Hi
I´ve just installed cvs version of sems and trying with ser_0.8.14, please see the errors below.
Rafael
sems -f /usr/local/etc/sems/sems.conf.default &
[1] 31598
[root@gkproxy01 sems]#
Configuration:
configuration file: /usr/local/etc/sems/sems.conf.default
Ser's FIFO: /tmp/vm_ser_fifo
our FIFO: /tmp/am_fifo
plug-in path: /usr/local/lib/sems/plug-in/
daemon mode: 0
local IP: 200.110.2.131
(31599) WARNING: onLoad (Conference.cpp:77): no join_sound specified in configuration
(31599) WARNING: onLoad (Conference.cpp:78): file for module conference.
(31599) WARNING: onLoad (Conference.cpp:84): no drop_sound specified in configuration
(31599) WARNING: onLoad (Conference.cpp:85): file for module conference.
(31599) WARNING: reloadModuleConfig (SemsConfiguration.cpp:66): no configuration found for module isdngw, maybe you want to specify config.isdngw in config file.
(31599) WARNING: reloadModuleConfig (SemsConfiguration.cpp:66): no configuration found for module semstalk, maybe you want to specify config.semstalk in config file.
(31599) WARNING: onLoad (Semstalk.cpp:70): no configuration specified for module ttsannounce.
(31599) WARNING: onLoad (Semstalk.cpp:71): using default configuration:
(31599) WARNING: onLoad (Semstalk.cpp:77): FestivalServer = "localhost"
(31599) WARNING: onLoad (Semstalk.cpp:78): FestivalPort = "1314"
(31599) WARNING: onLoad (Semstalk.cpp:79): FestivalVoice = "rab_diphone"
(31599) WARNING: onLoad (Semstalk.cpp:80): Caching = true
(31599) WARNING: onLoad (Semstalk.cpp:81): CachePath = "/tmp/"
(31599) WARNING: reloadModuleConfig (SemsConfiguration.cpp:66): no configuration found for module ttsannounce, maybe you want to specify config.ttsannounce in config file.
(31599) WARNING: onLoad (TTSAnnounce.cpp:74): no configuration specified for module ttsannounce.
(31599) WARNING: onLoad (TTSAnnounce.cpp:75): using default configuration:
(31599) WARNING: onLoad (TTSAnnounce.cpp:85): AnnouncePath = "texts/"
(31599) WARNING: onLoad (TTSAnnounce.cpp:86): AnnounceFile = "default.txt"
(31599) WARNING: onLoad (TTSAnnounce.cpp:87): FestivalServer = "localhost"
(31599) WARNING: onLoad (TTSAnnounce.cpp:88): FestivalPort = "1314"
(31599) WARNING: onLoad (TTSAnnounce.cpp:89): FestivalVoice = "rab_diphone"
(31599) WARNING: onLoad (TTSAnnounce.cpp:90): Caching = true
(31599) WARNING: onLoad (TTSAnnounce.cpp:91): CachePath = "/tmp/"
[root@gkproxy01 sems]#
[root@gkproxy01 sems]#
[root@gkproxy01 sems]# (31599) ERROR: reply (AmRequest.cpp:391): AmRequestUAS::reply: 500 command 't_reply' not available
(31599) ERROR: reply (AmRequest.cpp:391): AmRequestUAS::reply: 500 command 't_reply' not available
(31599) ERROR: run (AmSession.cpp:183): 500 could not send response.
(31599) ERROR: reply (AmRequest.cpp:391): AmRequestUAS::reply: 500 command 't_reply' not available
---------VoiceMail ser(port5090) Debug --->>-
[root@gkproxy01 etc]# 4(31339) SIP Request:
4(31339) method: <INVITE>
4(31339) uri: <sip:6605454@call.millicom.com.pe:5090>
4(31339) version: <SIP/2.0>
4(31339) parse_headers: flags=1
4(31339) Found param type 232, <branch> = <z9hG4bK41ed.05c45f56.0>; state=16
4(31339) end of header reached, state=5
4(31339) parse_headers: Via found, flags=1
4(31339) parse_headers: this is the first via
4(31339) After parse_msg...
4(31339) preparing to run routing scripts...
