April 2004 - sr-users - kamailio.org

[Serusers] Asterisk/Digium as PSTN gateway and SER as backend soft switch?
by Tony Lum 12 Jul '04

12 Jul '04

Hi, Has any one done using Asterisk/Digium T1 card as PSTN gateway, and SER as the backend soft switch? I am looking to install Asterisk boxes in a few cities to act as PSTN gateways and using SER to accept these calls. SER will have to communicate with other long distance carriers to allow world wide calling. Will this combination work? I like to know your comments or suggestions on this combination. thanks Tony Lum

9 14

[Serusers] call logs
by Adrian Georgescu 04 May '04

04 May '04

You have several choices (like always on Unix): 1. Do like this: if (method=="INVITE") { # save messages in mysql for CDRTool exec_msg("/etc/ser/sertrace.py; exit 1"); }; Make your own script to replace /etc/ser/sertrace.py You can log the whole SIP packet from STDIN to mysql and index it by environment variable set by SER (SIP_HF_CALLID). To scale it up a bit you demonize the logger to reuse only one mysql connection. 2. Log all SIP packets to a log file with a script like this one: <<<<<<<<<<<<<<<<<<<<<<<<<<<< #!/bin/sh # # chkconfig: 2345 20 80 # description: controls execution of SIP sniffer # processname: ng NAME=ngrep OPTIONS="-L -d bond0 -t port 5060" BIN=/usr/bin/ngrep ROTATE=/usr/sbin/rotatelogs RETVAL=0 # See how we were called. case "$1" in start) killall $NAME echo -n "Starting SIP sniffer: " $BIN $OPTIONS 2>&1 | $ROTATE /var/log/ser/trace/sip 86400 & RETVAL=$? echo "ngrep." ;; stop) echo -n "Stopping SIP ngrep: " killall $NAME RETVAL=$? echo "done." ;; restart) #$0 stop $0 start RETVAL=$? ;; *) echo "Usage: $0 {start|stop|restart}" exit 1 esac exit $RETVAL <<<<<<<<<<<<<<<<<<<<<<<< -- Adrian How to get the call log and registration request in a different file other that /var/log/messages. Also, can i get some more details on autherisation and accountingpackets. Rt now, iam getting only details given below. ACC: transaction answered: method=BY E, i-uri=sip:12334 at 10.10.0.1, o-uri=sip:12334 at 10.10.0.1, call_id =4F6F069B-4BF4-4CC7-B99C-4E24F0E2C667 at 192.168.10.154, from=Manu Sip <sip:manu at s ip.test.net>;tag=2654486816, code=200 thank you. MANU.

2 1

[Serusers] using voicemail
by Andrea Bondavalli 03 May '04

03 May '04

Hi all, I am configuring SER in order to use the SEMS's voicemail when one of my subscribed users is not avaiable or when all the INVITEs to their contacts fail (fork). I'am using the following configuration: route { .... if (isflagset(4)) { /*voicemail enabled*/ t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); break; } } failure_route[1] { t_reply("100","Trying -- going to voicemail"); if (method=="INVITE" || method=="REFER") { log("**************** vm start - begin ******************\n"); log("**************** voicemail ******************\n"); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm start - end ******************\n"); } else if (method=="BYE") { log("**************** vm end - begin ******************\n"); log("**************** voicemail ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm end - end ******************\n"); }; } ... This works fine when the system calls a user's contact whose username part matches the subscribed user name. Unfortunately, if the system tries to reach such a contact at a phone number (using a gateway) the result is that SEMS reports: "Error: 404 voicemail: no email address for user 034823213213." Can you help me ? Best regards, Andrea

2 4

[Serusers] PSTN termination
by radan 30 Apr '04

30 Apr '04

Hi ! I have a problem with correct accounting for calls to the PSTN SIP UA implemented inside gateway (Mediatrix 2402 ) sends ACK to SER even if the connection wasn't estabilished (no answer or reject by second site). in this way I have a few second in my stats for not estabilished connections. Where is problem ? Inside SER config, gateway config or PBX config ?? How do you resolve this problem ? Thanks Andrzej

2 1

[Serusers] Call drops with 60 sec and Buffer size exceeded message
by Jadylson Bomfim 30 Apr '04

30 Apr '04

Hi guys, I was create a script to route some kinds of routes calls to PSTN ( like that ). So after 59 sec. it is droped. Someone can help me ? Thank you all for attention and kindness in advance. Jadylson Bomfim if (uri=~"sip:0[0-9]*@.*") { route(6); break; }; route[6] { setflag(1); if (uri=~"^sip:0[1-9]*@") { strip(1); prefix("01155"); rewritehostport("ip address:5060"); acc_rad_request("Saindo Para PSTN no Brasil"); if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("ip address","5060"); } else { if (uri=~"^sip:00[0-9][0-9]*@") { # ... forward to gateways then; #Cheque se inicia com 00 if (uri=~"^sip:00") { # strip the leading "1" strip(2); prefix("011"); # if you have passed through all the checks, let your call go to GW! rewritehostport("ip address:5060"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("ip address","5060"); }; } else { if (uri=~"^sip:001*@") { # ... forward to gateways then; #Cheque se inicia com 00 if (uri=~"^sip:00") { # strip the leading "1" strip(2); # if you have passed through all the checks, let your call go to GW! rewritehostport("ip address:5060"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("ip address","5060"); }; }; }; }; } Jadylson da Rocha Passos Bomfim Redevox Telecom Uberlandia +55 34 3234-7813 São Paulo +55 11 5055-6888 Móvel +55 34 9103-6854 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.643 / Virus Database: 411 - Release Date: 25/3/2004

1 0

[Serusers] Problem with registration of the SIP agent
by bbleuez＠numericable.fr 30 Apr '04

30 Apr '04

Hello, I am Benoit and I am working on the SER and the server and the database is working well. The problem is when I try to registrer SIP agent, the error message is "Error 483:" Too Many Hops". In the file ser.cfg there is a function mf_process_maxforward_header("10"), and the problem is that I don't know where is this function and how does it testing the max forward header because the SIP agent that I use set this field to 70. Kind regards Benoit

