Charles, you can read through the README that is included with mediaproxy (the server) and the README that ins included in the ser mediaproxy module.
Basically you need to first start the mediaproxy Python server. Once that is running you can start ser.
rtpgenerator.py is just a stress testing tool.
dnssrv is not required if you're just using one mediaproxy instance. If you ever get to multi-host deployments, then you may want to use dns srv records for load balancing.
P
On Tue, 22 Feb 2005 00:30:40 +0800, Charles Wang lazy.charles@gmail.com wrote:
Dear Paul :
If I want to run a mediaproxy mode, what do I have to run? serctl and mediaproxy.py and what? Is it necessary about rtpgenerator.py? And is it is necessary to setup a SRV service on DNS?
After your explain, and I download the last Python and run mediaproxy well.
But my call busy forward is still failed.
On Mon, 21 Feb 2005 08:02:03 -0500, Java Rockx javarockx@gmail.com wrote:
Are you saying that you have to NATed clients that can communicate?
If so, the reason is that they are being the same firewall and thus can directly reach each other.
Regards, Paul
On Mon, 21 Feb 2005 20:54:33 +0800, Charles Wang lazy.charles@gmail.com wrote:
Dear Paul:
I guess that you are right. But I can not understand why I can make a call from two different NATs? Don't they communicate via my ser? Or just signal channel via ser?
Thank you for your explain.
Charles
On Mon, 21 Feb 2005 07:41:58 -0500, Java Rockx javarockx@gmail.com wrote:
MediaProxy is a two-part project. SER only includes the interface to the actual mediaproxy. You must install the actual mediaproxy server, which is written in Python.
http://www.ag-projects.com/MediaProxy.html
Regards, Paul
On Mon, 21 Feb 2005 20:26:19 +0800, Charles Wang lazy.charles@gmail.com wrote:
Dear Paul:
So surprised it is. I never see any executable file or script named "mediaproxy.py" in my source files. I guess that the mediaproxy is only a module including in "sip_router/modules/mediaproxy/mediaproxy.so". When I can get a executable version of mediaproxy or where the mediaproxy.py is?
And also I miss the "proxydispatcher". I can find a directory named "dispatcher" under my sip_router/modules.
I usually start my ser with "/usr/local/sbin/serctl start".
Please let me know where I can get these files what you said.
On Mon, 21 Feb 2005 07:04:48 -0500, Java Rockx javarockx@gmail.com wrote:
Charles,
How are you starting mediaproxy? Below is the start script I use.
Regards, Paul
#!/bin/sh # # chkconfig: 2345 90 20 # description: VoIP RTP Proxy Server # # processname: mediaproxy # pidfile: /var/run/mediaproxy.pid
# source function library . /etc/rc.d/init.d/functions
PATH=/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin
INSTALL_DIR="/usr/local" RUNTIME_DIR="/var/run"
PROXY="$INSTALL_DIR/mediaproxy/mediaproxy.py" DISPATCHER="$INSTALL_DIR/mediaproxy/proxydispatcher.py" PROXY_PID="$RUNTIME_DIR/mediaproxy.pid" DISPATCHER_PID="$RUNTIME_DIR/proxydispatcher.pid"
# Options for mediaproxy and dispatcher. Do not include --pid <pidfile> # --pid <pidfile> will be added automatically if needed. PROXY_OPTIONS="--ip=24.48.42.20 --listen=127.0.0.1" DISPATCHER_OPTIONS="domain://sip.mycompany.com"
NAME="mediaproxy" DESC="SER MediaProxy server"
test -f $PROXY || exit 0 test -f $DISPATCHER || exit 0
if [ "$PROXY_PID" != "/var/run/mediaproxy.pid" ]; then PROXY_OPTIONS="--pid $PROXY_PID $PROXY_OPTIONS" fi if [ "$DISPATCHER_PID" != "/var/run/proxydispatcher.pid" ]; then DISPATCHER_OPTIONS="--pid $DISPATCHER_PID $DISPATCHER_OPTIONS" fi
start() { echo -n "Starting $DESC: $NAME" $PROXY $PROXY_OPTIONS $DISPATCHER $DISPATCHER_OPTIONS echo "." }