4(31339) DEBUG : is_maxfwd_present: searching for max_forwards header
4(31339) parse_headers: flags=128
4(31339) Found param type 232, <branch> = <z9hG4bKc400663fa4248>; state=16
4(31339) end of header reached, state=5
4(31339) parse_headers: Via found, flags=128
4(31339) parse_headers: this is the second via
4(31339) end of header reached, state=9
4(31339) DEBUG: get_hdr_field: <To> [36]; uri=[sip:6605454@call.millicom.com.pe]
4(31339) DEBUG: to body [<sip:6605454@call.millicom.com.pe>
]
4(31339) get_hdr_field: cseq <CSeq>: <248> <INVITE>
4(31339) DEBUG: get_hdr_body : content_length=182
4(31339) DEBUG: is_maxfwd_present: value = 69
4(31339) check_self - checking if host==us: 20==13 && [call.millicom.com.pe] == [200.110.2.131]
4(31339) check_self - checking if port 5090 matches port 5090
4(31339) check_self - checking if host==us: 20==9 && [call.millicom.com.pe] == [127.0.0.1]
4(31339) check_self - checking if port 5090 matches port 5090
4(31339) DEBUG: t_addifnew: msg id=2 , global msg id=0 , T on entrance=0xffffffff
4(31339) parse_headers: flags=-1
4(31339) found end of header
4(31339) parse_headers: flags=60
4(31339) t_lookup_request: start searching: hash=56852, isACK=0
4(31339) DEBUG: RFC3261 transaction matching failed
4(31339) DEBUG: t_lookup_request: no transaction found
4(31339) DEBUG: add_param: tag=c400663fa4
4(31339) end of header reached, state=29
4(31339) DEBUG: t_check: msg id=2 global id=2 T start=0x422b65a0
4(31339) DEBUG: t_check: T alredy found!
4(31339) parse_headers: flags=-1
4(31339) check_via_address(200.110.2.131, 200.110.2.131, 0)
4(31339) WARNING:vqm_resize: resize(0) called
4(31339) DEBUG: reply sent out. buf=0x80dbf40: SIP/2.0 1..., shmem=0x422b7e10: SIP/2.0 1
4(31339) DEBUG: t_reply: finished
4(31339) **************** vm start - begin ******************
4(31339) parse_headers: flags=-1
4(31339) DEBUG: t_check: msg id=2 global id=2 T start=0x422b65a0
4(31339) DEBUG: t_check: T alredy found!
4(31339) record_route->nameaddr.uri: sip:6605454@200.110.2.131;ftag=c400663fa4;lr=on
4(31339) vm: first proxy has loose routing.
4(31339) vm: calculated route: <sip:6605454@200.110.2.131;ftag=c400663fa4;lr=on>
4(31339) vm: next r-uri: sip:6603000@10.0.0.236
4(31339) parse_headers: flags=-1
4(31339) query="select email_address from subscriber where username='6605454'"
4(31339) vm: write_to_vm_fifo: <0.2
sip_request.voicemail
INVITE
6605454
rrisco(a)millicom.com.pe
call.millicom.com.pe
200.110.2.131
5090
sip:6605454@call.millicom.com.pe:5090
sip:6603000@10.0.0.236
<sip:6603000@call.millicom.com.pe>
<sip:6605454@call.millicom.com.pe>
c4d59b00-c29b-667d-833f-0002a40055b2(a)10.0.0.236
c400663fa4
.
248
56852:790846447
<sip:6605454@200.110.2.131;ftag=c400663fa4;lr=on>
sip:6605454@200.110.2.131;ftag=c400663fa4;lr=on
P-MsgFlags: 0
Min-SE: 1800
Date: Wed, 29 Apr 1970 04:53:24 GMT
.