4 4

[Serusers] problem with BYE
by Vitaly Nikolaev 30 Apr '04

30 Apr '04

Hi all, I have very strange problem with SER. I am making incoming call from PSTN to SER, then number translation (exec_dset) from 12124051759 -> 250000007 And then it hitting my ATA (same with motorolla, innomedia) When I then hung up on ATA side it sends BYE (which goes as loose routing) to SER -> CISCO, CISCO answer OK to SER then to ATA but ATA looks like does not like this OK... sending BYE again and again I have attached config to the letter.. pleas look at it and may be you will have some correction PS: Absolutely same problem I have with incoming call via TCP from another provider Thank you! [root@ser etc]# ngrepcr -q 12124051759 U 209.227.182.44:51835 -> 209.227.182.43:5060 INVITE sip:12124051759@209.227.182.43:5060 SIP/2.0 Via: SIP/2.0/UDP 209.227.182.44:5060 From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43> Date: Fri, 30 Apr 2004 15:30:12 GMT Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 459440422-2582778328-2951146144-256312224 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:2126882781@209.227.182.44>;party=calling;screen=yes;privacy=off Timestamp: 1083339012 Contact: <sip:2126882781@209.227.182.44:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 426 v=0 o=CiscoSystemsSIP-GW-UserAgent 2183 5028 IN IP4 209.227.182.44 s=SIP Call c=IN IP4 209.227.182.44 t=0 0 m=audio 19266 RTP/AVP 18 4 0 100 101 19 c=IN IP4 209.227.182.44 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=direction:active U 209.227.182.43:5060 -> 209.227.182.44:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 209.227.182.44:5060 From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43> Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 101 INVITE Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 209.227.182.43:5060 "Noisy feedback tells: pid=6791 req_src_ip=209.227.182.44 req_src_port=51835 in_uri=sip:1212 4051759@209.227.182.43:5060 out_uri=sip:250000007@216.236.137.226:1102;user=phone;transport=udp via_cnt==1" U 209.227.182.43:5060 -> 216.236.137.226:1102 INVITE sip:250000007@216.236.137.226:1102;user=phone;transport=udp SIP/2.0 Record-Route: <sip:12124051759@209.227.182.43;ftag=DC7456A0-3FE;lr=on> Via: SIP/2.0/UDP 209.227.182.43;branch=z9hG4bK8198.2b7612d2.0 Via: SIP/2.0/UDP 209.227.182.44:5060 From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43> Date: Fri, 30 Apr 2004 15:30:12 GMT Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 459440422-2582778328-2951146144-256312224 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 5 Remote-Party-ID: <sip:2126882781@209.227.182.44>;party=calling;screen=yes;privacy=off Timestamp: 1083339012 Contact: <sip:2126882781@209.227.182.44:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 426 P-hint: request forced to rtp proxy v=0 o=CiscoSystemsSIP-GW-UserAgent 2183 5028 IN IP4 209.227.182.44 s=SIP Call c=IN IP4 209.227.182.43 t=0 0 m=audio 55824 RTP/AVP 18 4 0 100 101 19 c=IN IP4 209.227.182.43 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=direction:active U 216.236.137.226:1102 -> 209.227.182.43:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.227.182.43;branch=z9hG4bK8198.2b7612d2.0 Via: SIP/2.0/UDP 209.227.182.44:5060 From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43>;tag=3720024524 Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 101 INVITE Timestamp: 1083339012 Server: Cisco ATA 186 v3.0.0 atasip (031210A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 U 216.236.137.226:1102 -> 209.227.182.43:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.227.182.43;branch=z9hG4bK8198.2b7612d2.0 Via: SIP/2.0/UDP 209.227.182.44:5060 From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43>;tag=3720024524 Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 101 INVITE Timestamp: 1083339012 Server: Cisco ATA 186 v3.0.0 atasip (031210A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 U 209.227.182.43:5060 -> 209.227.182.44:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.227.182.44:5060 From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43>;tag=3720024524 Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 101 INVITE Timestamp: 1083339012 Server: Cisco ATA 186 v3.0.0 atasip (031210A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 P-hint: fixed NAT contact for response U 216.236.137.226:1102 -> 209.227.182.43:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 209.227.182.43;branch=z9hG4bK8198.2b7612d2.0 Via: SIP/2.0/UDP 209.227.182.44:5060 Record-Route: <sip:12124051759@209.227.182.43;ftag=DC7456A0-3FE;lr=on> From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43>;tag=3720024524 Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 101 INVITE Timestamp: 1083339012 Contact: <sip:250000007@172.30.200.59:5060;user=phone;transport=udp> Server: Cisco ATA 186 v3.0.0 atasip (031210A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 204 Content-Type: application/sdp v=0 o=250000007 85231 85231 IN IP4 172.30.200.59 s=ATA186 Call c=IN IP4 172.30.200.59 t=0 0 m=audio 16386 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 U 209.227.182.43:5060 -> 209.227.182.44:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 209.227.182.44:5060 Record-Route: <sip:12124051759@209.227.182.43;ftag=DC7456A0-3FE;lr=on> From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43>;tag=3720024524 Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 101 INVITE Timestamp: 1083339012 Contact: <sip:250000007@216.236.137.226:1102;user=phone;transport=udp> Server: Cisco ATA 186 v3.0.0 atasip (031210A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 205 Content-Type: application/sdp P-hint: fixed NAT contact for response P-hint: response forced to rtp proxy v=0 o=250000007 85231 85231 IN IP4 172.30.200.59 s=ATA186 Call c=IN IP4 209.227.182.43 t=0 0 m=audio 55824 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 U 209.227.182.44:57102 -> 209.227.182.43:5060 ACK sip:12124051759@209.227.182.43:5060;ftag=DC7456A0-3FE;lr=on SIP/2.0 Via: SIP/2.0/UDP 209.227.182.44:5060 From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43>;tag=3720024524 Date: Fri, 30 Apr 2004 15:30:12 GMT Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 Route: <sip:250000007@216.236.137.226:1102;user=phone;transport=udp> Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK U 209.227.182.43:5060 -> 216.236.137.226:1102 ACK sip:250000007@216.236.137.226:1102;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 209.227.182.43;branch=0 Via: SIP/2.0/UDP 209.227.182.44:5060 From: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE To: <sip:12124051759@209.227.182.43>;tag=3720024524 Date: Fri, 30 Apr 2004 15:30:12 GMT Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 Max-Forwards: 5 Content-Length: 0 CSeq: 101 ACK U 216.236.137.226:1102 -> 209.227.182.43:5060 BYE sip:12124051759@209.227.182.43 SIP/2.0 Route: <sip:2126882781@209.227.182.44> Via: SIP/2.0/UDP 172.30.200.59:5060 From: sip:12124051759@209.227.182.43;tag=3720024524 To: sip:2126882781@209.227.182.44;tag=DC7456A0-3FE Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 1 BYE User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A) Content-Length: 0 U 209.227.182.43:5060 -> 209.227.182.44:5060 BYE sip:2126882781@209.227.182.44 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 209.227.182.43;branch=z9hG4bKb7a6.78fba574.0 Via: SIP/2.0/UDP 172.30.200.59:5060;rport=1102;received=216.236.137.226 From: sip:12124051759@209.227.182.43;tag=3720024524 To: sip:2126882781@209.227.182.44;tag=DC7456A0-3FE Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 1 BYE User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A) Content-Length: 0 U 209.227.182.44:5060 -> 209.227.182.43:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 209.227.182.43;branch=z9hG4bKb7a6.78fba574.0,SIP/2.0/UDP 172.30.200.59:5060;rport=1102;received=216.236.137.2 26 From: <sip:12124051759@209.227.182.43>;tag=3720024524 To: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE Date: Fri, 30 Apr 2004 15:30:21 GMT Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 1 BYE U 209.227.182.43:5060 -> 216.236.137.226:1102 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.200.59:5060;rport=1102;received=216.236.137.226 From: <sip:12124051759@209.227.182.43>;tag=3720024524 To: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE Date: Fri, 30 Apr 2004 15:30:21 GMT Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 1 BYE U 216.236.137.226:1102 -> 209.227.182.43:5060 BYE sip:12124051759@209.227.182.43 SIP/2.0 Route: <sip:2126882781@209.227.182.44> Via: SIP/2.0/UDP 172.30.200.59:5060 From: sip:12124051759@209.227.182.43;tag=3720024524 To: sip:2126882781@209.227.182.44;tag=DC7456A0-3FE Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 1 BYE User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A) Content-Length: 0 U 209.227.182.43:5060 -> 216.236.137.226:1102 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.200.59:5060;rport=1102;received=216.236.137.226 From: <sip:12124051759@209.227.182.43>;tag=3720024524 To: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE Date: Fri, 30 Apr 2004 15:30:21 GMT Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 1 BYE U 216.236.137.226:1102 -> 209.227.182.43:5060 BYE sip:12124051759@209.227.182.43 SIP/2.0 Route: <sip:2126882781@209.227.182.44> Via: SIP/2.0/UDP 172.30.200.59:5060 From: sip:12124051759@209.227.182.43;tag=3720024524 To: sip:2126882781@209.227.182.44;tag=DC7456A0-3FE Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 1 BYE User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A) Content-Length: 0 U 209.227.182.43:5060 -> 216.236.137.226:1102 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.200.59:5060;rport=1102;received=216.236.137.226 From: <sip:12124051759@209.227.182.43>;tag=3720024524 To: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE Date: Fri, 30 Apr 2004 15:30:21 GMT Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 1 BYE U 216.236.137.226:1102 -> 209.227.182.43:5060 BYE sip:12124051759@209.227.182.43 SIP/2.0 Route: <sip:2126882781@209.227.182.44> Via: SIP/2.0/UDP 172.30.200.59:5060 From: sip:12124051759@209.227.182.43;tag=3720024524 To: sip:2126882781@209.227.182.44;tag=DC7456A0-3FE Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 1 BYE User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A) Content-Length: 0 U 209.227.182.43:5060 -> 216.236.137.226:1102 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.200.59:5060;rport=1102;received=216.236.137.226 From: <sip:12124051759@209.227.182.43>;tag=3720024524 To: <sip:2126882781@209.227.182.44>;tag=DC7456A0-3FE Date: Fri, 30 Apr 2004 15:30:21 GMT Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 1 BYE U 216.236.137.226:1102 -> 209.227.182.43:5060 BYE sip:12124051759@209.227.182.43 SIP/2.0 Route: <sip:2126882781@209.227.182.44> Via: SIP/2.0/UDP 172.30.200.59:5060 From: sip:12124051759@209.227.182.43;tag=3720024524 To: sip:2126882781@209.227.182.44;tag=DC7456A0-3FE Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 1 BYE User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A) Content-Length: 0 U 209.227.182.43:5060 -> 209.227.182.44:5060 BYE sip:2126882781@209.227.182.44 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 209.227.182.43;branch=z9hG4bKb7a6.88fba574.0 Via: SIP/2.0/UDP 172.30.200.59:5060;rport=1102;received=216.236.137.226 From: sip:12124051759@209.227.182.43;tag=3720024524 To: sip:2126882781@209.227.182.44;tag=DC7456A0-3FE Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 1 BYE User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A) Content-Length: 0 U 209.227.182.44:5060 -> 209.227.182.43:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 209.227.182.43;branch=z9hG4bKb7a6.88fba574.0,SIP/2.0/UDP 172.30.200.59:5060;rport=1102;received=216.236.137.2 26 From: sip:12124051759@209.227.182.43;tag=3720024524 To: sip:2126882781@209.227.182.44;tag=DC7456A0-3FE Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 1 BYE Content-Length: 0 U 209.227.182.43:5060 -> 216.236.137.226:1102 SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 172.30.200.59:5060;rport=1102;received=216.236.137.226 From: sip:12124051759@209.227.182.43;tag=3720024524 To: sip:2126882781@209.227.182.44;tag=DC7456A0-3FE Call-ID: 1B72FBC0-99F211D8-AFEAEAA0-F4703A0(a)209.227.182.44 CSeq: 1 BYE Content-Length: 0

2 1

[Serusers] source address of outbound tcp connections
by Alexander Mayrhofer 30 Apr '04

30 Apr '04

Hi, i'm trying to use t_relay_to_tcp() on a machine with several IP addresses on just one ethernet interface. SER seems not to set the source address of the TCP connection, so the OS (more or less) randomly chooses one of the virtual addresses on the interface, which makes in turn filtering based on source ip address a B<- (pain in the a** ;). Is there any way to configure SER 0.8.12 to use a specific source address in outbound tcp connections (it works for udp, though)? thanks axelm

2 1

[Serusers] setuid and setgid
by Zeus Ng 30 Apr '04

30 Apr '04

In the main.c file, there is a comment saying that 'we need to open fifo before do_suid() and start the fifo server after all the socket are initialized, to inherit them'. I'm wondering why SER has to do it like that. Can't SER lower its privileges first (do_suid), then open fifo and start the fifo server? The reason being that I want to run SER with an unprivileged user. With the current implementation, the fifo file will be created by root which is not accessible to anyone else. Unless I change the fifo_mode to 0666, which I don't want to, I can't make SER more secure and yet accessible. I've tried to move the do_suid() before init_fifo_server() and compile successfully. The server can run as ser:apache with fifo_mode set to 0660. That way, my Apache server can talk to SER yet limiting other user directly talking to SER via serctl. However, I would like to know the reason why it is implemented the way it is now. Maybe I'm not insightful to see the obviously reason. Also, will unix_sock get something like unix_sock_mode in the future? Zeus Ng ********************************************************************** 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, you are prohibited from reading, copying, distributing and using the information. Please contact the sender immediately by return email and destroy the original message. ******************************************************************

2 1

Re: [Serusers] rewrite & ACK forwarding problem
by lgfausak＠august.net 30 Apr '04

30 Apr '04

I have also had problems with getting the ACK back. I don't completely understand your configuration, you must allow for packets going both directions, right? Here is my config : route { # check to see if the message has been around too long # probably means that it is looping # if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; # # make sure the length of the message isn't too long! # if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); break; }; # # do the loose-routing thing, this is important! # if(loose_route()) { log(1,"doing top loose route"); t_relay(); break; }; # this is where I was dropping the ACKS. # I was simply dropping these, but they must be relayed # because they can be ACKs if(!(uri==myself)) { if(!t_relay()) { sl_reply_error(); break; }; break; }; This gets the ACKs through for me. By the way, I have this configured with Cisco ATAs, version 2.16. ---greg > >I have the same problem and posed it to the group yesterday ([Serusers] >Ignored 200 OK message.) So far the only workaround that I have found is to >use the rules in my gateway to rewrite the dialed digits before sending them >to the PSTN PRI, thus leaving the origianl URI intact for SIP >communications. > >One person told me that this is a bug in the Cisco ATA, but it happens on my >IPDialog phones also. It seems to me that the INVITE is being processed by >the SER dial rules and is rewritten, but the ACK is not. > >Sean >_______________________________________________ > >Sean Robertson > >NETXUSA >p. 800-289-6389 >f. 864-233-4344 "Ask me about Voice over IP." >http://www.netxusa.com/ > >----- Original Message ----- >From: "Alexander Mayrhofer" <axelm(a)nic.at> >To: <serusers(a)lists.iptel.org> >Sent: Friday, June 27, 2003 12:15 PM >Subject: [Serusers] rewrite & ACK forwarding problem > > >> >> Hi, >> >> we're running SER together with a PSTN Gateway. Before a call get's >> forwarded to the gateway, we are rewriting the request URI to make >> rewriting on the GW as simple as possible: >> >> route { >> ... >> strip(3); # +43xxx -> xxx >> prefix("0"); # xxx -> 0xxx >> rewritehostport(xxx.xxx.xxx.xxx, 5060); # request to gateway >> route(1); >> break; >> ... >> >> SIP call flow looks like (record route enabled): >> >> (1) phone -> SER >> INVITE sip:*43699xxxxxxxx@nic.at43.at SIP/2.0 >> >> (2) SER -> phone >> SIP/2.0 100 trying -- your call is important to us >> >> (3) SER -> GW >> INVITE sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> (4) GW -> SER >> SIP/2.0 100 Trying >> >> (5) GW -> SER >> SIP/2.0 183 Session Progress >> >> (6) SER -> phone >> SIP/2.0 183 Session Progress >> >> (7) GW -> SER >> SIP/2.0 180 Ringing >> >> (8) SER -> phone >> SIP/2.0 180 Ringing >> >> (9) GW -> SER >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxxx@xx.xx.xx.xx:5060> >> >> (10) SER -> phone >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxx@xx.xx.xx.xx:5060> >> >> [ call established, we can talk, but ... ] >> >> (11) phone -> SER >> ACK sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> --> Here starts the problem. That ACK (11) never gets forwarded to the >> Gateway, so after a few seconds, the GW starts over at (9). Those three >> packets (9-11) repeat a few times until GW runs into a timeout and drops >> the call. >> >> I have the impression that SER can't match the packet to the previous >> requests because of the rewritten URI. Is that correct? >> >> The only output at debug level 3 is: >> >> Warning: sl_send_reply: I won't send a reply for ACK!! >> >> Is that a routing goof somewhere in our scripts or is that a more >> generic problem? Is the problem that the warning indicates somehow >> related to the fact that the ACK is not being forwarded? >> >> Help appreciated. >> >> cheers >> >> axelm >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

