Hi!
I have problems receiving calls when I'm registered to iptel.org using
Windows Messenger 4.7 (which usually workes fine).
Every incoming INVITE will be answered with a 400 Bad Request by
Messenger. It is independent from the caller's user agent (tested with
messenger and xlite). Furthermore, if the caller dials to me directly
without using iptel's proxy it also works. Therefore, I think the
problem is related to iptel's proxy.
Below is the dump of the incoming INVITE which is a Bad Request (says
Messenger). I couldn't find any problems, accept that the
Proxy-Authorization: header is still present.
regards,
klaus
INVITE sip:128.131.80.218:9227 SIP/2.0
Max-Forwards: 10
Record-Route:
<sip:klaus3000@195.37.77.101;ftag=db80b783-e78f-44e3-999d-392dc5604172;lr>
Via: SIP/2.0/UDP 195.37.77.101;branch=z9hG4bK5b8a.4006c915.1
Via: SIP/2.0/UDP 128.131.80.129:10585
From: "kurth" <sip:kurth@iptel.org>;tag=db80b783-e78f-44e3-999d-392dc5604172
To: <sip:klaus3000@iptel.org>
Call-ID: 9c446f6d-49a7-4e64-8271-b67640df2a72(a)128.131.80.129
CSeq: 2 INVITE
Contact: <sip:128.131.80.129:10585>
User-Agent: Windows RTC/1.0
Proxy-Authorization: Digest username="kurth", realm="iptel.org",
algorithm="md5", uri="sip:klaus3000@iptel.org",
nonce="409a60508b7d5cdbd4c83d73d676693c35750df5",
response="1ed41c442263572c152711fd1670a341"
Content-Type: application/sdp
Content-Length: 454
P-hint: USRLOC
v=0
o=pc129 0 0 IN IP4 128.131.80.129
s=session
c=IN IP4 128.131.80.129
b=CT:1000
t=0 0
m=audio 40746 RTP/AVP 97 111 112 6 0 8 4 5 3 101
a=rtpmap:97 red/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:6 DVI4/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
I've started mediaproxy as writen in Readme, but when i try to start ser I
can see this.
any idea?
0(0) DEBUG: FIFO created @ /tmp/ser_fifo
0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432
0(0) init_unixsock_socket: No unix domain socket will be opened
1(0) DEBUG: init_mod_child (-1): tm
1(0) DEBUG: callid: '5d3ffff7-0(a)127.0.0.1'
1(0) DEBUG: init_mod_child (-1): usrloc
1(0) get_connection(): Connection not found in the pool
1(0) DEBUG: init_mod_child (-1): auth_db
1(0) get_connection(): Connection not found in the pool
1(0) DEBUG: init_mod_child (-1): domain
1(0) DEBUG: init_mod_child (-1): nathelper
1(0) ERROR: send_rtpp_command: can't connect to RTP proxy
1(0) WARNING: rtpp_test: can't get version of the RTP proxy
1(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
2(0) INFO: fifo process starting: 17183
2(0) DEBUG: init_mod_child (-2): tm
2(0) DEBUG: callid: '5d3ffff7-0(a)127.0.0.1'
2(0) DEBUG: init_mod_child (-2): usrloc
2(0) get_connection(): Connection not found in the pool
2(0) DEBUG: init_mod_child (-2): auth_db
2(0) get_connection(): Connection not found in the pool
2(0) DEBUG: init_mod_child (-2): domain
2(0) DEBUG: init_mod_child (-2): nathelper
2(0) ERROR: send_rtpp_command: can't connect to RTP proxy
2(0) WARNING: rtpp_test: can't get version of the RTP proxy
2(0) WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo...
2(0) DEBUG: register_fifo_cmd: new command (print) registered
2(0) DEBUG: register_fifo_cmd: new command (uptime) registered
2(0) DEBUG: register_fifo_cmd: new command (version) registered
2(0) DEBUG: register_fifo_cmd: new command (pwd) registered
2(0) DEBUG: register_fifo_cmd: new command (arg) registered
2(0) DEBUG: register_fifo_cmd: new command (which) registered
2(0) DEBUG: register_fifo_cmd: new command (ps) registered
2(0) DEBUG: register_fifo_cmd: new command (kill) registered
2(0) WARNING: no fifo_db_url given - fifo DB commands disabled!
0(17177) DEBUG: init_mod_child (1): tm
0(17177) DEBUG: callid: '5d3ffff7-17177(a)127.0.0.1'
0(17177) DEBUG: init_mod_child (1): usrloc
0(17177) get_connection(): Connection not found in the pool
0(17177) DEBUG: init_mod_child (1): auth_db
0(17177) get_connection(): Connection not found in the pool
0(17177) DEBUG: init_mod_child (1): domain
0(17177) get_connection(): Connection found in the pool
0(17177) DEBUG: init_mod_child (1): nathelper
0(17177) ERROR: send_rtpp_command: can't connect to RTP proxy
0(17177) WARNING: rtpp_test: can't get version of the RTP proxy
0(17177) WARNING: rtpp_test: support for RTP proxyhas been disabled
temporarily
0(17177) NOTICE:init_avp_child: no avp_db_url specified -> feature disabled
Hi,
here is my python script in order to send emails.
For example i call my mailbox to listen messages if I'm offline i wish
to received a voicemail.