v=0
o=- 10212804 10212804 IN IP4 10.0.0.236
s=AddPac Gateway SDP
c=IN IP4 10.0.0.236
t=0 0
m=audio 23622 RTP/AVP 4 18 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
>
4(31339) DEBUG: write_to_vm_fifo: write completed
4(31339) DEBUG: add_to_tail_of_timer[0]: 0x422b66dc
4(31339) **************** vm start - end ******************
4(31339) DEBUG:destroy_avp_list: destroing list (nil)
4(31339) receive_msg: cleaning up
9(31349) ERROR: fifo_server: command t_reply is not available
9(31349) ERROR: fifo_server: command must begin with :: ringing
9(31349) ERROR: fifo_server: command must begin with :: 00007B6F0D16CE54
9(31349) ERROR: fifo_server: command must have at least 3 chars
9(31349) INFO: fifo_server: command empty
9(31349) ERROR: fifo_server: command t_reply is not available
9(31349) ERROR: fifo_server: command must have at least 3 chars
9(31349) ERROR: fifo_server: command must begin with :: 00007B6F0D16CE54
9(31349) ERROR: fifo_server: command must begin with :: Content-Type: application/sdp
9(31349) ERROR: fifo_server: command must begin with :: v=0
9(31349) ERROR: fifo_server: command must begin with :: s=session
9(31349) ERROR: fifo_server: command must begin with :: t=0 0
9(31349) ERROR: fifo_server: command must begin with :: a=rtpmap:0 /
9(31349) INFO: fifo_server: command empty
9(31349) ERROR: fifo_server: command t_reply is not available
9(31349) ERROR: fifo_server: command must begin with :: could not send response.
9(31349) ERROR: fifo_server: command must begin with :: 00007B6F0D16CE54
9(31349) ERROR: fifo_server: command must have at least 3 chars
9(31349) INFO: fifo_server: command empty
3(31337) SIP Request:
3(31337) method: <CANCEL>
3(31337) uri: <sip:6605454@call.millicom.com.pe:5090>
3(31337) version: <SIP/2.0>
3(31337) parse_headers: flags=1
.
.
.
"Rafael J. Risco G.V" <rafael_rgv(a)yahoo.com> wrote:
Hello
I´m using ser 0.8.14 with sems (cvs...co -r rel_0_8_12 answer_machine) voicemail its ok but now I would like to use mp3 recording, Do I have to recomplile from cvs HEAD? ...please send me some advice to run last version of sems with ser_0.8.14
thank you
Rafael Risco
Millicom Peru SA
---------------------------------
Do You Yahoo!?
Todo lo que quieres saber de Estados Unidos, América Latina y el resto del Mundo.
Visíta Yahoo! Noticias.
_______________________________________________
Sems mailing list
sems(a)lists.iptel.org
http://lists.iptel.org/cgi-bin/mailman/listinfo/sems
---------------------------------
Do You Yahoo!?
Todo lo que quieres saber de Estados Unidos, América Latina y el resto del Mundo.
Visíta Yahoo! Noticias.
Hi,
I do not pretend that this is the best way, but I use sipsak and
location database and send every minute message to all registered
contacts.
I do not really want to make SER do this stuff because it is a bit out
of scope of SIP PROXY server...
> -----Original Message-----
> From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
On
> Behalf Of John Todd
> Sent: Monday, August 23, 2004 10:30 AM
> To: Jiri Kuthan
> Cc: serusers(a)lists.iptel.org
> Subject: RE: [Serusers] NAT ping and consumer router
>
>
> The problem, as I see it from this discussion, is that some devices
> do not work correctly with the simple UDP packet sent to port 5060 of
> the remote UA, because there is no reply packet which is what keeps
> the NAT mapping of some NAT router/translators. I don't see this as
> a UA problem; if there is no NAT translation, then even the
> best-programmed UA can't receive an inbound INVITE.
>
> The manner in which Asterisk handles this type of keepalive is
> somewhat simple but novel, and may be worth examination. Every X
> seconds, an OPTIONS request is made to the remote UA by the server.
> Even if the UA does not support the OPTIONS query, it typically hands
> back a SIP error, which serves the purpose of keeping the NAT
> translations open. If the device supports OPTIONS, then a "normal"
> SIP reply is sent, also serving the intended purpose.
>
> Perhaps instead of a UDP packet with no content, a SIP OPTIONS
> request could be sent by SER. This could perhaps be an selective
> flag associated with the NAT support in SER, so that either the dummy
> packet or the OPTIONS packet could be transmitted by the module.
>
> There are other solutions here, like reducing the interval of
> REGISTER requests to serve the same purpose of refreshing NAT table
> mappings. However, one could argue that this method has a much
> higher load than an OPTIONS packet, especially when scaling across
> thousands or tens of thousands of clients in an environment where
> external databases (i.e. Radius, SQL, etc) are used for
> authentication lookups.
>
> Note that there have been numerous examples of such poorly-written
> SIP stacks on UA devices that they would crash on an OPTIONS request.
> Their repair is outside the scope of SER or this discussion.