4 5

[Serusers] RADIUS between multiple domains + Fail Over Gateways
by Alan Litster 30 Apr '04

30 Apr '04

Hello List, I've been using SER with RADIUS successfully now for a few months and am very pleased with the result. It's used for authenticating users accessing our gateways. I know have a new requirement to extend this to provide authentication for remote domains. The setup being as follows. We've got SER running with FreeRADIUS, then at the remote sites we will have the same plus Asterisk that is to act as a local gateway. I've configured the local FreeRADIUS instance to proxy the requests for the remote SIP domains to the remote RADIUS server. Unfortunately this doesn't work and I'm not sure why. The SUA gets asked by the remote SIP proxy to authenticate, it then forwards the INVITE to the local SER instance which then gets the LOCAL RADIUS to do another auth. This doesn't work. However if I disable the local auth and leave the remote auth enabled it works fine. Has anyone successfully managed to get proxied radius auth to work? My other question is to do with getting SER to send the INVITE to a different gateway if the primary one is at capacity/out of action? Is there an example of this sort of config? Kind Regards, Alan ------------------------------------------------------------------------------------------------------- This email, and any files transmitted with it, is copyright and may contain confidential information. The contents are intended for the use of the addressee(s) only. Unauthorized use may be unlawful. If you receive this email by mistake, please advise sender immediately. The views of the author may not necessarily constitute the views of Telco Electronics Limited. Nothing in this mail shall bind Telco Electronics Limited in any contract or obligation. Telco Electronics Limited 6-8 Oxford Court Brackley Northants NN13 7XY Tel 07000 701999 Fax 07000 701777

4 7

[Serusers] (no subject)
by Matt Hess 30 Apr '04

30 Apr '04

Is anyone working on a module to extend ser into a larger softswitch role like something for handling sigtran?

2 1

[Serusers] ZyXEL ACK Problem.
by markus monka 30 Apr '04

30 Apr '04

It is a ZyXEL Phone connected to a ZyXEL Wlan Router. Sending INVITE INVITE => SER => ASTERISK ASTERISK send "Session Progress" everything is fine: ASTERISK sends "OK" Session starts. I can hear my Voice. After 6 Seconds Session breaks up. Something seems to be strange with the ACK the ZyXEL Phones sends: PHONE => PROXY ACK sip:0xxxxx@21.1.7.9 SIP/2.0 Via:SIP/2.0/UDP 21.28.17.18:5060;branch=z9hGddbkb8a7ca191bf4fd From:Zyxel<sip:1111111@domain.de;user=phone>;tag=C2E2D6CF845EA11AEBC2 To:<sip:0xxxxxxxxx@domain.de;user=phone>;tag=as4f89971f Call-ID:15482-E655-54C3-B488-1D32D8DF7653@21.28.17.18 CSeq:1 ACK User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone Route:<sip:49xxxxx@21.1.7.8;ftag=C2E2D6CF845EA11AEBC2;lr=on>,<sip:492405690929@21.28.17.18:5060> When loggin with debug = 10 , i do not see some errors in syslog. The Gateway then sends 6 more "OK" and the terminate by sending "BYE". The Phone answers with "200 OK" .. ? Regards, Markus

2 1

[Serusers] SER+SERWEB again!!!!
by gaillac harry 30 Apr '04

30 Apr '04

Hi all, I solved my problem SEND IM, missed calls, Voicemail. I do hope that could help somebody !!! However when I call an offline user1 he received a missed calls from user2 but user1 don't receive email notification in his mailbox although voicemail system answer the user2's call??? Harry --------------------------------------------------------- # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ acc_log_request("404 missed call\n"); acc_db_request("404 missed call", "missed_calls"); setflag(3); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; if(method=="BYE" || method=="CANCEL"){ if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; }; break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); 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(); }; } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO: the downstream UA does not support MESSAGE requests ...\n"); # we have changed the R-URI with the contact address -- ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted for delivery"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; }

2 1

[Serusers] IP to Domain mapping
by Kapil Dhawan 30 Apr '04

30 Apr '04

I have added a user with some domain as ip address and second with domain as domain name. They are not able to contact each other..... and how can i enable multiple domain..... regards _________________________________________________________________ Need quick cash? http://go.msnserver.com/IN/46923.asp Click here !

2 1

[Serusers] What is the recommended Mysql version for ser 0.8.12
by steve 30 Apr '04

30 Apr '04

I have version mysql-3.23.58-1.9 installed but seem to remember a thread about a later version being more compatible with some of the functions in SER. Thanks in advance.. Regards, Steve ________________________________________________________________________ This message has been checked for all known viruses by the CitC Virus Scanning Service powered by SkyLabs. For further information visit http://www.citc.it ___

2 1

Re: [Serusers] multiple domains and pstn
by gaillac harry 30 Apr '04

30 Apr '04

Hi, do you try pdt module for incoming call, does this module could work with sems isdngw plugin? Anybody could send us back an example ? Regards Le ven 30/04/2004 à 09:56, Richard a écrit : > Gaillac, > > I use the latest cvs. > > It has a rpid field. > > Richard > > --- gaillac harry <gaillacharry(a)yahoo.fr> wrote: > > Sorry not serweb but ser_mysql.sh. rpid field is not > > created into > > subscriber table from ser-0.8.12 stable with > > ser_mysql.sh > > > > Gaillac > > > > Le jeu 29/04/2004 19:30, Richard a crit : > > > I haven't tried serweb yet. So now only manually > > > update mysql database. > > > > > > --- gaillac harry <gaillacharry(a)yahoo.fr> wrote: > > > > Hi > > > > > > > > rpid field is not include in subscriber table > > from > > > > serweb-2004 tarball > > > > but CVS serweb . > > > > Which ser_mysql.sh serweb release have you > > installed > > > > ? > > > > > > > > I'm very interesting in your problem > > > > > > > > Gaillac harry > > > > > > > > Le jeu 29/04/2004 10:31, Richard a crit : > > > > > The rpid doesn't seem to be in the document. > > > > Anyway, I > > > > > dig into the source file. Here is what I > > found, > > > > rpid > > > > > is stored in ser database in mysql server. It > > is > > > > in > > > > > the subsriber table and rpid column. > > > > > > > > > > You need modparam("auth_db", "use_rpid", 1), > > > > because > > > > > the default value is 0 and not using rpid. > > > > > > > > > > Optional modparam("auth_db","rpid_column", > > > > "rpid"), > > > > > which is default. > > > > > > > > > > There are other params documented in the > > README > > > > file > > > > > in modules/auth directory. > > > > > > > > > > rpid is retrieved during proxy authentication. > > > > > > > > > > Function append_rpid_hf() adds the rpid value > > into > > > > the > > > > > header field "Remote-Party-ID". > > > > > > > > > > rpid doesn't have to be number only. It can be > > any > > > > > format. > > > > > > > > > > Here is an example, rpid columns has > > > > > "test" 8005551212 > > > > > > > > > > The header has a header > > > > > Remote-Party-ID: "test" > > > > > > > > > > > > > > > 8005551212;party=calling;id-type=subscriber;screen=yes > > > > > > > > > > It should give the caller name as test and id > > as > > > > > 8005551212, hopefully. ;) I should have a rpid > > > > capable > > > > > PSTN gateway next week and let you posted. > > > > > > > > > > Thanks, > > > > > Richard > > > > > > > > > > > > > > > > > > > > --- Richard <mypop3mail(a)yahoo.com> wrote: > > > > > > Think I got the answer for question 1. Use > > rpid > > > > > > stored > > > > > > in each subscriber in the database before > > > > forwarding > > > > > > the call to PSTN gateway. > > > > > > > > > > > > --- Richard <mypop3mail(a)yahoo.com> wrote: > > > > > > > Hi, > > > > > > > > > > > > > > I have a few questions about multiple > > domains > > > > and > > > > > > > call > > > > > > > from/to pstn. I don't want to change the > > > > ser.cfg > > > > > > > script everytime a new domain is added. > > How > > > > can I > > > > > > > support the following features? > > > > > > > > > > > > > > 1. Every domain has their own 3 or 4 digit > > > > dial > > > > > > > plan. > > > > > > > When they call PSTN, their caller id > > should be > > > > the > > > > > > > either a prefix plus their internal number > > or > > > > a > > > > > > > generic receptionist number, but not the > > 3/4 > > > > digit > > > > > > > internal number. Different domains have > > > > different > > > > > > > prefix. Is there a way to do this without > > > > changing > > > > > > > the > > > > > > > script for every domain? > > > > > > > > > > > > > > 2. Should interdomain calls be prohibited? > > Is > > > > it > > > > > > > preferred to send the call to PSTN and > > hairpin > > > > > > back? > > > > > > > I > > > > > > > know it consumes more resources but would > > it > > > > > > provide > > > > > > > better privacy? If that's the case, how to > > > > prevent > > > > > > > interdomain calls, i.e. how to verify from > > and > > > > to > > > > > > > URI > > > > > > > have the same host parts? > > > > > > > > > > > > > > 3. When PSTN calls come in, should I use > > > > > > > prefix2domain > > > > > > > in pdt module to convert the PSTN number > > to a > > > > > > local > > > > > > > domain number? Any pitfalls? > > > > > > > > > > > > > > This list has been great! Thanks for all > > your > > > > > > help. > > > > > > > > > > > > > > Best Regards, > > > > > > > Richard > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > __________________________________ > > > > > > > Do you Yahoo!? > > > > > > > Win a $20,000 Career Makeover at Yahoo! > > > > HotJobs > > > > > > > > > > > > > > > > > > > http://hotjobs.sweepstakes.yahoo.com/careermakeover > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Serusers mailing list > > > > > > > serusers(a)lists.iptel.org > > > > > > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > __________________________________ > > > > > > Do you Yahoo!? > > > > > > Win a $20,000 Career Makeover at Yahoo! > > HotJobs > > > > > > > > > > > > > > > > http://hotjobs.sweepstakes.yahoo.com/careermakeover > > > > > > > > > > > > > > _______________________________________________ > > > > > > Serusers mailing list > > > > > > serusers(a)lists.iptel.org > > > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > __________________________________ > > > > > Do you Yahoo!? > > > === message truncated === > > > > > > __________________________________ > Do you Yahoo!? > Win a $20,000 Career Makeover at Yahoo! HotJobs > http://hotjobs.sweepstakes.yahoo.com/careermakeover