stop () { echo -n "Stopping $DESC: $NAME" kill `cat $PROXY_PID` kill `cat $DISPATCHER_PID` echo "." }
case "$1" in start) start ;; stop) stop ;; restart|force-reload) stop #sleep 1 start ;; *) echo "Usage: /etc/init.d/$NAME {start|stop|restart|force-reload}" >&2 exit 1 ;; esac
exit 0
On Mon, 21 Feb 2005 13:39:10 +0800, Charles Wang lazy.charles@gmail.com wrote: > Dear ALL: > > I make the UA 1033 as busy status(pick up the phone and do nothing). > > Then I try to make a call from 1011 to 1033. It should be redirect to > a PSTN phone number( sip:0939749xxx@ser.xxx.net.tw ). > I find it call failure and jump to failure_route[1], then swicth to > route[3](for PSTN). > But when I dump the package using ngrep. I can't find it try to > connect PSTN trunk( xxx.xxx.190.243 is a CISCO 5300). > I can find "SER: Connecting to PSTN....." message list at log file. > Then the log display "SER: SIP Call On-Net section route(2)" message. > It means that the call return to route[2] then failed. > I guess that it stops or jump out route[3] after rewritehost(xxx.xxx.190.243). > > Can anyone help me to trace the bug in the route[3]? > > Best Regard > Charles > > Subset of ser.cfg about route[3: > ------------------------------------------------------------------------------------------------------- > route[3] { > log(1, "SER: Demestic Call Off-Net section route(3)\n"); > > # All Domestic Calls Go To CISCO 5300 > if (method=="INVITE") { > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > break; > } else if (!check_from()) { > log(1, "Spoofed SIP call attempt"); > sl_send_reply("403", "Use From=ID"); > break; > } else if (!(is_from_local() || is_uri_host_local())) { > sl_send_reply("403", "Please register to use our service"); > break; > }; > # enable caller id blocking for PSTN calls > if (isflagset(25)) { > append_rpid_hf(); > }; > }; > # SIP->PSTN calls get 45 seconds to timeout > log(1, "SER: Connecting to PSTN.....\n"); > avp_write("i:45", "inv_timeout"); > rewritehost("61.220.190.243"); > if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && > !search("^Route:")){ > sl_send_reply("479", "We don't forward to private IP addresses"); > break; > }; > if (method=="INVITE" || method=="ACK") { > use_media_proxy(); > }; > if (isflagset(31)) { # is voice mail? > t_on_failure("1"); > }; > t_on_reply("1"); > if (!t_relay()) { > if (method=="INVITE" || method=="ACK") { > end_media_session(); > }; > sl_reply_error(); > }; > } > > Dump using ngrep -d eth0 -W byline port 5060 > ------------------------------------------------------------------ > Notes: xxx.xxx.190.248 : is SER sip proxy > xxx.xxx.13.49 : is NAT > sip:1033@xxx.xxx.13.49:33536 : location of 1033 > sip:1011@xxx.xxx.13.49:35700 : location of 1011 > > interface: eth0 (xxx.xxx.190.240/255.255.255.240) > filter: ip and ( port 5060 ) > # > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > INVITE sip:1033@ser.xxx.net.tw SIP/2.0. > Via: SIP/2.0/UDP > 10.18.1.70:5060;rport;branch=z9hG4bK2754265110CD4289ACDA9EA0769C1A8F. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > To: sip:1033@ser.xxx.net.tw. > Contact: sip:1011@10.18.1.70:5060. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > CSeq: 26231 INVITE. > Max-Forwards: 70. > Content-Type: application/sdp. > User-Agent: X-PRO build 1082. > Content-Length: 264. > . > v=0. > o=1011 46237906 46237906 IN IP4 10.18.1.70. > s=X-PRO. > c=IN IP4 10.18.1.70. > t=0 0. > m=audio 8000 RTP/AVP 8 3 98 97 101. > a=rtpmap:8 pcma/8000. > a=rtpmap:3 gsm/8000. > a=rtpmap:98 iLBC/8000. > a=rtpmap:97 speex/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > > # > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:35700 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP > 10.18.1.70:5060;rport=35700;branch=z9hG4bK2754265110CD4289ACDA9EA0769C1A8F;received=xxx.xxx.13.49. