ID: AJn351hl3U1H58O at xxx.xxx.xxx.7
> CSeq: 101 INVITE
> Remote-Party-Id:
> <sip:6108712381 at xxx.xxx.xxx.7:5060>;party=calling;privacy=off;id-type=subscriber
> Expires: 180
> Allow: INVITE,CANCEL,BYE,ACK
> Contact: sip:xxx.xxx.xxx.7:5060
> User-Agent: Tekelec-7000/r4.0
> Content-Type: application/sdp
> Content-Length: 149
> P-hint: OFFLINE VOICEMAIL
>
> v=0
> o=- 1 1 IN IP4 xxx.xxx.xxx.7
> s=
> c=IN IP4 xxx.xxx.xxx.36
> t=0 0
> m=audio 35842 RTP/AVP 0
> a=ptime:20
> a=rtpmap:0 PCMU/8000
> a=nortpproxy:yes
>
>
> --- (17 headers 9 lines)---
> Using INVITE request as basis request - AJn351hl3U1H58O at xxx.xxx.xxx.7
> Sending to xxx.xxx.xxx.36 : 5060 (non-NAT)
> Found no matching peer or user for 'xxx.xxx.xxx.36:5060'
> Found RTP audio format 0
> Peer audio RTP is at port xxx.xxx.xxx.36:35842
> Found description format PCMU
> Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4
> (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
> (nothing), combined - 0x0 (nothing)
> Looking for 5414362504 in sip
> list_route: hop:
> <sip:xxx.xxx.xxx.36;ftag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7;lr=on>
> list_route: hop: <sip:xxx.xxx.xxx.7:5060>
> Transmitting (no NAT) to xxx.xxx.xxx.36:5060:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKef17.c97ef7d2.1
> Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-AJn351hl3U1H58O-0
> From:
> sip:6108712381 at xxx.xxx.xxx.7:5060;tag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7
> To: sip:5414362504 at xxx.xxx.xxx.36:5060
> all-ID: AJn351hl3U1H58O at xxx.xxx.xxx.7
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:5414362504 at xxx.xxx.xxx.80>
> Content-Length: 0
>
>
> ---
> -- Executing Wait("SIP/5060-b7d05f68", "1") in new stack
> -- Executing VoiceMail("SIP/5060-b7d05f68", "5414362504 at sip") in
> new stack
> We're at xxx.xxx.xxx.80 port 14018
> Answering with preferred capability 0x4 (ulaw)
> Answering with preferred capability 0x2 (gsm)
> Answering with preferred capability 0x8 (alaw)
> Answering with preferred capability 0x400 (ilbc)
> Reliably Transmitting (no NAT) to xxx.xxx.xxx.36:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKef17.c97ef7d2.1
> Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-AJn351hl3U1H58O-0
> Record-Route:
> <sip:xxx.xxx.xxx.36;ftag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7;lr=on>
> From:
> sip:6108712381 at xxx.xxx.xxx.7:5060;tag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7
> To: sip:5414362504 at xxx.xxx.xxx.36:5060;tag=as4c6bfaee
> Call-ID: AJn351hl3U1H58O at xxx.xxx.xxx.7
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:5414362504 at xxx.xxx.xxx.80>
> Content-Type: application/sdp
> Content-Length: 237
>
> v=0
> o=root 29275 29275 IN IP4 xxx.xxx.xxx.80
> s=session
> c=IN IP4 xxx.xxx.xxx.80
> t=0 0
> m=audio 14018 RTP/AVP 0 3 8 97
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> a=silenceSupp:off - - - -
>
> ---
> -- Playing 'vm-intro' (language 'en')