>
> JT
>
>
> At 2:56 AM +0200 on 8/23/04, Jiri Kuthan wrote:
> >I beg to disagree -- we should not create to much workarounds around
> >imperfect clients. In particular, incomplete NAT traversal support
> >is a serious shortcoming in a UA and I would discourage people from
> >using such devices.
> >
> >Other front to attack would be NATs -- there is an effort in IETF
> >focusing on that, but that's obviously an activity which has no
> >impact on currently installed base.
> >
> >-jiri
> >
> >At 01:57 AM 8/23/2004, Richard wrote:
> >>Hi Jesus,
> >>
> >>Changing UA is not always a viable solution due to pricing and other
> >>technical issues. Every UA has something broken in its
implementation
> and it
> >>would be very costly to change it because one thing (in this case,
NAT)
> is
> >>broken.
> >>
> >>Thanks,
> >>Richard
> >>
> >>
> >>-----Original Message-----
> >>From: Jesus Rodriguez [mailto:jesusr@voztele.com]
> >>Sent: Sunday, August 22, 2004 8:59 AM
> >>To: Richard
> >>Cc: serusers(a)lists.iptel.org
> >>Subject: Re: [Serusers] NAT ping and consumer router
> >>
> >>
> >>Use an UA that supports it (Sipura or Cisco for example).
> >>
> >>Saludos
> >>JesusR.
> > >
> >>-------------------------------
> >>Jesus Rodriguez
> >>VozTelecom Sistemas, S.L.
> >>jesusr(a)voztele.com
> >>http://www.voztele.com
> >>Tel. 902360305
> > >-------------------------------
> >--
> >Jiri Kuthan http://iptel.org/~jiri/
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
Hello,
I've just found out about SER and figured I'd try it out.
Our billing system is based of radius files. So I can use Radius with
SER for billing/accounting, but I'm confused if I also need to add MySQL
support? Is MySQL simply used for configuration storage?
Also, I'm installing via RPM and it complains about
"libradiusclient.so.0 is needed by ser-radius-0.8.12-0" so as per the
sites instructions I went to http://www.mcs.de/~lf/radius which doesn't
exist anymore. I'm using Fedora Core-1 so should I attempt to find an
rpm or should I scrap all the rpms and just install via source?
Thanks,
- Darren
There seems to be an error in the documentation of
ser "voicemail" module. In section 1.3.1 @
http://www.iptel.org/ser/doc/modules/html/vm.html#AEN61
Example 1-1 should read
modparam("voicemail", "db_url", "sql://username:password@localhost/ser")
instead of
modparam("vm", "db_url", "sql://username:password@localhost/ser")
This also applies to examples 1-2 through 1-5.
Cheers,
Marcel
Hi,
i copied my alias-table to voicemail and build a webfrontend
to file in the voicemail accounts.
so if user is offline or busy i use
if(!lookup("location")) {
if(lookup("voicemail")) {
# send to voicemail
}
}
now i have to restart the ser-server to get the
new voicemailaccounts into SER-MEM. Is there
a possibility to tell ser to sync db with mem?
Something like sending an USR Signal?
Thx and greets
Markus
Hi all,
can anybody give me a hint on the cause of the mediaproxy error:
error: uncaptured python exception, closing channel
<rtphandler.RTPStream connected 192.168.20.2:35001 at 0x405576cc>
(socket.error:(101, 'Network is unreachable')
[/usr/lib/python2.3/asyncore.py|readwrite|86]
[/usr/lib/python2.3/asyncore.py|handle_read_event|390]
[/usr/local/mediaproxy/modules/rtphandler.py|handle_read|715]
[/usr/local/mediaproxy/modules/rtphandler.py|sendto|628])
What network is unreachable?
192.168.20.2:35001?
Thanks in advance.
Franz
Hi all,
Im using linux9.0. I installed asterisk in linux. The machine has two interface one is public ip(203.xxx.xxx.xxx), another one is local ip(10.1.1.185). I install ser in linux machine. I want to configure ser for any incoming call from outside to another linux machine which is running in [ (10.1.1.180) ]. any one can help me regards this problem.
thanks in advance
Regards
Murali
Hello Everyone,
I am a newbie SER user.
I am working on a chat client application that supports VoIP and text
chat (both single and multiparty IM). I am using Microsoft's RTCDLL
with SER.