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > To: sip:1033@ser.xxx.net.tw;tag=67771a809cdfb71129a699a517fbb1f0.7608. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > CSeq: 26231 INVITE. > Proxy-Authenticate: Digest realm="ser.xxx.net.tw", > nonce="42196d5483841de0d6b79c7dfb4156e2c932e4cb". > Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). > Content-Length: 0. > Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=4199 > req_src_ip=xxx.xxx.13.49 req_src_port=35700 > in_uri=sip:1033@ser.xxx.net.tw out_uri=sip:1033@xxx.xxx.13.49:33536 > via_cnt==1". > . > > # > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > ACK sip:1033@ser.xxx.net.tw SIP/2.0. > Via: SIP/2.0/UDP > 10.18.1.70:5060;rport;branch=z9hG4bK2754265110CD4289ACDA9EA0769C1A8F. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > To: sip:1033@ser.xxx.net.tw;tag=67771a809cdfb71129a699a517fbb1f0.7608. > Contact: sip:1011@10.18.1.70:5060. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > CSeq: 26231 ACK. > Max-Forwards: 70. > Content-Length: 0. > . > > # > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > INVITE sip:1033@ser.xxx.net.tw SIP/2.0. > Via: SIP/2.0/UDP > 10.18.1.70:5060;rport;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > To: sip:1033@ser.xxx.net.tw. > Contact: sip:1011@10.18.1.70:5060. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > CSeq: 26232 INVITE. > Proxy-Authorization: Digest > username="1011",realm="ser.xxx.net.tw",nonce="42196d5483841de0d6b79c7dfb4156e2c932e4cb",response="7004beab12b3ac6874c5cd86e5659930",uri="sip:1033@ser.xxx.net.tw". > Max-Forwards: 70. > Content-Type: application/sdp. > User-Agent: X-PRO build 1082. > Content-Length: 264. > . > v=0. > o=1011 46238507 46238507 IN IP4 10.18.1.70. > s=X-PRO. > c=IN IP4 10.18.1.70. > t=0 0. > m=audio 8000 RTP/AVP 8 3 98 97 101. > a=rtpmap:8 pcma/8000. > a=rtpmap:3 gsm/8000. > a=rtpmap:98 iLBC/8000. > a=rtpmap:97 speex/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > > # > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:35700 > SIP/2.0 100 trying -- your call is important to us. > Via: SIP/2.0/UDP > 10.18.1.70:5060;rport=35700;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC;received=xxx.xxx.13.49. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > To: sip:1033@ser.xxx.net.tw. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > CSeq: 26232 INVITE. > Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). > Content-Length: 0. > Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=4201 > req_src_ip=xxx.xxx.13.49 req_src_port=35700 > in_uri=sip:1033@ser.xxx.net.tw out_uri=sip:1033@xxx.xxx.13.49:33536 > via_cnt==1". > . > > # > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:33536 > INVITE sip:1033@xxx.xxx.13.49:33536 SIP/2.0. > Record-Route: sip:xxx.xxx.190.248;ftag=3281269171;lr=on. > Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bKa7ac.b59d9107.0. > Via: SIP/2.0/UDP > 10.18.1.70:5060;received=xxx.xxx.13.49;rport=35700;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > To: sip:1033@ser.xxx.net.tw. > Contact: sip:1011@xxx.xxx.13.49:35700. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > CSeq: 26232 INVITE. > Proxy-Authorization: Digest > username="1011",realm="ser.xxx.net.tw",nonce="42196d5483841de0d6b79c7dfb4156e2c932e4cb",response="7004beab12b3ac6874c5cd86e5659930",uri="sip:1033@ser.xxx.net.tw". > Max-Forwards: 16. > Content-Type: application/sdp. > User-Agent: X-PRO build 1082. > Content-Length: 264. > . > v=0. > o=1011 46238507 46238507 IN IP4 10.18.1.70. > s=X-PRO. > c=IN IP4 10.18.1.70. > t=0 0. > m=audio 8000 RTP/AVP 8 3 98 97 101. > a=rtpmap:8 pcma/8000. > a=rtpmap:3 gsm/8000. > a=rtpmap:98 iLBC/8000. > a=rtpmap:97 speex/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > > # > U xxx.xxx.13.49:33536 -> xxx.xxx.190.248:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bKa7ac.b59d9107.0. > Via: SIP/2.0/UDP > 10.18.1.70:5060;received=xxx.xxx.13.49;rport=35700;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC. > Record-Route: sip:xxx.xxx.190.248;ftag=3281269171;lr=on. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > CSeq: 26232 INVITE. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > To: sip:1033@ser.xxx.net.tw;tag=A8VnZRzqB2nyNgYQ. > Contact: sip:1033@10.18.1.102:1718. > Content-Length: 0. > . > > # > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:33536 > ACK sip:1033@xxx.xxx.13.49:33536 SIP/2.0. > Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bKa7ac.b59d9107.0. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > To: sip:1033@ser.xxx.net.tw;tag=A8VnZRzqB2nyNgYQ. > CSeq: 26232 ACK. > User-Agent: Sip EXpress router(0.10.99-dev0 (i386/linux)). > Content-Length: 0. > . > > # > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > . > > # > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > . > > # > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:35700 > SIP/2.0 408 Request Timeout. > Via: SIP/2.0/UDP > 10.18.1.70:5060;rport=35700;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC;received=xxx.xxx.13.49. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > To: sip:1033@ser.xxx.net.tw;tag=5f573bbafd260ada15def89f1b1724a2-6b7e. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > CSeq: 26232 INVITE. > Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). > Content-Length: 0. > Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=4203 > req_src_ip=xxx.xxx.13.49 req_src_port=35700 > in_uri=sip:1033@ser.xxx.net.tw out_uri=sip:1033@xxx.xxx.13.49:33536 > via_cnt==0". > . > > # > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > ACK sip:1033@ser.xxx.net.tw SIP/2.0. > Via: SIP/2.0/UDP > 10.18.1.70:5060;rport;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC. > From: 1011 sip:1011@ser.xxx.net.tw;tag=3281269171. > To: sip:1033@ser.xxx.net.tw;tag=5f573bbafd260ada15def89f1b1724a2-6b7e. > Contact: sip:1011@10.18.1.70:5060. > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8@10.18.1.70. > CSeq: 26232 ACK. > Max-Forwards: 70. > Content-Length: 0. > . > > # > # > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:35700 > .... > # > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:33536 > .... > # > U xxx.xxx.13.49:33536 -> xxx.xxx.190.248:5060 > ................ > # > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > . > > _______________________________________________ > Serusers mailing list > serusers@lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Dear Paul:
Yes, I have study the readme, and both start daemon of mediaproxy Python server and start my ser with mediaproxy module supported.
I guess that my setting done. And I wanna forward a noanswer call from UA(1011) to PSTN(0939749xxx) via UA(1033).
In my log, the call is ringing from UA1011 to UA1033 first. Then after a few seconds, failure_route[1] trigger, and it shall be forward(busy) to a PSTN number. I define this number at usr_preferences table and attribute:fwdbusy, value:sip:0939749xxx@ser.xxx.net.tw.
I can find it try to connect to my CISCO trunk (xxx.xxx.190.243), but why is its uri/username still 1033 ? I guess that something is wrong in my ser.cfg.