1 0

[Serusers] Call Timeout ?
by Kannaiyan Natesan 29 Apr '04

29 Apr '04

Can I disconnect a call which is in progress through SER after a pre determined time? Kannaiyan

2 1

[Serusers] multiple domains and pstn
by Richard 29 Apr '04

29 Apr '04

Hi, I have a few questions about multiple domains and call from/to pstn. I don't want to change the ser.cfg script everytime a new domain is added. How can I support the following features? 1. Every domain has their own 3 or 4 digit dial plan. When they call PSTN, their caller id should be the either a prefix plus their internal number or a generic receptionist number, but not the 3/4 digit internal number. Different domains have different prefix. Is there a way to do this without changing the script for every domain? 2. Should interdomain calls be prohibited? Is it preferred to send the call to PSTN and hairpin back? I know it consumes more resources but would it provide better privacy? If that's the case, how to prevent interdomain calls, i.e. how to verify from and to URI have the same host parts? 3. When PSTN calls come in, should I use prefix2domain in pdt module to convert the PSTN number to a local domain number? Any pitfalls? This list has been great! Thanks for all your help. Best Regards, Richard __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

3 5

Re: [Serusers] acc
by Andrey Pavlenko 29 Apr '04

29 Apr '04

Здравствуйте, Jan! On Tue, Apr 27, 2004 at 05:31:26PM +0200, Jan Janak wrote: > Could you try to compile the attached version of ser ? Sorry. The reason apeared in FreeBSD ports patches... :( > > Jan. > > On 27-04 18:33, Andrey Pavlenko wrote: > > > > ????????????, Jan! > > > > On Tue, Apr 27, 2004 at 05:09:12PM +0200, Jan Janak wrote: > > > > > > Do you use -DUSE_TCP when compiling ? I am pretty sure the latest > > > stable works, I've just tried. USE_TCP is normally defined in > > > Makefile.defs > > > > Of course. By default defined... > > > > DEFS+= $(extra_defs) \ > > -DNAME='"$(NAME)"' -DVERSION='"$(RELEASE)"' -DARCH='"$(ARCH)"' \ > > -DOS='"$(OS)"' -DCOMPILER='"$(CC_VER)"' -D__CPU_$(ARCH) -D__OS_$(OS) \ > > -DCFG_DIR='"$(cfg-target)"'\ > > -DPKG_MALLOC \ > > -DSHM_MEM -DSHM_MMAP \ > > -DDNS_IP_HACK \ > > -DUSE_IPV6 \ > > -DUSE_TCP \ > > -DDISABLE_NAGLE \ > > -DF_MALLOC \ > > > > > > > > > > Jan. > > > > > > On 27-04 17:24, Andrey Pavlenko wrote: > > > > > > > > ????????????, Jan! > > > > > > > > On Tue, Apr 27, 2004 at 02:14:14PM +0200, Jan Janak wrote: > > > > > > > > > > There was some problem with ser from FreeBSD ports, try to get the > > > > > source from the CVS (with -r rel_0_8_12 option). > > > > > > > > With cvs revision the same problem... :( > > > > > > > > > > > > > > > > > > > > > > Jan. > > > > > > > > > > On 27-04 13:32, Andrey Pavlenko wrote: > > > > > > > > > > > > Hi! > > > > > > > > > > > > I need enable radius accounting in the SER. > > > > > > But have problem after load module acc.so: > > > > > > > > > > > > 0(22487) DEBUG: init_mod: acc > > > > > > acc - initializing > > > > > > 0(22487) find_export: found <load_tm> in module tm [/usr/local/lib/ser/modules/tm.so] > > > > > > 0(22487) find_export: found <register_tmcb> in module tm [/usr/local/lib/ser/modules/tm.so] > > > > > > 0(22487) find_export: <t_relay_to_tcp> not found > > > > > > 0(22487) ERROR: tm_bind: TM module function 't_relay_to_tcp' not found > > > > > > 0(22487) init_mod(): Error while initializing module acc > > > > > > ERROR: error while initializing modules > > > > > > 0(22487) DEBUG: tm_shutdown : start > > > > > > 0(22487) DEBUG: tm_shutdown : empting DELETE list > > > > > > 0(22487) DEBUG: tm_shutdown : empting hash table > > > > > > 0(22487) DEBUG: tm_shutdown: releasing timers > > > > > > 0(22487) DEBUG: tm_shutdown : removing semaphores > > > > > > 0(22487) DEBUG: tm_shutdown : done > > > > > > 0(22487) shm_mem_destroy > > > > > > 0(22487) destroying the shared memory lock > > > > > > > > > > > > In the ser.cfg: > > > > > > 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_radius.so" > > > > > > loadmodule "/usr/local/lib/ser/modules/acc.so" > > > > > > > > > > > > # ----------------- setting module-specific parameters --------------- > > > > > > > > > > > > modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") > > > > > > modparam("auth_radius", "service_type", 15) > > > > > > modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") > > > > > > > > > > > > Im building ser-0.8.12 from the FreeBSD (4.9) ports and define > > > > > > DEFS+=-DRAD_ACC > > > > > > LIBS=-L$(LOCALBASE)/lib -lradiusclient > > > > > > in the acc module Makefile > > > > > > > > > > > > Please help me correct this problem. > > > > > > > > > > > > > > > > > > > > > > -- > > > > With best regards! > > > > =============================================================================== > > > > Lance http://www.immt.net.ua > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > -- > > With best regards! > > ---------------------- > > NOC ISP Merlin Telecom > > Phone: 8 044 249-6570 > > =============================================================================== > > Lance > > ANP-RIPE Mailto: lance(a)merlin.net.ua > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > Scanned by evaluation version of Dr.Web antivirus Daemon > http://drweb.ru/unix/ > -- With best regards! ---------------------- NOC ISP Merlin Telecom Phone: 8 044 249-6570 =============================================================================== Lance ANP-RIPE Mailto: lance(a)merlin.net.ua

1 0

[Serusers] Looping and sending of multiple requests
by Shirley 29 Apr '04

29 Apr '04

Hi, Please help! I am trying to do call forwarding when user not online. But I run the route below, the call went through for the PSTN gateway but I can see multiple request. Anyone knoe what is happening ...thanks.!! route{ # 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; }; }; lookup("aliases"); if (uri=~"sip:1234[0-9]+@iptel.org") { rewritehostport("1.2.3.4:5060"); # first dump the message to a file using cat command exec_msg("printenv SRCIP > /tmp/exectest.txt; cat >> /tmp/exectest.txt"); # and then rewrite URI using external utility # note that the last echo command trashes input parameter if (exec_dset("echo $SIP_RURI ;echo SIP_SRCIP ;echo>/dev/null")) { sl_send_reply("300", "Redirect"); } else { sl_reply_error(); log(1, "alas, rewriting failed\n"); }; }; if (!uri==myself) { t_relay(); break; }; if (method=="INVITE") { fix_nated_sdp("3"); append_hf("P-hint:Invite"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("", "Connecting"); break; }; }; if (method=="ACK"){ sl_send_reply("", "ACK"); }; # 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(); break; }; } # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { sdp_mangle_port("-12000"); if (isflagset(6) && status=~"2[0-9][0-9]"){ append_hf("First Route"); fix_nated_contact(); fix_nated_sdp("1"); force_rtp_proxy(); } else if (status=~"2[0-9][0-9]"){ if (uri=~"sip:019[0-9]+@iptel.org") { append_hf("019"); fix_nated_contact(); fix_nated_sdp("3"); } else{ append_hf("2nd Route"); fix_nated_contact(); fix_nated_sdp("2"); fix_nated_sdp("3"); force_rtp_proxy(); } } else if (status=~"3[0-9][0-9]"){ append_hf("019"); fix_nated_contact(); fix_nated_sdp("3"); }; }

1 0

[Serusers] vm_test.sh
by gaillac harry 29 Apr '04

29 Apr '04

hi all, here is my vm_test.sh: here is the reply from /var/log/messages: Apr 29 14:34:51 serveur1 /usr/sbin/ser[3657]: ERROR: fifo_uac: Content-Type missing I'd like to test vm module and sems ??? Harry ---------------------------------------------------- #!/bin/bash loop=5 max_speed=30 SIP_DST="sip:harry@192.168.0.1" alarm=0 SER_FIFO="/tmp/ser_fifo" SIP_FROM="sip:webmaster@192.168.0.1" cat > $SER_FIFO << EOF :t_uac_dlg:am_fifo INVITE $SIP_DST $SIP_DST <$SIP_DST> <$SIP_FROM> 12345-tag-12345 100 1234576098621 Content-Type: application/sdp Contact: <sip:webmaster@192.168.0.11:5060> v=0 o=username 0 0 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 30000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 . EOF --------------------------------------------------------------

1 0

[Serusers] Problem with serweb 2004
by Richa_Ziad＠yahoo.fr 29 Apr '04

29 Apr '04

Hello, serusers, I have configured correctly SERWEB like mentionned in admin guide but: I type correctly the UN & password and click login, but no result displayed (the progress indicator in the buttom of the page can't continue). NB: I can access the admin interface when i type correctly the Admin name and password. Can someone help me to resolve this problem. Best regards.

