sr-users July 2004

sr-users@lists.kamailio.org

178 participants
449 discussions

by Shi

Using ser 0.8.12 Script : if (uri=~"^sip:9") { strip(1); forward(192.168.10.13, 5060); }; for some reason it doesn't seem to work Shi Hoch

19 years, 10 months

[Serusers] SER & b2bua

by Derek Toner

Hi all, I have been experimenting with Ser and b2bua, and a Grandstream Budge Tone 100 SIP phone, although the same problem occurs for Cisco ATAs & softphones. When I try to make a call, the call will succeed, but there are a number of issues at the moment * it takes a long amount of time to connect, usually over 10 seconds. In some cases, it will connect even when the caller has disconnected * with b2bua in between the Ser and the SIP gateway, the return code is 484 - Address Incomplete * without b2bua, the return code is 408 - Request Timeout * the Ser gets into a loop when receiving an ACK. I have a feeling that these are all related, and could be down to the ser.cfg file. Below is the ser.cfg file. Any and all help would be appreciated. Regards, Derek # ------------------------- request routing logic ------------------- # main routing logic route{ # Standard max_forward_header checks... # A REGISTER is when a client attempts to register with a SIP Server if(method=="REGISTER") { if (!performRegistration("4ecalls.com")) { sl_send_reply("404", "Not Found"); break; }; save("location"); sl_send_reply("200", "OK"); }; # It is an attmept to make a call if(method=="INVITE") { sl_send_reply("100", "Trying.."); log("Attempting to re-route call ...\n"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # If we get to here, we intend to re-route throgh a PSTN network log("Lookup failed ...\n"); sl_send_reply("100", "Trying.."); if (performInvite("4ecalls.com")) { log("Am checking to see if it is in our domain...\n"); if(uri=~"^sip:0[0-9]*") { log("Am really attempting to re-route call \n"); rewritehostport("<SIP GATEWAY IP>:5060"); forward(<SIP GATEWAY IP>, 5060); # This should forward to the B2BUA server # rewritehostport("<b2bua IP>:5065"); # forward(<b2bua IP>, 5065); # log("Am done with that call ...\n"); # We may want to send a reply back here to prevent the client from # sending too many INVITE requests sl_send_reply("181", "Forwarding call"); }; } else { log("The call was not logged ...\n"); }; log("The call was redirected - hopefully ...\n"); } # Set the accounting flag - XXX do we actually need this? # setflag(1); }; # An ACK is when a call is acknowlowdged to have occured if(method=="ACK") { setStartTimeActiveCall(""); sl_send_reply("200", "OK"); }; # A CANCEL is when a call is cancelled in the middle of an attempt if(method=="CANCEL") { # Remove the active details... removeActiveCall(""); }; # A BYE is for when a call finishes, so what we would want to do here is to track these # messages, as these contain the info for the billing if(method=="BYE") { log("Found BYE message ...\n"); # This call here should slow down the dispatching of messages t_reply("100", "Trying to bill..."); # This will process the message and do all necessary work on it... # In this case, we just want to perform the BYE actions... log("Checking to see if the destination is a SIP phone or not ...\n"); # This lookup is here so we can determine if the call was to a SIP softphone or ATA # If it was a SIP Softphone or an ATA, then we won't need to bil, as this is over IP if (!lookup("location")) { log("Destination is a PSTN number ...\n"); # This call will bill both the provider and customer performBillingOperation("4ecalls.com"); } }; # 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") { record_route(); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; }

19 years, 10 months

[Serusers] STUN Server

by Marcello Lupo

Hi guys, sorry if a little bit OT... Your opinion/suggestion on the best and stable/working STUN server for linux to allow SIP clients behind NAT to work? Thanks, Bye, MArcello