Charles
My log: ------------------------------------------------------------------------------------------------------------------ SER: SIP Call On-Net section route(2) command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local 61.229.13.49 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag: SER: SIP Call On-Net section route(2) PDT:prefix2domain: no prefix found in [1033] Time:[Tue Feb 22 00:50:57 2005] Method:<INVITE> r-uri:1033@ser.xxx.net.tw IP:<61.217.225.225> From:sip:1011@ser.xxx.net.tw To:sip:1033@ser.xxx.net.tw sip:1011@192.168.11.4:5060 SER: a INT user SER: BLIND CALL FORWARDING SER: Look aliases SER: Look location SER isflagset (sip) SER: Look aliases SER: Look location SER isflagset (sip) SER: SIP Call On-Net section route(2) command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local 61.229.13.49 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag: domain ser.xxx.net.tw doesn't define any mediaproxy. will use default mediaproxy for this call. command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local 61.229.13.49 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag:,dispatcher session 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4: started. listening on xxx.xxx.190.248:35006 command execution time: 6.19 ms forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35006' command execution time: 92.46 ms SER: Failure Route section failure_route(1) SER: fork to fwdnoanswer SER: No Answer Failure and Jump to route(3) SER: Demestic Call Off-Net section route(3) SER: Connecting to PSTN..... <================== after this, uri should be "0939749xxx@ser.xxx.net.tw" ????? command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag: domain ser.xxx.net.tw doesn't define any mediaproxy. <==== this phone number should not be 1033, it should be 0939749xxx (a PSTN number) ?????? will use default mediaproxy for this call. command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag:,dispatcher command execution time: 0.34 ms forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35006' command execution time: 3846.84 ms PDT:prefix2domain: no prefix found in [1033] SER: SIP Call On-Net section route(2) session 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4: ended (did timeout).
Still in condition. Help me please.
Best Regards Charles
On Tue, 22 Feb 2005 01:21:44 +0800, Charles Wang lazy.charles@gmail.com wrote:
Dear Paul:
Yes, I have study the readme, and both start daemon of mediaproxy Python server and start my ser with mediaproxy module supported.
I guess that my setting done. And I wanna forward a noanswer call from UA(1011) to PSTN(0939749xxx) via UA(1033).
In my log, the call is ringing from UA1011 to UA1033 first. Then after a few seconds, failure_route[1] trigger, and it shall be forward(busy) to a PSTN number. I define this number at usr_preferences table and attribute:fwdbusy, value:sip:0939749xxx@ser.xxx.net.tw.
I can find it try to connect to my CISCO trunk (xxx.xxx.190.243), but why is its uri/username still 1033 ? I guess that something is wrong in my ser.cfg.
Charles
My log:
SER: SIP Call On-Net section route(2) command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local 61.229.13.49 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag: SER: SIP Call On-Net section route(2) PDT:prefix2domain: no prefix found in [1033] Time:[Tue Feb 22 00:50:57 2005] Method:<INVITE> r-uri:1033@ser.xxx.net.tw IP:<61.217.225.225> From:sip:1011@ser.xxx.net.tw To:sip:1033@ser.xxx.net.tw sip:1011@192.168.11.4:5060 SER: a INT user SER: BLIND CALL FORWARDING SER: Look aliases SER: Look location SER isflagset (sip) SER: Look aliases SER: Look location SER isflagset (sip) SER: SIP Call On-Net section route(2) command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local 61.229.13.49 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag: domain ser.xxx.net.tw doesn't define any mediaproxy. will use default mediaproxy for this call. command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local 61.229.13.49 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag:,dispatcher session 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4: started. listening on xxx.xxx.190.248:35006 command execution time: 6.19 ms forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35006' command execution time: 92.46 ms SER: Failure Route section failure_route(1) SER: fork to fwdnoanswer SER: No Answer Failure and Jump to route(3) SER: Demestic Call Off-Net section route(3) SER: Connecting to PSTN..... <================== after this, uri should be "0939749xxx@ser.xxx.net.tw" ????? command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag: domain ser.xxx.net.tw doesn't define any mediaproxy. <==== this phone number should not be 1033, it should be 0939749xxx (a PSTN number) ?????? will use default mediaproxy for this call. command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536,totag:,dispatcher command execution time: 0.34 ms forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35006' command execution time: 3846.84 ms PDT:prefix2domain: no prefix found in [1033] SER: SIP Call On-Net section route(2) session 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4: ended (did timeout).