My question is, is there an issue with regards to using RTCDLL with
SER and using TCP as the transfer protocol? I asked this because if i
specify UDP as the transfer protocol im using for both the proxy and
registrar, text chat works just fine. If i change either of the
settings to TCP, i get the 481 error (Call Leg transaction does not
exist) whenever the contacted UA tries to send back message (reply) to
the caller/initiator of the text chat.
The following are the infos relevant to my system setup
OS version is : Linux Redhat 9
Version: ser 0.8.12 ( i 386/linux)
flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $ld:main.c.v 1.168 2003/10/12 15:09:08 andrei Exp $
main.c compiled on 13:09:22 Nov 21 2003 with gcc 2.95
ser config file
#
# $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 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=3
fork=yes
log_stderror=no
*/
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"
sip_warning=yes
server_signature=yes
uid="adrian"
gid="adrian"
listen=192.168.10.38
listen=127.0.0.1
# hostname matching an alias will satisfy the condition uri==myself".
alias=gandalf.hq1.astra.ph
#alias=hq1.astra.ph
alias=192.168.10.38
# ------------------ 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/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
loadmodule "/usr/lib/ser/modules/acc.so"
loadmodule "/usr/lib/ser/modules/exec.so"
loadmodule "/usr/lib/ser/modules/group.so"
loadmodule "/usr/lib/ser/modules/print.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/uri.so"
loadmodule "/usr/lib/ser/modules/pa.so"
# ----------------- setting module-specific parameters ---------------
# ------------- tm parameters
modparam("tm", "fr_timer", 10)
modparam("tm", "fr_inv_timer", 20)
# ------------- rr parameters
# set ";lr" tag to “;lr=trueâ€<9d>
modparam("rr", "enable_full_lr", 1)
# ------------- accounting parameters
modparam("acc", "log_missed_flag", 3)
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1)
# ------------- usrloc parameters
# 2 enables write-back to persistent mysql storage for speed
# disable=0, write-through=1
modparam("usrloc", "db_mode", 0)
# minimize write back window - default is 60 seconds
modparam("usrloc", "timer_interval", 10)
# database location
modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser")
# ------------- auth parameters
# database location
modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser")
# allows clear text passwords in the mysql database
modparam("auth_db", "calculate_ha1",1)# yes)
# name of password column in mysql database
modparam("auth_db", "password_column", "password")
# pa initialization
#modparam("pa", "default_expires", 3600)
# ------------------------- 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 (len_gt( 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;
};
lookup("aliases");
# 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("gandalf.hq1.astra.ph",
"subscriber")) {
www_challenge("gandalf.hq1.astra.ph", "1");
break;
};
# only registered users are allowed
# if (!is_user("replicator") & !check_to()) {
# log(1, "LOG: unregistered user registration
# attempt\n");
# sl_send_reply("403", "Only registered users are
# allowed");
# break;
# };
# it is an authenticated request, update Contact
# database now
# if (!save("location")) {
# sl_reply_error();
# };
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
#if(t_newtran()){
# if (method=="SUBSCRIBE" || method=="PUBLISH"){
# handle_subscription("registrar");
# 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();
};
}
end of config file
As i said before, if i use UDP as my transport protocol, everything
works fine....the call initiator can send messages and the receiver
can reply back.
But when i use TCP, even if the call initiator can still send messages
like before without any problem, the other client UA (receiver) cannot
send messages back anymore....here is the sample captured SIP Messages
when the caller sends text chat messages.
T 192.168.10.42:1283 -> 192.168.10.38:5060 [AP]
MESSAGE
sip:rose@192.168.10.38;transport=tcp;ftag=c3366ad7dc9c41d799d979d8aa40b6ef;lr=on
SIP/2.0..Via: SIP/2.0/TCP 192.1
68.10.42:14457..Max-Forwards: 70..From: "jon"
<sip:jon@gandalf.hq1.astra.ph>;tag=c3366ad7dc9c41d799d979d8aa40b6ef;epid=b
f0baf3a75..To:
<sip:rose@gandalf.hq1.astra.ph>;tag=0ce6d678daf24305a094f59a177bdac7..Call-ID:
e4256863ae5e4c82b2c1138152
e1221e@192.168.10.42..CSeq: 3 MESSAGE..Route:
<sip:rose@gandalf.hq1.astra.ph:11818;maddr=192.168.10.27;transport=tcp>..C
ontact:
<sip:jon@gandalf.hq1.astra.ph:14457;maddr=192.168.10.42;transport=tcp>..User-Agent:
RTC/1.2..Content-Type: text/
enriched; charset=UTF-8..Content-Length:
230....{\rtf1\ansi\ansicpg1252\deff0\deflang1033{\fonttbl{\f0\fnil\fcharset0
Ar
ial;}{\f1\fnil\fcharset0 Tahoma;}}..{\colortbl
;\red0\green0\blue255;}..\viewkind4\uc1\pard\cf1\b\fs18 jon : \cf0\b0\fs1
6 hello po\par..\pard\f1\fs17\par..}..