1 0

[Serusers] SER Monitoring
by Markos Vakondios 29 Apr '04

29 Apr '04

Hello, I would like to ask if the updating of the server_monitoring and server_monitoring_agg has been implemented in 0.8.12. Thanks

4 8

RE: [Serusers] Using parameters from the database
by Shirley 28 Apr '04

28 Apr '04

Hi Andy, Thank you for your email. I have having no more pending branches in the sip message when I do a PC to PSTN through aliases. PLease help. Regards, Shirley > -----Original Message----- > From: Andreas Granig [SMTP:a.granig@inode.at] > Sent: Tuesday, April 27, 2004 10:50 PM > To: Shirley > Subject: Re: [Serusers] Using parameters from the database > > Shirley wrote: > > Can I ask you another question, if I were to forward to PSTN gateway can > i > > still use this method?If yes is the coding the same? Thanks!! > > Yes of course. The host part of the aliased URI points to the new > location (e.a. a PSTN-GW or another SIP-Proxy), and the Userpart is the > Username/Number to forward to. You can just use t_relay() for forwarding > then. > > hth, > Andy

1 0

[Serusers] OPTIONS method
by Richard 28 Apr '04

28 Apr '04

Hi, I am testing ser with Asterisk which sends an OPTIONS method every minute. The uri is sip:<my-ip-address>. What's the proper way to handle this? Thanks, Richard __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

4 4

[Serusers] uri==myself and is_uri_host_local
by Richard 28 Apr '04

28 Apr '04

Hi, In a multiple domain setup, are two functions uri==myself and is_uri_host_local the same thing? is_uri_host_local is not presented in the README, but is exported. Is it a obsolete function? Thanks, Richard __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

2 5

[Serusers] Cisco BYE messages
by Helge Waastad 28 Apr '04

28 Apr '04

Hello, I wonder if it is a bug, non standard compliance or whatever - When the PSTN side initiate a call to the IP leg, and the PSTN side hang up first, the Cisco gateway issues a BYE message. - When the IP side initiates the call to PSTN, and the PSTN hangs up first, the Cisco does NOT issue a BYE message. I want the Cisco to issue a BYE when the PSTN side hangs up if is initiated on either leg. I'm running a 3640 with this image: c3640-js-mz.122-15.T.bin best regards, hw

1 0

[Serusers] About NAThelper in iptel.org and ser.cfg
by jimmy huang 28 Apr '04

28 Apr '04

Hi all I am trying nathelper with rtpproxy to let client behind nat/firewall can make a call , I have some questions... (1) Client A (kphone 3.14)------------------iptel.org-----------------------------NAT-------------------------Client B (kphone 3.14) 61.217.126.64 195.37.77.101 61.217.xxx.xxx 77.77.77.17 B call A , and the call has setup , also , both A and B send rtp packet to 195.37.77.101 , and receive in the port which describe in SDP but both client can't receive any voice ... maybe it's something wrong in client ... so is anyone have try if this scenario can work ? or iptel.org not support client behind NAT/ firewall ?? (2) before trying (1) , i have build the environment Client A (kphone 3.14)-------------SIP server SER + Rtpproxy ---------------------NAT/Firewall-------------------------Client B (kphone 3.14) private IP public IP public IP private IP whatever A call B , or B call A , the call can setup , but after forwarding by SER the SDP didn't modify correctly , all the same as client send it should modify to Ser server's ip and port , but it seems not... i think maybe is my ser.cfg has some mistake could anyone give some idea where should i add or modify something ? here is my ser.cfg which refered to (http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch04s07.ht…) regards jimmy ============================================================================================ # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) #/* Uncomment these lines to enter debugging mode debug=7 #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/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" #++++++++++ jimmy added ++++++++++++++++++ loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/nathelper.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" # ----------------- setting module-specific parameters --------------- #++++++++++ jimmy added ++++++++++++++++++ #we will you flag 6 to mark NATed contacts modparam("registrar","nat_flag",6) #Enable NAT pinging modparam("nathelper","natping_interval",3) #ping only contacts that are known to be behind NAT modparam("nathelper","ping_nated_only",1) #---------------------------------------------------- # -- 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") # -- 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{ #++++++++++ jimmy added ++++++++++++++++++ if(nat_uac_test("3")) { if((method == "REGISTER") || !(search("^Record-Route:"))) { log("LOG:Someone trying to register from private IP, rewriting\n"); fix_nated_contact(); if(method == "INVITE") { fix_nated_sdp("3"); }; force_rport(); setflag(6); }; }; #---------------------------------------------------- # 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") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); 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(); }; } #++++++++++ jimmy added +++++++++++++++++++++ route[1] { if(uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")) { sl_send_reply("479","we don't forward to private IP address"); break; }; if(isflagset(6)) { force_rtp_proxy(); t_on_reply("1"); append_hf("P-Behind-NAT: Yes\r\n"); } if(!t_relay()) { sl_reply_error(); break; } } onreply_route[1] { if(status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); }; } #---------------------------------------------------- ============================================================================================

1 0

Re: [Serusers] The problem when enable the MySql
by Klaus Darilion 28 Apr '04

28 Apr '04

Hi! I'm not sure about this, but I think the problem is that the hashed password also include an random generated string. What you can do is to authenticate only on the first server and trust the request automaticall if REGISTER request comes from the other server. regards, klaus John LI wrote: > Hi Klaus, > > Thanks for your response, I have another question: > I installed 2 Ser on the different machine, and add the 2 user 3001 3002 on > both Ser. I configuer 2 SIP client to Regiseter to the same ser, there is no > problem to make call to each other, but when I try to registered 2 sip > client on the different machine, the problem happend. > > For example, 2 ser server IP is A=192.168.0.10 and B=192.168.0.20 > > I changed ser.cfg in ser A as > > if (!src_ip==192.168.0.20) { > log("*****replicate***"); > t_replicate("192.168.0.20", "5060"); > > and on B > > if (!src_ip==192.168.0.10) { > log("*****replicate***"); > t_replicate("192.168.0.10", "5060"); > > And when I try to regiseter SIP client on A, it can register on A, then the > Ser A will send a Register request to B like this: > > U 192.168.0.10:5060 -> 192.168.0.20:5060 > REGISTER sip:ny1.signalc.com SIP/2.0..Record-Route: > <sip:192.168.0.10;ftag=35a > 3@ny1.signalc.com..From: > 3001<sip:3001@ny1.signalc.com;user=phone>;tag=35adc-1 > =phone>..CSeq: 102 REGISTER..Via: SIP/2.0/UDP > 192.168.0.10;branch=z9hG4bK247f. > 060..Contact: <sip:3001@192.168.0.30:5060;user=phone>..Max-Forwards: > 69..User- > ..Expires: 900..Supported: timer..Authorization: Digest > username="3001",realm= > > ,response="4890af0391f9985e7d28a4f98e9132ab",nonce="4064805a3152161d3bb4ba9d > 50 > > U 192.168.0.20:5060 -> 192.168.0.10:5060 > SIP/2.0 401 Unauthorized..Call-ID: > 55af-6a37c-4000fbb3@ny1.signalc.com..From: > ag=35adc-1408..To: > 3001<sip:3001@ny1.signalc.com;user=phone>;tag=b27e1a1d33761 > R..Via: SIP/2.0/UDP 192.168.0.10;branch=z9hG4bK247f.7ee9ba07.0..Via: > SIP/2.0/U > est realm="ny1.signalc.com", > nonce="406490a267fa70c71b2e27a481dfbde8b5fd6bd4". > inux))..Content-Length: 0..Warning: 392 192.168.0.20:5060 "Noisy feedback > tell > c_port=5060 in_uri=sip:ny1.signalc.com out_uri=sip:ny1.signalc.com > via_cnt==2" > > > Ser A will try to register 3001 to Ser B, but Ser B dose not authenticate > the 3001 although the 3001 has the same user ID and Password on the Ser B. > > I wonder if the modification on the ser.cfg is correct, should I do more > things for this? > > Thanks > > John > > > ----- Original Message ----- > From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> > To: "John LI" <john(a)signalc.com> > Cc: "Serusers" <serusers(a)lists.iptel.org> > Sent: Friday, March 26, 2004 4:36 AM > Subject: Re: [Serusers] The problem when enable the MySql > > > >>Hi John! >> >>comments inline >> >>PS: Please always cc: to the list. >> >>John LI wrote: >> >> >>>Hi Klaus, >>> >>>Thanks for your instaruction, after I do all those steps, the Ser with > > the > >>>Digest creditial works fine now. >>> >>>My Questions is, after I complile and installed the ser, is the Ser > > database > >>>installed? >> >>No, the ser database is not installed by "make install". You have to use >>the ser_mysql.sh script to create the database. >> >> >>>Need I to use the serctl create to create the database ser >>>database again(from my practice, the Ser DB has been setup after all > > those > >>>steps), need I to install the ser-mysql-0.8.12-0 RPM package again? >> >>Once you have created the databse, you inly have to re-create it, if you >>are using a new version of ser which uses a new format. >> >>If you compile ser from source, you don't need any of the rpm packages. >> >>regards, >>klaus >> >> >>>Thanks for you advice again. >>> >>>John >>> >>> >>> >>>----- Original Message ----- >>>From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> >>>To: "John LI" <john(a)signalc.com> >>>Cc: <serusers(a)lists.iptel.org> >>>Sent: Thursday, March 25, 2004 3:07 AM >>>Subject: Re: [Serusers] The problem when enable the MySql >>> >>> >>> >>> >>>>compile ser by calling make: >>>># make >>>> >>>>compile ser modules by calling: >>>># make modules >>>> >>>>compile mysql ser modules by calling: >>>># make modules modules=modules/mysql >>>> >>>>install ser: >>>># make install >>>> >>>>install mysql module: >>>># cp modules/mysql/mysql.so /usr/local/lib/ser/modules/ >>>> >>>> >>>>regards, >>>>klaus >>>> >>>>John LI wrote: >>>> >>>> >>>>>Dear Sir, >>>>> >>>>>I am installing the Ser on Redhat 9.0 platform. >>>>> >>>>>I download ser-0.8.12_src.tar.gz, and compile the souce code, and >>>>>install it. >>>>>The Ser working fine with out the MySQL enabled. >>>>> >>>>>I download the MySQL packges from www.mysql.com <http://www.mysql.com> >>>>> >>>>>MySQL-client-4.0.18-0.i386.rpm >>>>>MySQL-server-4.0.18-0.i386.rpm >>>>>MySQL-shared-4.0.18-0.i386.rpm >>>>> >>>>>and install them, and Mysql can start up. >>>>> >>>>>After this, I download the ser-0.8.12_src.tar.gz >>>>> >>>>>and try to install it, and it give me out the error messge >>>>>error: Failed dependencies: >>>>> ser = 0.8.12 is needed by ser-mysql-0.8.12-0 >>>>> >>>>>I don't know what this means. >>>>> >>>>>BTW, when i enable the MySQL in ser.cfg, it give out the following > > error > >>>>>messages: >>>>>0(25007) ERROR: load_module: could not open module >>>>></usr/local/lib/ser/modules/mysql.so>: >>>>>/usr/local/lib/ser/modules/mysql.so: cannot open shared object file: No >>>>>such file or directory >>>>>I wonder how can I get the mysql.so module. >>>>> >>>>>I am the fresh man in using Ser, please give me some instructions. >>>>> >>>>>John >>>>> >>>>> >> >>>>------------------------------------------------------------------------ >>>> >>>>>_______________________________________________ >>>>>Serusers mailing list >>>>>serusers(a)lists.iptel.org >>>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>> >> > >

