sr-users February 2005

sr-users@lists.kamailio.org

215 participants
467 discussions

[Serusers] Carrier-grade framework for SER

by Andreas Granig

Hi guys, I'd like to propose another possibility for a highly-available and scalable system design based on SER: The problems I've encountered for scalable systems are: - Distribution of the user location and alias location among the nodes (user location is based on registrations, alias location comes from web interfaces and is used for call forwarding). - Reloading up to date location tables after breakdown and recovery of a node So I'm just thinking loud about the following provisioning system: - Write a client which fulfills the this demands: - Receive one or more locations from SER via a SER module or from a web application and distribute them to other known clients. Take care of retransmissions if a client isn't reachable or reports a temporary failure. - Receive one or more locations from other clients and write them into the SER FIFO. If writing into the FIFO fails, try to write directly into the database (location-table, alias-table etc.). Report a temporary failure if this also fails. Maybe a centralized server should be used which receives the locations from the clients and distributes them to other clients, so that the nodes just know about the server and nothing about other nodes. This would make integration of new nodes easy. On the other hand, it's another single point of failure, so a decentralized solution should be considered. But that would mean that you've to inform every node about the existence of a new node. The protocol used between the nodes should be simple and fast. So I think SOAP drops out here. Maybe XMLRPC or ICE (http://www.zeroc.com/ice.html) could be used. One might think now why not just use replication on SIP layer, but t_replicate only supports one peer and you've no possibility to get locations on a node while it's down. Replication of other location tables like the alias-table is also not possible. I'd be willing to release these parts as GPL for creating an open framework for carrier-grade SER integration, so any feedback, improvements or flames are highly welcome. Cheers, Andy

16 years, 5 months

[Serusers] Modifying "From" header field of an INVITE message

by scm-j＠nuntius.com

I am relatively new to SER. My SIP service provider requires me to present the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows: From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx> They want it in E.164 format as follows: From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx> I see function only to modify the SIP URI but not the other header or SDP fields. Could someone help me with a code snippet that I can use in ser.cfg to do this conversion? Regards, SCM

16 years, 10 months

[Serusers] Authenticate to Active Directory

by Bob McCormick

Does anyone know if there is a way to authenticate SER to MS Active Directory? I'd like to use an existing user database instead of having to manage users myself, and most of our users are in Active Directory. Has anyone tried this? What are most SER users using for authentication? Is anyone here tying into any kind of existing organization user database?

17 years, 3 months

[Serusers] Over-long fields can crash SER with PostgreSQL

by Karl A. Krueger

We are using SER with the PostgreSQL module, so that SER and our Asterisk PSTN gateway and voice mail system can use the same back-end database for user records. (Anyone who's interested in how we did this, let me know -- it's still a little shaky, but it pretty much works.) Recently we have encountered a problem: it seems that if any database error occurs, SER hangs or crashes. We have so far been able to mostly work around these conditions by being Really Careful with the database, but it is a little unsettling. The database schema we're working from is practically identical to the MySQL schema that ships with SER. Notably for this problem the lengths of all CHARACTER VARYING fields are the same. The User-Agent field is CHARACTER VARYING(50), for instance. So ... today boss got a copy of Xten's "eyeBeam" software, set it up to register to our SER ... and SER promptly locked up. I looked in the PostgreSQL database log and found this: Feb 10 13:39:32 mabell postgres[13058]: [267] ERROR: value too long for type character varying(50) Whoops. SER tried to insert eyeBeam's User-Agent data, PostgreSQL returned a data integrity error, and SER went bye-bye. So I did a little ALTER TABLE and adjusted the field to length 128 ... then put this in ser.cfg: # Disallow long user-agent if (search("^User-Agent: .{100,}") ) { log(1, "LOG: User agent too long\n"); sl_send_reply("479", "User agent too long!"); break; }; But this is obviously a kludge ... and doesn't fix the problem of one of the *other* fields being over-long. Anyone have any thoughts on this? Robustness patches for the PostgreSQL module or usrloc? Better ser.cfg workarounds to protect the world from malformed or over-long SIP header fields? Should I just change all the database fields to TEXT instead of CHARACTER VARYING(whatever) ? -- Karl A. Krueger <kkrueger(a)whoi.edu> Network Security -- Linux/Unix Systems Support -- Etc. Woods Hole Oceanographic Institution

17 years, 7 months

[Serusers] ENUM variable TLD?

by David

Hi Jiri. Two questions please. 1.) Jiri Kuthan HYPERLINK "mailto:serusers%40iptel.org?Subject=%5BSerusers%5D%20ENUM%20variable%20TLD% 3F&In-Reply-To=4032B769.8000804%40august.net"jiri at iptel.org Wed Feb 18 02:25:07 CET 2004 * Previous message: HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/006025.html"[Seruser s] ENUM variable TLD? * Next message: HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/006022.html"[Seruser s] current uri * Messages sorted by: HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/date.html#6026"[ date ] HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/thread.html#6026"[ thread ] HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/subject.html#6026"[ subject ] HYPERLINK "http://lists.iptel.org/pipermail/serusers/2004-February/author.html#6026"[ author ] _____ on development branch there is an updated enum module which allows you to use private enum trees the way you are asking. If you need it, there should be no conflicts if you take devel version of enum and put it in your 8.12 source tree. -jiri At 01:52 AM 2/18/2004, Greg Fausak wrote: >Normally I do enum lookups inside my >domain, however, I want to be able to do >an enum lookup at other TLD. Is there a >way to do that? Maybe enum_query("e164.other.com")?? > I have downloaded the latest code as you suggested but have not been able to figure out how to specify a different TLD. I have looked a documentation but that has not helped either. Can you please tell me what the function I need is called? 2.) Is there an updated ENUM module that would allow me to preform an ENUM query based on a spcified prefix? In other words, what if I would like to preform an ENUM query when a number comes in with a "**" instead of the "+". I have modified the c code to do this for me but was wondering if you have released a "production" version of this code. Thanks, David Schwartz --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.796 / Virus Database: 540 - Release Date: 11/13/2004