##########
T 192.168.10.38:33101 -> 192.168.10.27:11818 [AP]
MESSAGE sip:192.168.10.27:11818;transport=tcp SIP/2.0..Record-Route:
<sip:rose@192.168.10.38;transport=tcp;ftag=c3366ad7
dc9c41d799d979d8aa40b6ef;lr=on>..Via: SIP/2.0/TCP
192.168.10.38;branch=z9hG4bKd26d.d1049a22.4;i=01..Via: SIP/2.0/TCP 192
.168.10.42:14457..Max-Forwards: 69..From: "jon"
<sip:jon@gandalf.hq1.astra.ph>;tag=c3366ad7dc9c41d799d979d8aa40b6ef;epid
=bf0baf3a75..To:
<sip:rose@gandalf.hq1.astra.ph>;tag=0ce6d678daf24305a094f59a177bdac7..Call-ID:
e4256863ae5e4c82b2c11381
52e1221e@192.168.10.42..CSeq: 3 MESSAGE..Contact:
<sip:jon@gandalf.hq1.astra.ph:14457;maddr=192.168.10.42;transport=tcp>
..User-Agent: RTC/1.2..Content-Type: text/enriched;
charset=UTF-8..Content-Length: 230....{\rtf1\ansi\ansicpg1252\deff0\
deflang1033{\fonttbl{\f0\fnil\fcharset0 Arial;}{\f1\fnil\fcharset0
Tahoma;}}..{\colortbl ;\red0\green0\blue255;}..\viewk
ind4\uc1\pard\cf1\b\fs18 jon : \cf0\b0\fs16 hello
po\par..\pard\f1\fs17\par..}..
#
T 192.168.10.27:11818 -> 192.168.10.38:33101 [AP]
SIP/2.0 200 OK..Via: SIP/2.0/TCP
192.168.10.38;branch=z9hG4bKd26d.d1049a22.4;i=01..Via: SIP/2.0/TCP
192.168.10.42:14457.
.From: "jon"
<sip:jon@gandalf.hq1.astra.ph>;tag=c3366ad7dc9c41d799d979d8aa40b6ef;epid=bf0baf3a75..To:
<sip:rose@gandalf.
hq1.astra.ph>;tag=0ce6d678daf24305a094f59a177bdac7..Call-ID:
e4256863ae5e4c82b2c1138152e1221e@192.168.10.42..CSeq: 3 MES
SAGE..Contact:
<sip:rose@gandalf.hq1.astra.ph:11818;maddr=192.168.10.27;transport=tcp>..User-Agent:
RTC/1.2..Content-Len
gth: 0....
##
T 192.168.10.38:5060 -> 192.168.10.42:1283 [AP]
SIP/2.0 200 OK..Via: SIP/2.0/TCP 192.168.10.42:14457..From: "jon"
<sip:jon@gandalf.hq1.astra.ph>;tag=c3366ad7dc9c41d799d
979d8aa40b6ef;epid=bf0baf3a75..To:
<sip:rose@gandalf.hq1.astra.ph>;tag=0ce6d678daf24305a094f59a177bdac7..Call-ID:
e42568
63ae5e4c82b2c1138152e1221e@192.168.10.42..CSeq: 3 MESSAGE..Contact:
<sip:rose@gandalf.hq1.astra.ph:11818;maddr=192.168.1
0.27;transport=tcp>..User-Agent: RTC/1.2..Content-Length: 0....