9 22

[Serusers] SERWEB+SER help!!!!!
by gaillac harry 27 Apr '04

27 Apr '04

Here is my ser.cg file. I try to configure send IM, Missed calls, Voicemail. I just can configure Send IM. Now I can call isdngw, ivr,.. I need help to configure lookup location logic for missed calls and voicemail (offline users) Anybody could help me to configure the others functions (problem with logic !!). harry Regards # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.0.1 port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/vm.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/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_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # -- 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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # --registrar params-- modparam("registrar", "use_domain", 1) # --domain params-- modparam("domain", "db_url", "mysql://ser:heslo@localhost/ser") modparam("domain", "db_mode", 1) # Use chaching # --acc params-- modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 3) # --msilo params-- modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@192.168.0.1") # --uri params-- #modparam("uri", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("uri", "subscriber_table", "subscriber") # --vm params-- modparam("voicemail", "db_url", "mysql://ser:heslo@localhost/ser") # ------------------------- 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; }; # --isdn gateway-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:[0-9]{10}@.*") { route(2); break; }; }; # --ivr conf-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:5000@.*") { route(4); break; }; }; # --conference-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:6000@.*") { route(5); break; }; }; # --play an annoucement-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:7000@.*") { route(6); 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 (is_from_local()) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); 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(); }; } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO: the downstream UA does not support MESSAGE requests ...\n"); # we have changed the R-URI with the contact address -- ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted for delivery"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } failure_route[4] { route(3); #append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); } route[2] { # ############################## # # isdngw specific configuration # # ############################## # if(t_newtran()){ if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ # send a response right at the start to avoid retransmissions t_reply("100","Trying -- just wait a minute !"); # isdngw only gets activated on invite requests if(method=="INVITE"){ # filename is defined in sems.conf. if(uri=~"sip:[0-9]{10}@.*"){ if(!vm("/tmp/am_fifo","isdngw")){ log("could not contact isdngw\n"); t_reply("500","could not contact isdngw"); }; # Allow the announcement module of sems to be used as well. # This can be useful for testing the isdngw. } else if(uri=~"sip:7000@.*"){ if(!vm("/tmp/am_fifo","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact announcement"); }; # we dont feel responsible for sip addresses of any other kind, # so send the right error code. } else { t_reply("404","Not Found"); }; # stop routing here, the message is now processed by the media server break; }; # The following handles the call termination, we must pass these requests # to the media server as follows. Again make shure the fifo name and permissions # are set correctly (like im sems.conf). if((method=="BYE")||(method=="CANCEL")){ if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; break; }; # other methods than INVITE, BYE and CANCEL are not handled by this SIP Server # so we sent an error message } else { log("ERROR: method not supported\n"); t_reply("500", "sorry, method not supported"); }; } else { # for any reason the transaction could not be created, send error code log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; # in isdngw.conf. Don't change this setting. t_relay(); # end of routing. } route[3] { ############################################ # Voicemail specific configuration - begin # ############################################ if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ if(!vm("/tmp/am_fifo","voicemail")) { log("couldn't contact announcement server\n"); t_reply("500", "couldn not contact announcement server"); }; break; }; if(method=="BYE" || method=="CANCEL") { if(!vm("/tmp/am_fifo","bye")) { log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; break; }; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } route[4] { ###################################### # ivr specific configuration - begin # ###################################### if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if (uri=~"sip:5000@.*") { if (!vm("/tmp/am_fifo", "ivr")) { log("couldn't contact ivr server\n"); t_reply("500", "couldn not contact ivr server"); }; }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact ivr\n"); t_reply("500","could not contact ivr"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } route[5] { # #################################### # conference specific configuration # # #################################### if(t_newtran()){ if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ # send a response right at the start to avoid retransmissions t_reply("100","Trying -- just wait a minute !"); # isdngw only gets activated on invite requests if(method=="INVITE"){ # filename is defined in sems.conf. if(uri=~"sip:6000@.*"){ if(!vm("/tmp/am_fifo","conference")){ log("could not contact conference\n"); t_reply("500","could not contact conference"); }; # Allow the announcement module of sems to be used as well. # This can be useful for testing the conference. } else if(uri=~"sip:5000@.*"){ if(!vm("/tmp/am_fifo","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact announcement"); }; # we dont feel responsible for sip addresses of any other kind, # so send the right error code. } else { t_reply("404","Not Found"); }; # stop routing here, the message is now processed by the media server break; }; # The following handles the call termination, we must pass these requests # to the media server as follows. Again make shure the fifo name and permissions # are set correctly (like im sems.conf). if((method=="BYE")||(method=="CANCEL")){ if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; break; }; # other methods than INVITE, BYE and CANCEL are not handled by this SIP Server # so we sent an error message } else { log("ERROR: method not supported\n"); t_reply("500", "sorry, method not supported"); }; } else { # for any reason the transaction could not be created, send error code log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; # in isdngw.conf. Don't change this setting. t_relay(); # end of routing. } route[6] { ###################################### # announcement configuration - begin # ###################################### if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if (uri=~"sip:7000@.*") { if (!vm("/tmp/am_fifo", "announcement")) { log("couldn't contact ivr server\n"); t_reply("500", "couldn not contact announcement"); }; }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact annoucement\n"); t_reply("500","could not contact annoucement"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } route[7] { # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); break; }; if (t_newtran()) { t_reply("404", "Not Found"); acc_db_request("404 missed call", "missed_calls"); }; }

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[Serusers] unauthorized error
by Ronald Ramos 27 Apr '04

27 Apr '04

Hi, What does this Unauthorized error mean? U 203.87.131.194:5060 -> 203.131.137.13:5061 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 203.131.137.13:5061;branch=z9hG4bKa7a4e17569a95335..From: "Ron Ramos" <sip:20 01(a)203.87.131.194>;tag=f73d945fb0a46be2..To: <sip:914082495780@203.87.131.194>;tag=3027420bad6eab8ed312f4f108c3114a.753b ..Call-ID: bbae276274aeab1b@192.168.1.101..CSeq: 4883 INVITE..WWW- Authenticate: Digest realm="silverbackasp.com", nonce= "408f56b1d3dc00a410bcfcd1842fe4c88c961176"..Server: Sip EXpress router (0.8.12 (i386/linux))..Content-Length: 0..Warning : 392 203.87.131.194:5060 "Noisy feedback tells: pid=5238 req_src_ip=203.131.137.13 req_src_port=5061 in_uri=sip:914082 495780(a)203.87.131.194 out_uri=sip:914082495780@203.87.131.194 via_cnt==1".... Thank you Ronald

2 3

[Serusers] how to forward ACK to another SIP server
by Ronald Ramos 27 Apr '04

27 Apr '04

Hi All, I have successfully called to PSTN unfortunately I got cut off in about a minute, the gateway provider told us that he can't see an ACK from my SIP, but on my SIP, I received an ACK from the SIP phone. How can I forward that ACK to the gateway providers SIP? Thank You Ronald

2 1

[Serusers] acc
by Andrey Pavlenko 27 Apr '04

27 Apr '04

Hi! I need enable radius accounting in the SER. But have problem after load module acc.so: 0(22487) DEBUG: init_mod: acc acc - initializing 0(22487) find_export: found <load_tm> in module tm [/usr/local/lib/ser/modules/tm.so] 0(22487) find_export: found <register_tmcb> in module tm [/usr/local/lib/ser/modules/tm.so] 0(22487) find_export: <t_relay_to_tcp> not found 0(22487) ERROR: tm_bind: TM module function 't_relay_to_tcp' not found 0(22487) init_mod(): Error while initializing module acc ERROR: error while initializing modules 0(22487) DEBUG: tm_shutdown : start 0(22487) DEBUG: tm_shutdown : empting DELETE list 0(22487) DEBUG: tm_shutdown : empting hash table 0(22487) DEBUG: tm_shutdown: releasing timers 0(22487) DEBUG: tm_shutdown : removing semaphores 0(22487) DEBUG: tm_shutdown : done 0(22487) shm_mem_destroy 0(22487) destroying the shared memory lock In the ser.cfg: 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_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") Im building ser-0.8.12 from the FreeBSD (4.9) ports and define DEFS+=-DRAD_ACC LIBS=-L$(LOCALBASE)/lib -lradiusclient in the acc module Makefile Please help me correct this problem. -- With best regards! =============================================================================== Lance ANP-RIPE http://www.immt.net.ua

2 4

[Serusers] cvs code compile error
by Lambros Lambrinos 27 Apr '04

27 Apr '04

Hi, Just unchecked the latest code from CVS but on compile i get the following error: [sip_router]# make all Extra objs: gcc -Wl,-O2 -Wl,-E action.o crc.o daemonize.o data_lump.o data_lump_rpl.o dprint.o dset.o error.o fifo_server.o flags.o forward.o hash_func.o ip_addr.o main.o md5.o md5utils.o modparam.o msg_translator.o pass_fd.o proxy.o qvalue.o re.o receive.o resolve.o route.o route_struct.o script_cb.o socket_info.o sr_module.o stats.o tcp_main.o tcp_read.o timer.o tsend.o udp_server.o unixsock_server.o usr_avp.o mem/f_malloc.o mem/mem.o mem/memtest.o mem/q_malloc.o mem/shm_mem.o mem/vq_malloc.o parser/hf.o parser/msg_parser.o parser/parse_content.o parser/parse_cseq.o parser/parse_disposition.o parser/parse_diversion.o parser/parse_event.o parser/parse_expires.o parser/parse_fline.o parser/parse_from.o parser/parse_hname2.o parser/parse_hostport.o parser/parse_nameaddr.o parser/parse_param.o parser/parser_f.o parser/parse_rpid.o parser/parse_rr.o parser/parse_to.o parser/parse_uri.o parser/parse_via.o parser/digest/digest.o parser/digest/digest_parser.o parser/digest/param_parser.o parser/contact/contact.o parser/contact/parse_contact.o db/db.o db/db_fifo.o lex.yy.o cfg.tab.o -lfl -ldl -lresolv -o ser dset.o: In function `print_dset': /home/ser/sip_router/dset.c:214: undefined reference to `print_q' /home/ser/sip_router/dset.c:237: undefined reference to `print_q' collect2: ld returned 1 exit status make: *** [ser] Error 1 Anyone has any idea where this function is meant to be declared ? Regards, Lambros

2 1

Re: [Serusers] seweb-2004
by Klaus Darilion 27 Apr '04

27 Apr '04

1. no direct mails -> mail to the list! 2. maybe you should make yourself familiar with php, mysql and SQL first! 3. gaillac harry wrote: > Here is my missed_calls.php > > Where is the SQL query file ??? > > $q="select username, domain from ".$config->table_aliases. > " where > 'sip:".$auth->auth["uname"]."@".$config->default_domain."'=contact"; > this is an SQL query oh, here is antohter one > > $q="delete from ".$config->table_missed_calls. > " where username='".$row."' and domain='".$dom."' ". > " and time<'".gmdate("Y-m-d H:i:s", $page_loaded_timestamp)."'"; > $res=mySQL_query($q); > if (!$res) {$errors[]="error in SQL query, line: ".__LINE__; } and again a query! > */ > $q="(SELECT t1.from_uri, t1.sip_from, t1.time, t1.sip_status ". > "FROM missed_calls t1 ". > "WHERE t1.username='".$auth->auth["uname"]."' and > t1.domain='".$config->default_domain."' ) ". > "UNION ". > "(SELECT t1.from_uri, t1.sip_from, t1.time, t1.sip_status ". > "FROM missed_calls t1, aliases t2 ". > "WHERE 'sip:".$auth->auth["uname"]."@".$config->default_domain."'". > "=t2.contact AND t2.username=t1.username AND t2.domain=t1.domain ) > ". > "ORDER BY time DESC"; That's probably not very useful for you (as it looks like you have no glue what you are talking about - sorry), but I will give you another hint from your previous mail: > >I've still a problem with missed_calls.php from serweb "error in SQL > > query, line: 77 No missed calls". --> check line 77 and you will find a query just before it! klaus