19 years, 10 months

[Serusers] on_reply processing failed

by Andrea Girardi

hi to all when I try to send a message from Iptel to my Sip Account on my SIP Proxy, on log I can see this message: Jul 6 13:59:25 webby /usr/sbin/ser[28075]: method <MESSAGE> from <sip:captain.sipgiandrea@iptel.org> - <195.37.77.101> source address Jul 6 13:59:25 webby /usr/sbin/ser[28075]: request URI <giandrea(a)sip.quellidelpirana.it> - to URI <sip:giandrea@sip.quellidelpirana.it> Jul 6 13:59:25 webby /usr/sbin/ser[28075]: Red Route 1... Jul 6 13:59:25 webby /usr/sbin/ser[28079]: ERROR: on_reply processing failed Jul 6 13:59:25 webby /usr/sbin/ser[28079]: ERROR: on_reply processing failed Jul 6 13:59:25 webby /usr/sbin/ser[28075]: ERROR: on_reply processing failed why? what does it means? bye. Andrea

19 years, 10 months

RE: [Serusers] SER + Asterisk Voicmail forward on failure

by Muhammad Nasim

I added t_check_status as follows: failure_route[1] { if(t_check_status("487")){ log("\n MARKER IN FAILURE ROUTE START"); revert_uri(); rewritehostport("telth3.dyndns.org:5061"); append_branch(); log("\n MARKER IN FAILURE ROUTE AFTER APPEND BRANCH\n"); t_relay_to_udp("telth3.dyndns.org","5061"); log("\n MARKER IN FAILURE ROUTE END"); break; }; } Now it hangs up after 15 seconds as it should but no forwarding to voicemail happens.... Also where as before if the destination was busy forwarding to voicemail would occur but now it doesn't. Am I using this correctly? Muhammad Nasim TelAppliant Ltd Tel: 020 7740 3492 Int: +44 20 7740 3492 Main: 0845 004 4040 Fax: 0845 004 4041 www.telappliant.com TelAppliant is the UK's provider of IP telephony solutions for small and medium-sized businesses _____ The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying or distribution or any action taken or omitted to be taken in reliance on it is prohibited and may be unlawful. -----Original Message----- From: GR S [mailto:gr_sh2003@yahoo.com] Sent: 05 July 2004 14:27 To: Muhammad Nasim Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Asterisk Voicmail forward on failure Hi, --- Muhammad Nasim <muhammad(a)telappliant.com> wrote: > > There is a CANCEL message but then another INVITE is sent from SER for > some reason even though I do not dial again or anything. Try adding t_check_status("487") in your failure route. Regards, ===== Girish Gopinath <gr_sh2003(a)yahoo.com> __________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail

19 years, 10 months

[Serusers] SER+NAT+RTP PROXY

by Karunakar Chemudugunta

Hi, I would like to use SER as a voip proxy as part of our communitiy for voip services. Please advice me which versions are stable and tested to install with SER+ NAT+ RTP PROXY combinations. and please point me docs if any. Thanks in advance. Regards, krc