______________________________________________________________
And here is the sample SIP captured message if the receiver tries to
reply back to the caller....the one with the 481 error
T 192.168.10.27:1077 -> 192.168.10.38:5060 [AP]
MESSAGE
sip:rose@192.168.10.38;transport=tcp;ftag=c3366ad7dc9c41d799d979d8aa40b6ef;lr=on
SIP/2.0..Via: SIP/2.0/TCP 192.1
68.10.27:11818..Max-Forwards: 70..From:
<sip:rose@gandalf.hq1.astra.ph>;tag=0ce6d678daf24305a094f59a177bdac7..To:
"jon"
<sip:jon@gandalf.hq1.astra.ph>;tag=c3366ad7dc9c41d799d979d8aa40b6ef;epid=bf0baf3a75..Call-ID:
e4256863ae5e4c82b2c1138152
e1221e@192.168.10.42..CSeq: 1 MESSAGE..Route:
<sip:jon@gandalf.hq1.astra.ph:14457;maddr=192.168.10.42;transport=tcp>..Co
ntact:
<sip:rose@gandalf.hq1.astra.ph:11818;maddr=192.168.10.27;transport=tcp>..User-Agent:
RTC/1.2..Content-Type: text/
enriched; charset=UTF-8..Content-Length:
249....{\rtf1\ansi\ansicpg1252\deff0\deflang13321{\fonttbl{\f0\fnil\fcharset0
A
rial;}{\f1\fnil\fcharset0 Tahoma;}}..{\colortbl
;\red0\green0\blue255;}..\viewkind4\uc1\pard\cf1\b\fs18 rose : \cf0\b0\f
s16 r\lang1033 eply\par..\pard\lang13321\f1\fs17\par..}..
##########
T 192.168.10.38:33101 -> 192.168.10.27:11818 [AP]
MESSAGE sip:192.168.10.27:11818;transport=tcp SIP/2.0..Record-Route:
<sip:rose@192.168.10.38;transport=tcp;ftag=0ce6d678
daf24305a094f59a177bdac7;lr=on>..Via: SIP/2.0/TCP
192.168.10.38;branch=z9hG4bKf26d.b82505f.4;i=11..Via: SIP/2.0/TCP 192.
168.10.27:11818..Max-Forwards: 69..From:
<sip:rose@gandalf.hq1.astra.ph>;tag=0ce6d678daf24305a094f59a177bdac7..To:
"jon"
<sip:jon@gandalf.hq1.astra.ph>;tag=c3366ad7dc9c41d799d979d8aa40b6ef;epid=bf0baf3a75..Call-ID:
e4256863ae5e4c82b2c113815
2e1221e@192.168.10.42..CSeq: 1 MESSAGE..Contact:
<sip:rose@gandalf.hq1.astra.ph:11818;maddr=192.168.10.27;transport=tcp>
..User-Agent: RTC/1.2..Content-Type: text/enriched;
charset=UTF-8..Content-Length: 249....{\rtf1\ansi\ansicpg1252\deff0\
deflang13321{\fonttbl{\f0\fnil\fcharset0 Arial;}{\f1\fnil\fcharset0
Tahoma;}}..{\colortbl ;\red0\green0\blue255;}..\view
kind4\uc1\pard\cf1\b\fs18 rose : \cf0\b0\fs16 r\lang1033
eply\par..\pard\lang13321\f1\fs17\par..}..
##
T 192.168.10.38:5060 -> 192.168.10.27:1077 [AP]
SIP/2.0 408 Request Timeout..Via: SIP/2.0/TCP 192.168.10.27:11818..From:
<sip:rose@gandalf.hq1.astra.ph>;tag=0ce6d678daf
24305a094f59a177bdac7..To: "jon"
<sip:jon@gandalf.hq1.astra.ph>;tag=c3366ad7dc9c41d799d979d8aa40b6ef;epid=bf0baf3a75..Ca
ll-ID: e4256863ae5e4c82b2c1138152e1221e@192.168.10.42..CSeq: 1
MESSAGE..Server: Sip EXpress router (0.8.11 (i386/linux))
..Content-Length: 0..Warning: 392 192.168.10.38:5060 "Noisy feedback tells:
pid=5107 req_src_ip=192.168.10.27 req_src_p
ort=1077
in_uri=sip:rose@192.168.10.38;transport=tcp;ftag=c3366ad7dc9c41d799d979d8aa40b6ef;lr=on
out_uri=sip:192.168.10.
27:9287;transport=tcp via_cnt==0"....