> Retransmitting #1 (no NAT) to xxx.xxx.xxx.36:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKef17.c97ef7d2.1
> Via: SIP/2.0/UDP
> xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-AJn351hl3U1H58O-0
> Record-Route:
> <sip:xxx.xxx.xxx.36;ftag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7;lr=on>
> From:
> sip:6108712381 at xxx.xxx.xxx.7:5060;tag=AJn351hl3U1H58O-IPTrunk-70-17-23atxxx.xxx.xxx.7
> To: sip:5414362504 at xxx.xxx.xxx.36:5060;tag=as4c6bfaee
> Call-ID: AJn351hl3U1H58O at xxx.xxx.xxx.7
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:5414362504 at xxx.xxx.xxx.80>
> Content-Type: application/sdp
> Content-Length: 237
>
> v=0
> o=root 29275 29275 IN IP4 xxx.xxx.xxx.80
> s=session
> c=IN IP4 xxx.xxx.xxx.80
> t=0 0
> m=audio 14018 RTP/AVP 0 3 8 97
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> a=silenceSupp:off - - - -
>
> ---
>
>
>
>
>
> ** ser.cfg
>
> # whatever.net - SER Configuration File
> #
>
> debug=3
> fork=yes
> log_stderror=yes
>
>
> listen=xxx.xxx.xxx.36
> port=5060
> children=4
> alias=whatever.net
> alias=hostname.whatever.net
>
> dns=no
> rev_dns=no
>
> fifo="/tmp/ser_fifo"
> fifo_db_url="mysql://ser:serpass@localhost/ser"
>
> # Modules
>
> 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/nathelper.so"
> loadmodule "/usr/local/lib/ser/modules/textops.so"
> loadmodule "/usr/local/lib/ser/modules/uri_db.so"
> loadmodule "/usr/local/lib/ser/modules/uri.so"
> loadmodule "/usr/local/lib/ser/modules/xlog.so"
> loadmodule "/usr/local/lib/ser/modules/options.so"
> loadmodule "/usr/local/lib/ser/modules/exec.so"
>
>
> modparam("auth_db|uri_db|usrloc|permissions", "db_url",
> "mysql://ser:serpass@localhost/ser")
>
> modparam("auth_db", "calculate_ha1", 1)
> modparam("auth_db", "password_column", "password")
>
> modparam("nathelper", "natping_interval", 30)
> modparam("nathelper", "ping_nated_only", 1)
> modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
>
> modparam("usrloc", "db_mode", 2)
>
> modparam("registrar", "nat_flag", 6)
>
> modparam("rr", "enable_full_lr", 1)
>
> modparam("tm", "fr_inv_timer", 20)
> modparam("tm", "fr_timer", 20)
>
>
>
> #
> # Route Section
> #
>
> route {
>
> # Basic Checks
> 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();
> };
>
> if (method=="BYE" || method=="CANCEL") {
> unforce_rtp_proxy();
> }
>
> ##
> # Loose Route Section
> ##
>
> if (loose_route()) {
> if (has_totag() && (method=="INVITE" ||
> method=="ACK")) { if (nat_uac_test("19")) {
> setflag(6);
> force_rport();
> fix_nated_contact();
> };
> force_rtp_proxy("l");
> };
> route(1);
> break;
> };
>
> ##
> # Message Processing Section
> ##
>
> if (uri!=myself) {
> route(1);
> break;
> };
>
> if (uri==myself) {
>
> if (method=="CANCEL") {
> route(3);
> break;
> } else if (method=="INVITE") {
> route(3);
> break;
> } else if (method=="REGISTER") {
> route(2);
> break;
> } else if (method=="NOTIFY") {
> sl_send_reply("200", "Shut up.");
> break;
> } else if (method=="OPTIONS" && (!