19 years, 10 months

Re: [Serusers] SER + Asterisk Voicmail forward on failure

by Muhammad Nasim

Ok I am calling from 8001 to 8002. Both phones have external ip addresses and are Snom 200s. I am enclosing the trace from the destination phone 8002. There is a CANCEL message but then another INVITE is sent from SER for some reason even though I do not dial again or anything. Received from udp:217.14.132.171:5060 at 2/7/2004 20:21:27:154 (1103 bytes): INVITE sip:8002@62.3.220.72:5060;line=xpyent8j SIP/2.0 Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bK0025.4c00bf37.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-ch6u5dfvgqq6 From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone> Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 1 INVITE Max-Forwards: 69 Contact: <sip:8001@80.254.236.54:5060;line=tttr4hak> P-Key-Flags: keys="3" User-Agent: snom200-2.05c Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 7200 Content-Type: application/sdp Content-Length: 283 v=0 o=root 1288172798 1288172798 IN IP4 80.254.236.54 s=call c=IN IP4 80.254.236.54 t=0 0 m=audio 10656 RTP/AVP 0 8 3 18 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ -------- Sent to udp:217.14.132.171:5060 at 2/7/2004 20:21:27:218 (618 bytes): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bK0025.4c00bf37.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-ch6u5dfvgqq6 Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=ufkt4pmqom Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 1 INVITE Contact: <sip:8002@62.3.220.72:5060;line=xpyent8j> Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ -------- Received from udp:217.14.132.171:5060 at 2/7/2004 20:21:27:368 (663 bytes): PRACK sip:8002@62.3.220.72:5060;line=xpyent8j SIP/2.0 Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKdf15.cbd9da25.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-8kfa6d9stdpq From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=ufkt4pmqom Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 2 PRACK Max-Forwards: 69 Contact: <sip:8001@80.254.236.54:5060;line=tttr4hak> RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ -------- Sent to udp:217.14.132.171:5060 at 2/7/2004 20:21:27:382 (463 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKdf15.cbd9da25.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-8kfa6d9stdpq Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=ufkt4pmqom Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 2 PRACK Contact: <sip:8002@62.3.220.72:5060;line=xpyent8j> Content-Length: 0 ------------------------------------------------------------------------ -------- Received from udp:217.14.132.171:5060 at 2/7/2004 20:21:41:980 (371 bytes): CANCEL sip:8002@62.3.220.72:5060;line=xpyent8j SIP/2.0 Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bK0025.4c00bf37.0 From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 To: <sip:8002@telth3.dyndns.org;user=phone> CSeq: 1 CANCEL User-Agent: Sip EXpress router(0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 ------------------------------------------------------------------------ -------- Sent to udp:217.14.132.171:5060 at 2/7/2004 20:21:42:001 (284 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bK0025.4c00bf37.0 From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=ufkt4pmqom Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 1 CANCEL Content-Length: 0 ------------------------------------------------------------------------ -------- Sent to udp:217.14.132.171:5060 at 2/7/2004 20:21:42:080 (417 bytes): SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bK0025.4c00bf37.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-ch6u5dfvgqq6 From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=ufkt4pmqom Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 1 INVITE Contact: <sip:8002@62.3.220.72:5060;line=xpyent8j> Content-Length: 0 ------------------------------------------------------------------------ -------- Received from udp:217.14.132.171:5060 at 2/7/2004 20:21:42:144 (1284 bytes): INVITE sip:8002@62.3.220.72:5060;line=xpyent8j SIP/2.0 Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKdf15.dbd9da25.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-tm87ieaevg8l From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone> Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 2 INVITE Max-Forwards: 69 Contact: <sip:8001@80.254.236.54:5060;line=tttr4hak> P-Key-Flags: keys="3" User-Agent: snom200-2.05c Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 7200 Proxy-Authorization: Digest username="8001",realm="asterisk",nonce="032718ae",uri="sip:8002@telth3.d yndns.org;user=phone",response="ad458c68cafe4662063a56d198dee0a0",algori thm=md5 Content-Type: application/sdp Content-Length: 283 v=0 o=root 1288172798 1288172798 IN IP4 80.254.236.54 s=call c=IN IP4 80.254.236.54 t=0 0 m=audio 10656 RTP/AVP 0 8 3 18 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ -------- Sent to udp:217.14.132.171:5060 at 2/7/2004 20:21:42:209 (618 bytes): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKdf15.dbd9da25.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-tm87ieaevg8l Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=5zof1ywigy Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 2 INVITE Contact: <sip:8002@62.3.220.72:5060;line=xpyent8j> Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ -------- Received from udp:217.14.132.171:5060 at 2/7/2004 20:21:42:355 (380 bytes): ACK sip:8002@62.3.220.72:5060;line=xpyent8j SIP/2.0 Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bK0025.4c00bf37.0 From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 To: <sip:8002@telth3.dyndns.org;user=phone>;tag=ufkt4pmqom CSeq: 1 ACK User-Agent: Sip EXpress router(0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 ------------------------------------------------------------------------ -------- Received from udp:217.14.132.171:5060 at 2/7/2004 20:21:42:389 (663 bytes): PRACK sip:8002@62.3.220.72:5060;line=xpyent8j SIP/2.0 Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKef15.0c2383b1.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-n0eybk6vz0ld From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=5zof1ywigy Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 3 PRACK Max-Forwards: 69 Contact: <sip:8001@80.254.236.54:5060;line=tttr4hak> RAck: 1 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ -------- Sent to udp:217.14.132.171:5060 at 2/7/2004 20:21:42:409 (463 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKef15.0c2383b1.