17 years, 11 months

[Serusers] SER Reliability

by Ricardo Martinez

Hello list. I have a question about reliability in SER. Suppose that for certain reason, the process "ser" fails, and therefore the SIP Server is down. Is there a way to keep trace of the "ser" process status?? and if the process is not running start a script to restart the process. I know that maybe the "crond" command could be useful for this purpose. Does someone have any clue about this? Or maybe SER has his own "crond" module or something.? I really need and advice about this issue. Thanks in advance Regards, Ricardo Martinez Ogalde.-

17 years, 11 months

[Serusers] parallel forking best practice?

by Jev

Hi All, For the 0_9_0 branch, what is the 'best practice' for implementing parallel forking? We used to be able to point one alias at several accounts, but that feature has regressed. Thanks, -Jev

17 years, 11 months

[Serusers] INVITE timers in/out

by Daniel Poulsen

Hi, I have in my ser.cfg: modparam("tm","fr_inv_timer",40) set as a timer so that INVITEs coming from PSTN callers go to voicemail after 40 seconds. This works great for inbound called, however it seems to be a limitation for outbound calls. If the outbound INVITE doesn't receive an ACK in that time period, the call gets cancelled and drops to fast busy. Is there a way around this? Can I set different timers for inbound and outbound calls? 40 seconds seems to be a good amount time so that most outbound calls get answered by voicemail or actual people, but occastionally users will still get a fast busy. Any advice is greatly appreciated. Dan

18 years

[Serusers] call busy forward failed

by Charles Wang

Dear ALL: I try to make a call from UA1 to UA2, and UA2 is busy. So the call will forward to another UA3. UA1 ==> UA2(busy) ==> UA3 But when I watch the log, I find that miss the following log (diff with call from UA1 to UA2 and directly forward to UA3) UA1 ==> UA2(callfwd) ==> UA3 Feb 22 12:15:07 ser mediaproxy[18476]: session 16AA8C75-5EAC-403C-85DA-C5D9BDFD15C7(a)10.18.1.70: started. listening on xxx.xxx.190.248:35026 When I make a callfwd call, the call will run route[2] but not failure_route[1]. UA1==>UA2(callfwd)==>route[2]==>UA3 And I make a fwdbusy call, the call will run route[2] then pass failure_route[1], and return to route[2]. UA1==>UA2(fwdbusy)==>failure_route[1]==>route[2] =XXX=> UA3 ???? Why does the method be failed? Do I must "end_media_session()" before start a busy call? How can I modify it? My snippet ser.cfg : -------------------------------------------------------------------------------------------------------------- route[2] { log(1, "SER: SIP Call On-Net section route(2)\n"); if ((method=="INVITE") && !allow_trusted()) { 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; }; }; 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(); }; t_on_failure("1"); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { log(1, "SER: Failure Route section failure_route(1)\n"); # if caller hung up then don't sent to voicemail if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("486")) { # forward busy is flag 26 if (avp_pushto("$ruri", "s:fwdbusy")) { log(1, "SER: fork to fwdbusy\n"); avp_delete("s:fwdbusy"); append_branch(); resetflag(26); # test for domestic PSTN gateway if (uri=~"^sip:0[0-9]{9}@") { # if (avp_check("$fwd_busy_type", "eq/dom/i")) { # test for domestic PSTN gateway log(1, "SER: Busy Failure and Jump to route(3)\n"); route(3); } else if (uri=~"^sip:002[1-9][0-9]*@") { # } else if (avp_check("$fwd_busy_type", "eq/int/i")) { # test for international PSTN gateway log(1, "SER: Busy Failure and Jump to route(6)\n"); route(6); } else { # default to sip call log(1, "SER: Busy Failure and Jump to route(2)\n"); route(2); }; break; }; }; # here we can have either voicemail __OR__ forward no answer if (isflagset(27) && t_check_status("408")) { # forward no answer is flag 27 if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { avp_pushto("$ruri", "s:fwdnoanswer"); log(1, "SER: fork to fwdnoanswer\n"); avp_delete("s:fwdnoanswer"); append_branch(); resetflag(27); if (uri=~"^sip:0[0-9]{9}@") { # if (avp_check("$fwd_no_answer_type", "eq/dom/i")) { # test for domestic PSTN gateway log(1, "SER: No Answer Failure and Jump to route(3)\n"); route(3); } else if (uri=~"^sip:002[1-9][0-9]*@") { # } else if (avp_check("$fwd_no_answer_type", "eq/int/i")) { # test for international PSTN gateway log(1, "SER: No Answer Failure and Jump to route(6)\n"); route(6); } else { # default to sip call log(1, "SER: No Answer Failure and Jump to route(2)\n"); route(2); }; break; }; } else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) { avp_delete("$voicemail"); log(1, "SER: No Answer Failure and Jump to route(4)\n"); route(4); break; }; }

18 years

Re: [Serusers] error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy

by Java Rockx

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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(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 > > > > > > > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

18 years

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sr-users February 2005