> uri=~"sip:.*[@]+.*")) { options_reply();
> break;
> };
>
> lookup("aliases");
> if (uri!=myself) {
> route(1);
> break;
> };
>
> if (!lookup("location") && method!="ACK") {
> sl_send_reply("404", "User Not Found");
> break;
> };
> };
>
> route(1);
> }
>
> route[1] {
>
> #
>
>
> ----------------------------------------------------------------- #
> Default Message Handler #
> -----------------------------------------------------------------
>
> # NAT Fix
> if (method=="INVITE" && nat_uac_test("19") && !isflagset(9)) {
> xlog("L_ERR", "Fixing NAT on INVITE for %ct.\n");
> force_rport();
> fix_nated_contact();
> force_rtp_proxy();
> setflag(9);
> };
>
> t_on_reply("1");
>
> if (!t_relay()) {
> if (method=="INVITE" && isflagset(6)) {
> unforce_rtp_proxy();
> };
> sl_reply_error();
> };
> }
>
> route[2] {
>
> #
>
>
> ----------------------------------------------------------------- #
> REGISTER Message Handler #
> ----------------------------------------------------------------
>
> if (!search("^Contact:\ +\*") && nat_uac_test("19")) {
> setflag(6);
> fix_nated_register();
> force_rport();
> };
>
> sl_send_reply("100", "Trying");
>
> if (!www_authorize("whatever.net","subscriber")) {
> www_challenge("whatever.net","0");
> break;
> };
>
> if (!check_to()) {
> sl_send_reply("401", "Unauthorized");
> break;
> };
>
> consume_credentials();
>
> if (!save("location")) {
> sl_reply_error();
> };
> }
>
> route[3] {
>
> #
>
>
> ----------------------------------------------------------------- #
> CANCEL and INVITE Message Handler #
> -----------------------------------------------------------------
>
> if (nat_uac_test("19")) {
> setflag(6);
> }
>
> lookup("aliases");
> if (uri!=myself) {
> route(1);
> break;
> };
>
> if (!lookup("location")) {
>
> if (uri=~"^sip:911@") { # Pass 911 calls first
> route(5);
> break;
> };
> if (uri=~"^sip:[0-9]{7}@") { # EXPAND 7-DIGIT CALL
> exec_dset("/root/7digit.pl");
> };
>
> # Try location lookup again
>
> if (!lookup("location")) {
>
> if (uri=~"^sip:[0-9]{10}@") {
> # DB Lookup of NPANXX route
> exec_dset("/root/npa_nxx.pl");
> route(1);
> break;
> };
> if (uri=~"^sip:[0-9]{11}@") {
> # DB Lookup of NPANXX route
> strip(1);
> exec_dset("/root/npa_nxx.pl");
> route(1);
> break;
> };
> if (uri=~"^sip:555@") { # Voicemail Access
> route(6);
> break;
> };
> if (uri=~"^sip:556@") { # Voicemail Access
> route(6);
> break;
> };
>
> # Call error message
> route(7);
> # sl_send_reply("404", "User Not Found");
> break;
> };
> };
>
> # if (!proxy_authorize("whatever.net","subscriber")) {
> # proxy_challenge("whatever.net","0");
> # break;
> # } else if (!check_from()) {
> # sl_send_reply("403", "Use From=ID");
> # break;
> # };
> # consume_credentials();
>
> if (isflagset(6) && !isflagset(9)) {
> force_rport();
> fix_nated_contact();
> force_rtp_proxy();
> setflag(9);
> };
>
> t_on_reply("1");
> t_on_failure("1");
>
> if (!t_relay()) {
> if(isflagset(6)) {
> unforce_rtp_proxy();
> };
> sl_reply_error();
> };
> }
>
> route[5] {
>
> ###
> # Local PSTN Gateway
> ###
> rewritehost("xxx.xxx.xxx.7");
> route(1);
> }
>
>
> route[6] {
>
> ###
> # Calls to VM
> ###
> rewritehost("xxx.xxx.xxx.80");
> route(1);
> }
>
> route[7] {
>
> ###
> # Call Failure
> ###
> rewriteuri("sip:7110000001 at xxx.xxx.xxx.80");
> route(1);
> }
>
>
>
> failure_route[1] {
>
> ###
> # Voicemail
> ###
> xlog("L_ERR", " Failure Route 1 hit.\n");
>
> if (t_check_status("487")) {
> xlog("L_ERR", " Failure Route 1 - 487 detected, breaking.\n");
> break;
> };
>
> rewritehost("xxx.xxx.xxx.80");
> append_hf("P-hint: OFFLINE VOICEMAIL\r\n");
> append_branch();
> route(1);
> }
>
>
>
> onreply_route[1] {
>
> if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
> if (!search("^Content-Length:\ +0")) {
> force_rtp_proxy();
> };
> };
>
> if (nat_uac_test("1")) {
> fix_nated_contact();
> };
> }
>
>
> *********************************************
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