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-n0eybk6vz0ld Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=5zof1ywigy Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 3 PRACK Contact: <sip:8002@62.3.220.72:5060;line=xpyent8j> Content-Length: 0 ------------------------------------------------------------------------ -------- Received from udp:217.14.132.171:5060 at 2/7/2004 20:21:42:428 (663 bytes): PRACK sip:8002@62.3.220.72:5060;line=xpyent8j SIP/2.0 Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKef15.1c2383b1.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-9xq7m10s97a8 From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=5zof1ywigy Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 3 PRACK Max-Forwards: 69 Contact: <sip:8001@80.254.236.54:5060;line=tttr4hak> RAck: 1 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ -------- Sent to udp:217.14.132.171:5060 at 2/7/2004 20:21:42:430 (463 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKef15.0c2383b1.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-n0eybk6vz0ld Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=5zof1ywigy Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 3 PRACK Contact: <sip:8002@62.3.220.72:5060;line=xpyent8j> Content-Length: 0 ------------------------------------------------------------------------ -------- Received from udp:217.14.132.171:5060 at 2/7/2004 20:21:42:969 (663 bytes): PRACK sip:8002@62.3.220.72:5060;line=xpyent8j SIP/2.0 Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKef15.1c2383b1.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-9xq7m10s97a8 From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=5zof1ywigy Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 3 PRACK Max-Forwards: 69 Contact: <sip:8001@80.254.236.54:5060;line=tttr4hak> RAck: 1 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ -------- Sent to udp:217.14.132.171:5060 at 2/7/2004 20:21:42:971 (463 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKef15.0c2383b1.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-n0eybk6vz0ld Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=5zof1ywigy Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 3 PRACK Contact: <sip:8002@62.3.220.72:5060;line=xpyent8j> Content-Length: 0 ------------------------------------------------------------------------ -------- Received from udp:217.14.132.171:5060 at 2/7/2004 20:21:44:991 (663 bytes): PRACK sip:8002@62.3.220.72:5060;line=xpyent8j SIP/2.0 Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKef15.1c2383b1.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-9xq7m10s97a8 From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=5zof1ywigy Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 3 PRACK Max-Forwards: 69 Contact: <sip:8001@80.254.236.54:5060;line=tttr4hak> RAck: 1 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ -------- Sent to udp:217.14.132.171:5060 at 2/7/2004 20:21:44:993 (463 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 217.14.132.171;branch=z9hG4bKef15.0c2383b1.0 Via: SIP/2.0/UDP 80.254.236.54:5060;branch=z9hG4bK-n0eybk6vz0ld Record-Route: <sip:8002@217.14.132.171;ftag=cdwfwaa0yn;lr=on> From: "8001" <sip:8001@telth3.dyndns.org>;tag=cdwfwaa0yn To: <sip:8002@telth3.dyndns.org;user=phone>;tag=5zof1ywigy Call-ID: 3c30253dbf80-rb6it812yf1g@80-254-236-54 CSeq: 3 PRACK Contact: <sip:8002@62.3.220.72:5060;line=xpyent8j> Content-Length: 0 Muhammad Nasim TelAppliant Ltd Tel: 020 7740 3492 Int: +44 20 7740 3492 Main: 0845 004 4040 Fax: 0845 004 4041 www.telappliant.com TelAppliant is the UK's provider of IP telephony solutions for small and medium-sized businesses _____ The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying or distribution or any action taken or omitted to be taken in reliance on it is prohibited and may be unlawful. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: 30 June 2004 17:01 To: Muhammad Nasim Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Asterisk Voicmail forward on failure use ethereal to watch if there is a CANCEL sent from ser to your UA after 15s. klaus Muhammad Nasim wrote: > Having searched the groups I have come up with the following in ser.cfg > to forward calls from SER to Asterisk voicemail on timeout. > > The system should work as follows. E.g dial 8003. Ring phone at 8003 for > 15 seconds then go to voicemail (at asterisk). > > Phenomena: > > 1. When the failure_route is exited though I can see some traffic > hitting asterisk, the original phone at 8003 continues to ring. > > 2. If I press Deny on the phone at 8003 TWICE then the call goes to > voicemail > > 3. If 8003 is on the offhook, call automatically goes to voicemail > > Points 2 and 3 are pretty much what I want. > > But in point 1, what am I doing wrong that doesn't cause the > t_on_failure to stop ringing the destination phone. > > I include the relevant part of my ser.cfg below > > ............................ > > > log("\nMARKER VOICEMAIL ATTEMPT STARTS HERE\n"); > t_on_failure("1");#send to voicemail now > log("\nMARKER VOICEMAIL ATTEMPT STOPS HERE\n"); > > ............................ > ............................ > > > failure_route[1] { > > > log("\n MARKER IN FAILURE ROUTE START"); > if(t_check_status("408")){ > log("\nMARKER We have hit second status 408\n"); > }; > revert_uri(); > rewritehostport("telth3.dyndns.org:5061"); > append_branch(); > log("\n MARKER IN FAILURE ROUTE AFTER APPEND BRANCH\n"); > t_relay_to_udp("telth3.dyndns.org","5061"); > log("\n MARKER IN FAILURE ROUTE END"); > break; > > > } > > ~ > > Muhammad Nasim > TelAppliant Ltd > Tel: 020 7740 3492 > Int: +44 20 7740 3492 > Main: 0845 004 4040 > Fax: 0845 004 4041 > www.telappliant.com > > > TelAppliant is the UK's provider of IP telephony solutions for small and > medium-sized businesses > > _____ > > The information in this message is confidential and may be legally > privileged. It is intended solely for the addressee. Access to this > message by anyone else is unauthorized. If you are not the intended > recipient, any disclosure, copying or distribution or any action taken > or omitted to be taken in reliance on it is prohibited and may be > unlawful. > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