3 8

[Serusers] lookup("...")
by Atle Samuelsen 27 Apr '04

27 Apr '04

Hey Guys (And most likely No girls :( ) I'v got this scenario now : if a user does have voicemail everything is great!. No problems. Thou, my case is that I've got users who does not have a voicemail service, and may disconnect from my gw. Then when a imcoming call coms, the "caller" get's a 404 (per today). So the way I want it to be is : if (!lookup("location"){ (lookup("subscirber"){ log(1,"user's not online but does exist"); sl_send_reply("404", "Not found,but exists"); break; } log(1,"404 , user not found"); sl_send_reply("404","Not found"); break; } If I do it like this,the !lookup("subscriber") will fail, because it cant do this. How can I get this to work ? I see 2 difrent way's of doing it. 1: Create a database with the name users, and dump all my users from this into that database,, then have the same structure etc as I have in the locations database.. .. As far as I see , this will be a ugly hack. 2: make a new function for this .. How and what do you guys mean ? Thanks for a quick answer :) You guys are REALLY helpful both night and day. - Atle

4 3

[Serusers] 9 seconds
by Kapil Dhawan 27 Apr '04

27 Apr '04

Hi i am using default file for ser.cfg and grandstream 286....any call made from garndstream gets disconnected in 9 seconds...cisco works fine...i traced logs also....but didn't get any clue... _________________________________________________________________ Deals can't get any better. Products at Lowest Prices. http://go.msnserver.com/IN/47508.asp Only on baazee.com

3 2

[Serusers] Aliases
by Girish 27 Apr '04

27 Apr '04

i think this may be the problem 1.uncomment this one modparam("usrloc", "db_mode", 0) 2.give this syntax correctly modparam("usrloc", "db_url", "sql//ser:heslo@localhost/ser")

2 1

Re: [Serusers] SER 0.8.12 and RFC
by CURRO_DOMINGUEZ 27 Apr '04

27 Apr '04

Hello, Klaus, Jan, thank you very much for your answers. Below is the list of RFCs I'm interested in. As Klaus says, maybe some of them don't need explicit support, because ser can route these new messages. Or maybe there are RFCs that are not fully supported. Please, let me know if this is the case and if it could be necessary writting a new module or changing SER core. By the way, can SER initiate a session release (it means, generate a BYE) or generate a REGISTER to an application server? If not, what would be necessary from your point of view? Again, thank you very much for your time and knowledge. I know that there are a lot of questions in this message, so thank you. Curro (These five RFCs are related to new messages, so I suppose Yes) RFC 2976 (October 2000): "The SIP INFO method". RFC 3262 (June 2002): "Reliability of provisional responses in Session Initiation Protocol (SIP)". (can SER generate PRACK, besides proxying it?) RFC 3515 (April 2003): "The Session Initiation Protocol (SIP) REFER method". RFC 3311 (September 2002): "The Session Initiation Protocol (SIP) UPDATE method". RFC 3428 (December 2002): "Session Initiation Protocol (SIP) Extension for Instant Messaging". RFC 3313 (January 2003): "Private Session Initiation Protocol (SIP) Extensions for Media Authorization". RFC 3265 (June 2002): "Session Initiation Protocol (SIP) Specific Event Notification". (As far as I know, SER handles SUBSCRIBE and NOTIFY, but I want to know whether SER can act as the subscriber to the event information, it means, generate SUBSCRIBE) RFC 3327 (December 2002): "Session Initiation Protocol Extension Header Field for Registering Non-Adjacent Contacts". RFC 3325 (November 2002): "Private Extensions to the Session Initiation Protocol (SIP) for Network Asserted Identity within Trusted Networks". RFC 3323 (November 2002): "A Privacy Mechanism for the Session Initiation Protocol (SIP)". RFC 3608 (October 2003): "Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration". RFC 3486 (February 2003): "Compressing the Session Initiation Protocol (SIP)". RFC 3455 (January 2003): "Private Header (P-Header) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP)". RFC 3329 (January 2003): "Security Mechanism Agreement for the Session Initiation Protocol (SIP)". ----- Mensaje Original ----- De: Jan Janak <jan(a)iptel.org> Fecha: Lunes, Abril 12, 2004 10:28 pm Asunto: Re: [Serusers] SER 0.8.12 and RFC > There is no such list. If you need it then put together a list of RFCs > you are interested in and we will add yes/no to each entry in the > list. > Jan. > > On 12-04 12:38, CURRO_DOMINGUEZ wrote: > > Hello all the list > > > > I'm looking some information about SER at www.iptel.org but I've > found > > nothing. I would like to know the list of RFC that SER > implements > > (related to SIP or SIP extensions), besides RFC3261. > > > > I think it could be interesting to add this information at the > product > > sheet. > > > > Can you help me? > > > > Thank you very much > > > > Curro > > > > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

3 3

[Serusers] seweb-2004
by gaillac harry 27 Apr '04

27 Apr '04

Hello, I've still a problem with missed_calls.php from serweb "error in SQL query, line: 77 No missed calls". Harry

2 1

[Serusers] "nat_uac_test" - missing modules when starting SER?
by Marty Chen 27 Apr '04

27 Apr '04

hi, when ever i startup SER with this function used, "nat_uac_test", i get the cfg loading error "is some module missing" i know the function is in nathelper module, and it is loaded too. can someone tell me what to do in order to use this function? below is a sample code using nat_uac_test. thx in advance. marty. ============================================================== 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("ˆRecord-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 smart enough to be symmetric. In some phones, like # it takes a configuration option. With Cisco 7960, it is # called NAT_Enable=Yes, with kphone it is called # "symmetric media" and "symmetric signaling". (The latter # not part of public released yet.) 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 }; }; best regards, --------------------------------------------- Neo-Online Corporation Next Gen Networking Expert Marty Chen System Engineer Tel: +886-2-7707-7988 ext 151 Cell: +886-960-516-560 Email: marty-chen(a)neo-ol.com

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[Serusers] problems about the start and the stop of ser
by zhu qiang 27 Apr '04

27 Apr '04

I have installed ser in my computer 1, my computer 1 is in a LAN, and I have another computer 2 in the LAN, I have a msn messenger 4.6 installed in 2, and I configured msn 4.6 according to howto, I fill the IP address of 1 in server name or IP address in msn advanced configures. but when I want to register, it says that the network service is currently unavailable, but I can ping 1 form 2. when I start ser, I can stop it at the first time, When I restart it, if I stop in immediately, it is ok. but if after a few minuters, I want to stop it, it responses as the following /etc/rc.d/init.d/ser: line 195: kill: (4573) - no such process /etc/rc.d/init.d/ser: line 195: kill: (4572) - no such process .......................... Can anybody help me? _________________________________________________________ Do You Yahoo!? 惠普TT游戏剧，玩游戏，中大奖！ http://cn.rd.yahoo.com/mail_cn/tag/SIG=1402c0to2/**http%3A%2F%2Fhp.allyes.c…

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[Serusers] SERWEB+SER help!!!!! nobody could help me??
by gaillac harry 27 Apr '04

27 Apr '04

Here is my ser.cg file. I try to configure send IM, Missed calls, Voicemail. I just can configure Send IM. Now I can call isdngw, ivr,.. I need help to configure lookup location logic for missed calls and voicemail (offline users) Anybody could help me to configure the others functions (problem with logic !!). harry Regards # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.0.1 port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/vm.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/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_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # -- 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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # --registrar params-- modparam("registrar", "use_domain", 1) # --domain params-- modparam("domain", "db_url", "mysql://ser:heslo@localhost/ser") modparam("domain", "db_mode", 1) # Use chaching # --acc params-- modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 3) # --msilo params-- modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@192.168.0.1") # --uri params-- #modparam("uri", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("uri", "subscriber_table", "subscriber") # --vm params-- modparam("voicemail", "db_url", "mysql://ser:heslo@localhost/ser") # ------------------------- 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; }; # --isdn gateway-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:[0-9]{10}@.*") { route(2); break; }; }; # --ivr conf-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:5000@.*") { route(4); break; }; }; # --conference-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:6000@.*") { route(5); break; }; }; # --play an annoucement-- if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ if (uri=~"sip:7000@.*") { route(6); 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 (is_from_local()) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); 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(); }; } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO: the downstream UA does not support MESSAGE requests ...\n"); # we have changed the R-URI with the contact address -- ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted for delivery"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } failure_route[4] { route(3); #append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); } route[2] { # ############################## # # isdngw specific configuration # # ############################## # if(t_newtran()){ if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ # send a response right at the start to avoid retransmissions t_reply("100","Trying -- just wait a minute !"); # isdngw only gets activated on invite requests if(method=="INVITE"){ # filename is defined in sems.conf. if(uri=~"sip:[0-9]{10}@.*"){ if(!vm("/tmp/am_fifo","isdngw")){ log("could not contact isdngw\n"); t_reply("500","could not contact isdngw"); }; # Allow the announcement module of sems to be used as well. # This can be useful for testing the isdngw. } else if(uri=~"sip:7000@.*"){ if(!vm("/tmp/am_fifo","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact announcement"); }; # we dont feel responsible for sip addresses of any other kind, # so send the right error code. } else { t_reply("404","Not Found"); }; # stop routing here, the message is now processed by the media server break; }; # The following handles the call termination, we must pass these requests # to the media server as follows. Again make shure the fifo name and permissions # are set correctly (like im sems.conf). if((method=="BYE")||(method=="CANCEL")){ if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; break; }; # other methods than INVITE, BYE and CANCEL are not handled by this SIP Server # so we sent an error message } else { log("ERROR: method not supported\n"); t_reply("500", "sorry, method not supported"); }; } else { # for any reason the transaction could not be created, send error code log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; # in isdngw.conf. Don't change this setting. t_relay(); # end of routing. } route[3] { ############################################ # Voicemail specific configuration - begin # ############################################ if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ if(!vm("/tmp/am_fifo","voicemail")) { log("couldn't contact announcement server\n"); t_reply("500", "couldn not contact announcement server"); }; break; }; if(method=="BYE" || method=="CANCEL") { if(!vm("/tmp/am_fifo","bye")) { log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; break; }; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } route[4] { ###################################### # ivr specific configuration - begin # ###################################### if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if (uri=~"sip:5000@.*") { if (!vm("/tmp/am_fifo", "ivr")) { log("couldn't contact ivr server\n"); t_reply("500", "couldn not contact ivr server"); }; }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact ivr\n"); t_reply("500","could not contact ivr"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } route[5] { # #################################### # conference specific configuration # # #################################### if(t_newtran()){ if(method=="INVITE" || method=="BYE" || method=="CANCEL"){ # send a response right at the start to avoid retransmissions t_reply("100","Trying -- just wait a minute !"); # isdngw only gets activated on invite requests if(method=="INVITE"){ # filename is defined in sems.conf. if(uri=~"sip:6000@.*"){ if(!vm("/tmp/am_fifo","conference")){ log("could not contact conference\n"); t_reply("500","could not contact conference"); }; # Allow the announcement module of sems to be used as well. # This can be useful for testing the conference. } else if(uri=~"sip:5000@.*"){ if(!vm("/tmp/am_fifo","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact announcement"); }; # we dont feel responsible for sip addresses of any other kind, # so send the right error code. } else { t_reply("404","Not Found"); }; # stop routing here, the message is now processed by the media server break; }; # The following handles the call termination, we must pass these requests # to the media server as follows. Again make shure the fifo name and permissions # are set correctly (like im sems.conf). if((method=="BYE")||(method=="CANCEL")){ if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; break; }; # other methods than INVITE, BYE and CANCEL are not handled by this SIP Server # so we sent an error message } else { log("ERROR: method not supported\n"); t_reply("500", "sorry, method not supported"); }; } else { # for any reason the transaction could not be created, send error code log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; # in isdngw.conf. Don't change this setting. t_relay(); # end of routing. } route[6] { ###################################### # announcement configuration - begin # ###################################### if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if (uri=~"sip:7000@.*") { if (!vm("/tmp/am_fifo", "announcement")) { log("couldn't contact ivr server\n"); t_reply("500", "couldn not contact announcement"); }; }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact annoucement\n"); t_reply("500","could not contact annoucement"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); } route[7] { # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); break; }; if (t_newtran()) { t_reply("404", "Not Found"); acc_db_request("404 missed call", "missed_calls"); }; }