##
T 192.168.10.27:11818 -> 192.168.10.38:33101 [AP]
SIP/2.0 481 Call Leg/Transaction Does Not Exist..Via: SIP/2.0/TCP
192.168.10.38;branch=z9hG4bKf26d.b82505f.4;i=11..Via:
SIP/2.0/TCP 192.168.10.27:11818..From:
<sip:rose@gandalf.hq1.astra.ph>;tag=0ce6d678daf24305a094f59a177bdac7..To:
"jon" <
sip:jon@gandalf.hq1.astra.ph>;tag=c3366ad7dc9c41d799d979d8aa40b6ef;epid=bf0baf3a75..Call-ID:
e4256863ae5e4c82b2c1138152e
1221e@192.168.10.42..CSeq: 1 MESSAGE..User-Agent: RTC/1.2..Content-Length:
0....
I have no clue as to why it behaves this way while it does not so
using UDP. I don't know where the actual problem lies although i am
biased that more likely, it is microsoft's fault. ^_^
Please enlighten me, anything would greatly help.
Many thanks in advance....
--
Jonathan
hello richard,
what registration do you mean ? is it the useragent registration to the server
if it is so . then the answer is user agent ( xlite softphone ) is registring well .
if you mean that cpl script registration to the server then i have seen in debug
mode that the server is accepting the script file and checking whether the
file is in database or not if not then it is inserting that
when i use the following instructions i could not able to see the messeage "some one doing cpl registration ......" when the user is logging in where the cpl script has been
inserted for him.
if (search("Content-type:.*application/cpl\+xml")) {
xlog("L_INFO", "Someone doing CPL REGISTER from %is\n");
cpl_process_register();
break;
};
so where should i check now
with regards
serdiehard
Richard <richard(a)o-matrix.org> wrote:I'd suspect that the registration message didn't make it to the server. An
ethereal capture might help to find out the problem.
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of ser die
Sent: Saturday, August 21, 2004 1:36 AM
To: Richard
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] fifo + cpl , contenttype mime 0,0 found error
thanks richard,
i have been held up with other problems thats why i
could not able to do this in this 4 days time
any this fifo command is successfully loaded into the
server
its loading the cpl on to server and then there
inserted to the database
i have just kept these instructions in ser.cfg
if(method=="REGISTER")
{
cpl_process_register();
if(!www_authorize("xx.xx.xx.in", "subscriber")) {
www_challenge("xx.xx.xx.in", "0");
break;
};
save("location");
break;
};
but still the problem persists
the debug message shows
8(1925) DEBUG:parse_content_type_hdr: missing
Content-Typeheader
8(1925) DEBUG:cpl_process_register: Content-Type mime
found 0, 0
the xml file looks like this
1.0//EN' 'cpl.dtd'>
i couldnot get why the contenttype is not being
identifed
your help will be highly appreciated
with regards
rama kanth
--- Richard wrote:
> Hi,
>
>
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/cpl-c/cpl_l
oader.c?rev=1.9&content-type=text/vnd.viewcvs-markup
>
> In the command, it needs 4 lines,
> :LOAD_CPL:
> username
> cpl_filename
>
>
> Also the filename needs to be absolute path. I think
> that might be your problem.
>
> Richard
>
>
> --- ser die wrote:
>
> > hello friends ,
> >
> > when i try this command in the command prompt
> >
> > serctl fifo LOAD_CPL ::214103@xxx.xxx.xxx.in
> > ::cpl214103.xml
> >
> > in debug i have seen this
> > ***********************************************
> > 9(8689) DEBUG:cpl_load:
> > user@host=::214103@xxx.xxx.xxx.in
> > 9(8689) ERROR:cpl-c:cpl_load: unable to read
> > cpl_file
> > name from FIFO command
> > 9(8689) ERROR: fifo_server: command (LOAD_CPL)
> > processing failed
> >
> ***************************************************
> >
> > so can any body say the correct way
> >
> > with regards
> > serdiehard
> >
> >
> >
> >
> >
> > __________________________________
> > Do you Yahoo!?
> > New and Improved Yahoo! Mail - 100MB free storage!
> > http://promotions.yahoo.com/new_mail
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
>
>
> __________________________________
> Do you Yahoo!?
> Yahoo! Mail - 50x more storage than other providers!
> http://promotions.yahoo.com/new_mail
>
__________________________________
Do you Yahoo!?
New and Improved Yahoo! Mail - Send 10MB messages!
http://promotions.yahoo.com/new_mail
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
---------------------------------
Do you Yahoo!?
Win 1 of 4,000 free domain names from Yahoo! Enter now.