19 years, 10 months

[Serusers] ser help

by Amen Holmes

We are a European voip service provider, and would like to know if anyone can provide any paid SER consultancy. If so, please contact me directly. --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage!

19 years, 10 months

[Serusers] problem with recording 'BYE' messages by using fxs

by varala ramakanth

hello friends, iam using welltech two port fxs my first problem iam facing is when calling party ends the session in acc table of ser database iam getting the bye record but when called party ( destination ) ends the session iam not getting the record in the acc table iam using xlite as softphone so when any fxs phone starts the session with the xlite only invite is being stored in the acc table iam attaching my config file please guide me with regards rama kanth ser.cfg __________________________________________________ debug=8 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (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) 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/mysql.so" loadmodule "/usr/local/lib/ser/modules/exec.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" loadmodule "/usr/local/lib/ser/modules/domain.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_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.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("auth_radius","radius_config","/root/radiusclient.conf") modparam("auth_radius","service_type",15) 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) modparam("acc", "radius_flag", 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; }; # !! 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 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") { 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); # 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 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") { #cdr tool if ( method == "REGISTER" || method=="INVITE" || method=="BYE" || method == "CANCEL" ) { # save messages in mysql for CDRTool exec_msg("/usr/local/etc/ser/sertrace.py; exit 1"); }; #end of cdrtool # Uncomment this if you want to use digest authentication if (uri==myself) { if (method=="REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; if (method=="INVITE") { log(1, "INVITE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if ( method=="BYE" || method=="CANCEL" ) { log (1, "BYE or CANCEL\n"); setflag(1); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # 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(); }; } here is my ser.config __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

19 years, 10 months

[Serusers] SER and Allied Telesyn VP604 FXO gateway

by Andrei Verovski

Hi, I have fully configured SER server (which works nice with soft and hard phones), and Allied Telesyn VoIPTalk VP604 FXO gateway (which seems to be the same as Mediatrix 1204 FXO gateway). The problem is that this combo does not work, and I could not figure out why. I have filled all config values in VoIPTalk VP604 (server address/login, gateway map, etc.). When I call to the gateway, I got ring, then it pick ups, and then no signal tone. It looks like gateway cannot reach or login SIP server. Server login do not show even failed login attempts from this gateway. Anyone have any experience with that hardware ? Any help will be greatly appreciated. ********************************************* * Best Regards --- Andrei Verovski * * Personal Home Page * http://snow.prohosting.com/guru4mac/ * Mac, Linux, DTP, Development, IT WEB Site *********************************************

19 years, 10 months

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sr-users July 2004