3 4

Re: [Serusers] Accounting directly sent to mysql
by Ronald Ramos 27 Apr '04

27 Apr '04

Hi, I can't seem to find a tarball that has a Makefile where I can edit the acc, al I got so far are tarballs that contains the etc, lib, bin, share directories, all the modules are in *.so format now. Where can I download tarball distribution? Thank you Ronald If you use mysql accouting, the CDRs will be stored into the configured mysql database. If you use radius accounting, the CDRs will be sent to your radius server. klaus Nhadie wrote: > Thanks you so much. One more thing, where can I find a sample > configuration that has mysql accounting? Do I need RADIUS for these? If > so, in which database would the records be sent, the database on the > RADIUS or the SER database? > > --------- Original Message -------- > From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> > To: "Nhadie" <nhadie(a)tbgi.net.ph> > Cc: "Serusers" <serusers(a)lists.iptel.org> > Subject: Re: [Serusers] Accounting directly sent to mysql > Date: 25/04/04 10:19 > > > If you have ser isntalled via RPMs, then you can't account to mysql. You > have to compile it yourself. > > btw: forget agout the rpm version and uninstall it completly before > installing ser from source. > > klaus > > PS: please always cc: to the list > > Nhadie wrote: > > Hi, > > > > What if installed it via RPM? Isn't the mysql accounting > compiled? What > > do I need to configure on the ser.cfg to activate mysql accounting? > > Thank You > > > > Ronald > > > > --------- Original Message -------- > > From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> > > To: "ron(a)silverbackasp.com" <ron(a)silverbackasp.com> > > Cc: serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > > Subject: Re: [Serusers] Accounting directly sent to mysql > > Date: 24/04/04 11:55 > > > > > > yes, turn on accounting for mysql. Take a look into the Makefile in > > modules/acc, uncommment the line and re-compile. > > > > klaus > > > > Ronald Ramos wrote: > > > > > Hi, > > > > > > Can I automatically redirect all the time of call and end of call > > to mysql > > > server instead of the syslog? > > > > > > Thank you > > > > > > Ronald > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > <mailto:serusers@lists.iptel.org> > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > ________________________________________________ > > Message sent using UebiMiau 2.7 > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > ________________________________________________ > Message sent using UebiMiau 2.7 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] Date header in INVITEs
by John Todd 27 Apr '04

27 Apr '04

Has anyone already implemented a Date: header for INVITEs in an easy and fast method? I'm looking for a quick plug-in that will send the date in GMT format that looks like this (in strftime format): Date: %a, %d %b %Y %T GMT where %a is day of week (three letter), %d is the day of the month (01-31), %b is abbreviated month name (three digit), %Y is the year as a decimal number including century, and %T is the time in 24 hour notation (HH:MM:SS) I have some SIP UA devices which set the timestamp of outgoing calls on caller ID (analog phones.) SNTP is not an option, so using a Date: header is the next best thing. Maybe someone who has already incorporated this into SER might have something handy and save some time on my end... JT

3 2

[Serusers] problems about the start and the stop of ser
by zhu qiang 27 Apr '04

27 Apr '04

I have installed ser in my computer 1, my computer 1 is in a LAN, and I have another computer 2 in the LAN, I have a msn messenger 4.6 installed in 2, and I configured msn 4.6 according to howto, I fill the IP address of 1 in server name or IP address in msn advanced configures. but when I want to register, it says that the network service is currently unavailable, but I can ping 1 form 2. _________________________________________________________ Do You Yahoo!? 惠普TT游戏剧，玩游戏，中大奖！ http://cn.rd.yahoo.com/mail_cn/tag/SIG=1402c0to2/**http%3A%2F%2Fhp.allyes.c…

1 0

[Serusers] Using parameters from the database
by Shirley 26 Apr '04

26 Apr '04

Hi all, Sorry this question might sound stupid but please someone please kindly help. I need to use one of the subscriber table fields to do call forwarding. Please someone tell me is there any functions that I can use in the ser.cfg to retrieve values from the database as queries to input into the uri for call forwarding. Please help Please help..... Regards, Shirley

2 1

RE: [Serusers] SUBSCRIBE/NOTIFY and SCS-Client; SER sends wrong To header field
by Axel Böger 26 Apr '04

26 Apr '04

Hi, your fix works fine. Thanks for your help. Axel > -----Original Message----- > From: Jan Janak [mailto:jan@iptel.org] > Sent: Sunday, April 25, 2004 10:37 PM > To: Axel Böger > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SUBSCRIBE/NOTIFY and SCS-Client; SER > sends wrong To header field > > > Thanks for the report ! It should be fixed in the CVS. Could you, > please, retry and let me know ? > > Jan. > > On 22-04 09:53, Axel Böger wrote: > > Hi, > > > > I've tried to test the pa module (ser 0.8.12) in > conjunction with the > > SCS Client. > > The SCS Client appends a special generic param: 'ScsId=ch' at the > > From: header in the subscribe message. And the NOTIFY answer from > > ser contains this param but omits the semicolon. So the > NOTIFY message > > will be treated as wrong. > > > > The ser config for SUBSCRIBE method is straight forward: > > if (method=="SUBSCRIBE") > > { > > if (t_newtran()) > > { > > handle_subscription("registrar"); > > }; > > }; # subscribe > > > > The ngrep output for this transaction is: > > # > > U 195.4.20.71:5060 -> 194.97.99.50:5060 > > SUBSCRIBE sip:snom@fn-sip.mcbone.net SIP/2.0..Via: > SIP/2.0/UDP 195.4.20.71. > > .To: sip:snom@fn-sip.mcbone.net..From: > <sip:aboeger@fn-sip.mcbone.net>;tag > > =11b78ed1b717552;ScsId=ch..Call-ID: > dc8b5b893e68391d9b2dfc807f8e3ba2..CSeq: > > 1325665313 SUBSCRIBE..Max-Forwards: 70..Contact: > sip:195.4.20.71..Event: p > > resence..Expires: 3600..Content-Length: 0..User-Agent: > SCS/v3.1.12.33.... > > # > > U 194.97.99.50:5060 -> 195.4.20.71:5060 > > SIP/2.0 200 OK..Via: SIP/2.0/UDP 195.4.20.71..To: > sip:snom@fn-sip.mcbone.ne > > t;tag=91083b2aa33cb6d5c9260125dc95d5f0-efd4..From: > <sip:aboeger@fn-sip.mcb > > one.net>;tag=11b78ed1b717552;ScsId=ch..Call-ID: > dc8b5b893e68391d9b2dfc807f8 > > e3ba2..CSeq: 1325665313 SUBSCRIBE..Server: Sip EXpress > router (0.8.12a (i38 > > 6/linux))..Content-Length: 0..Warning: 392 > 194.97.99.50:5060 "Noisy feedbac > > k tells: pid=26439 req_src_ip=195.4.20.71 > req_src_port=5060 in_uri=sip:sno > > m(a)fn-sip.mcbone.net out_uri=sip:snom@fn-sip.mcbone.net > via_cnt==1".... > > # > > U 194.97.99.50:5060 -> 195.4.20.71:5060 > > NOTIFY sip:195.4.20.71 SIP/2.0..Via: SIP/2.0/UDP > 194.97.99.50;branch=z9hG4b > > K6d55.e01a2e91.0..To: > <sip:aboeger@fn-sip.mcbone.net>ScsId=ch;tag=11b78ed1b > > 717552..From: > sip:snom@fn-sip.mcbone.net;tag=91083b2aa33cb6d5c9260125dc95d5 > > f0-efd4..CSeq: 1 NOTIFY..Call-ID: > dc8b5b893e68391d9b2dfc807f8e3ba2..Content > > -Length: 302..User-Agent: Sip EXpress router(0.8.12a > (i386/linux))..Event: > > presence..Content-Type: > application/xpidf+xml..Subscription-State: active;e > > xpires=3600....<?xml version="1.0"?>..<!DOCTYPE presence > PUBLIC "//IETF//DT > > D RFCxxxx XPIDF 1.0//EN" > "xpidf.dtd">..<presence>..<presentity uri="sip:sno > > m(a)fn-sip.mcbone.net;method=SUBSCRIBE"/>..<atom > id="9r28r49">..<address uri= > > "sip:snom@fn-sip.mcbone.net">..<status > status="closed"/>..</address>..</ato > > m>..</presence>.. > > # > > U 195.4.20.71:5060 -> 194.97.99.50:5060 > > SIP/2.0 400 Syntax Error..Call-ID: > dc8b5b893e68391d9b2dfc807f8e3ba2..CSeq: > > 1 NOTIFY..From: > sip:snom@fn-sip.mcbone.net;tag=91083b2aa33cb6d5c9260125dc95 > > d5f0-efd4..Via: SIP/2.0/UDP > 194.97.99.50;branch=z9hG4bK6d55.e01a2e91.0..Con > > tent-Length: 136..Content-Type: text/plain....While > parsing "To": header ex > > pected CRLF.. Missing or invalid To header.. Missing or > invalid To header.. > > Missing or invalid To header.. > > # > > > > For me it seems to be an error in ser. Have anyone an idea > for a workaround? > > > > Regards, > > Axel > > > > > > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

Re: [Serusers] Accounting directly sent to mysql
by Nhadie 26 Apr '04

26 Apr '04

2 1

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