Wow,
I didn't know SER was so complicating, perhaps I should use asterisk, would asterisk support thousands of users much like SER?
I'm finding this all too much and all I want to do is allow people to signup and call eachother simular to iptel, but with NZ servers
Barry
-----Original Message----- From: serusers-bounces@lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Charles Wang Sent: Tuesday, 15 March 2005 5:20 a.m. To: serusers@lists.iptel.org Subject: Re: [Serusers] error: mediaproxy/sendMediaproxyCommand(): can'tconnect to MediaProxy
Still in condition. Help me please.
Best Regards Charles
On Tue, 22 Feb 2005 01:21:44 +0800, Charles Wang lazy.charles@gmail.com wrote:
Dear Paul:
Yes, I have study the readme, and both start daemon of mediaproxy Python server and start my ser with mediaproxy module supported.
I guess that my setting done. And I wanna forward a
noanswer call from
UA(1011) to PSTN(0939749xxx) via UA(1033).
In my log, the call is ringing from UA1011 to UA1033 first. Then after a few seconds, failure_route[1] trigger, and it shall be forward(busy) to a PSTN number. I define this number at usr_preferences table and
attribute:fwdbusy,
value:sip:0939749xxx@ser.xxx.net.tw.
I can find it try to connect to my CISCO trunk
(xxx.xxx.190.243), but
why is its uri/username still 1033 ? I guess that something
is wrong
in my ser.cfg.
Charles
My log:
SER: SIP Call On-Net section route(2) command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local 61.229.13.49 remote X-PRO=20build=201082
info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:3 09679536,totag:
SER: SIP Call On-Net section route(2) PDT:prefix2domain: no prefix found in [1033] Time:[Tue Feb
22 00:50:57
2005] Method:<INVITE> r-uri:1033@ser.xxx.net.tw
IP:<61.217.225.225>
From:sip:1011@ser.xxx.net.tw To:sip:1033@ser.xxx.net.tw sip:1011@192.168.11.4:5060 SER: a INT user SER: BLIND CALL FORWARDING SER: Look aliases SER: Look location SER isflagset (sip) SER: Look aliases SER: Look location SER isflagset (sip) SER: SIP Call On-Net section route(2) command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local 61.229.13.49 remote X-PRO=20build=201082
info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:3 09679536,totag:
domain ser.xxx.net.tw doesn't define any mediaproxy. will use default mediaproxy for this call. command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local 61.229.13.49 remote X-PRO=20build=201082
info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536
,totag:,dispatcher session 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4: started. listening on xxx.xxx.190.248:35006 command execution time: 6.19 ms forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35006' command execution time: 92.46 ms SER: Failure Route section failure_route(1) SER: fork to fwdnoanswer SER: No Answer Failure and Jump to route(3) SER: Demestic Call Off-Net section route(3) SER: Connecting to PSTN..... <================== after this, uri should be "0939749xxx@ser.xxx.net.tw" ????? command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082
info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:3 09679536,totag:
domain ser.xxx.net.tw doesn't define any mediaproxy. <====
this phone
number should not be 1033, it should be 0939749xxx (a PSTN number) ?????? will use default mediaproxy for this call. command request 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4 192.168.11.4:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082
info=from:1011@ser.xxx.net.tw,to:1033@ser.xxx.net.tw,fromtag:309679536
,totag:,dispatcher command execution time: 0.34 ms forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35006' command execution time: 3846.84 ms PDT:prefix2domain: no prefix found in [1033] SER: SIP Call On-Net section route(2) session 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session 88D1D3FD-0193-41D0-990C-06915D53D1BF@192.168.11.4: ended
(did timeout).
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