I don't know how to get the new wav recorded messages (path) from the
the mailbox.
I'm not a devolopper so I need help.
I don't test it because of I have to update Mysql !!
I do hope somebody could help me (and ser users) about this feature.
Regards
Harry
After replace my nathelper by the mediaproxy.so module and implement de mediaproxy service my SER crash after receiving a REGISTER message and do the fix_contact() command. The problem like to be the BUG: tcp_main_loop but i dont see what is the reason.
Any can help me with this ?
My mediaproxy.so module be compiled with the src files of the SER 0.8.12 (rpm). I have tried to compile with the last CVS version but the mysql.so module dont compile.
2(28514) SIP Request:
2(28514) method: <REGISTER>
2(28514) uri: <sip:arcotel.net>
2(28514) version: <SIP/2.0>
2(28514) parse_headers: flags=1
2(28514) end of header reached, state=5
2(28514) parse_headers: Via found, flags=1
2(28514) parse_headers: this is the first via
2(28514) After parse_msg...
2(28514) preparing to run routing scripts...
2(28514) parse_headers: flags=8
2(28514) DEBUG: add_param: tag=2207940617
2(28514) end of header reached, state=29
2(28514) DEBUG : is_maxfwd_present: searching for max_forwards header
2(28514) parse_headers: flags=128
2(28514) end of header reached, state=9
2(28514) DEBUG: get_hdr_field: <To> [39]; uri=[sip:ecolombo@arcotel.net]
2(28514) DEBUG: to body [1152321660 <sip:ecolombo@arcotel.net>]
2(28514) get_hdr_field: cseq <CSeq>: <1> <REGISTER>
2(28514) DEBUG: get_hdr_body : content_length=0
2(28514) found end of header
2(28514) DEBUG: is_maxfwd_present: max_forwards header not found!
2(28514) ROUTE_0: Fixing Contact 2(28514) parse_headers: flags=64
2(28514) parse_headers: flags=33554432
11(28562) BUG: tcp_main_loop: dead child 2
0(28511) child process 28514 exited by a signal 11
0(28511) core was not generated
0(28511) INFO: terminating due to SIGCHLD
1(28513) INFO: signal 15 received
Thanks
Ezequiel Colombo
ARCOTEL S.A.
Av. Córdoba 3016
(C1187AAR) C.A.B.A.
Cel: 15 44063002
Tel/Fax: (5411) 49643002/3332
hello friends,
i have the latest cvs reliable ser
and using the rtpproxy of v 1.4 2003/08/05
when iam running rtpproxy with ser
it s running and showing the addr1,addr2 etc
but when i run netstat its not showing
/var/run/rtpproxy.sock
so whats the problem
with regards
rama kanth
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Hi.
i'm Andrea, i have a problem with sems. I have the same configuration, ser +
sems but i don't receive the e-mail when a user is offline. The wav file are
correctly stored in /tmp but sems don't sent it.
How can i do?
Tank's.
Andrea.
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi, all. I'm trying to use IVR with ser...
but i have a little problem with:
on ivr.py
get_name(ivr.getTo()) return "None" ...
my ivr.py is default conf , i have only changed the message && wave paths...
here is post of the error:
- --- sems debug ---
Try playAndDetect detected
8601 <- this is get_name(ivr.getFrom()) and return value....
None <- This is get_name(ivr.getTo()) and return every time the same value.
play
record
Traceback (most recent call last):
File "/etc/ivr/ivr.py", line 56, in ?
if os.path.isdir(messages_path + callee) != True:
TypeError: cannot concatenate 'str' and 'NoneType' objects
(27083) WARNING: sig_usr (sems.cpp:72): signal 17 received
- ---
so as we can see here... the script cannot create dir /tmp/ivr/$calee, because
calee=None ... :-( and all sems system crash...
is anyone have same problems? can you give me some solutions?
p.s
Excuse my bad english... :)
- --
Regards, Kamen Sharlandjiev
System Administrator
NetBG Communication
Tel: +359 2 962 43 52
+359 2 962 53 93
- --
Public GPG key at: http://pgp.mit.edu
pub 1024D/C6347D3D 2003-03-19 Kamen Sharlandjiev (Comment) <stone(a)netbg.com>
Key fingerprint = D6AA 55BD F0CB FC6B 8C22 A13B 91D1 55C3 C634 7D3D
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQFAtbMnkdFVw8Y0fT0RAvKzAJwNex+o1NkKed5xBo+NlQzPZRK6CgCfUET2
ntUWJ/JK4+HxrdDIl7jXMww=
=n0u+
-----END PGP SIGNATURE-----
Hi all,
Just a simple question.
How to create a mailbox for each user (there is ivr.py script) with
password and/or mail forwarding:
For example I dial my mailbox 1000 to listen my messages.
Now I'm offline somebody call me he's forwarded to my mailbox and record
a message.
Is it possible to send that message to my email box with attachement.
Regards
harry
Hello I have a question... as I am able redireccionar a call sip to a team h323 and vice versa??
Client
|
|
SIP Server <----> H323
|
|
client
Saludos!
Sergio Gerardo Festa
Dto. Técnica
Tel. 011-5219-0654 // 4334-1014
24HS 011-15-5183-9203
Hi,
I'm sorry that I run into so many problems with cpl
scripts.
I want to send a call to multiple phones. For example,
if there is a call to 222, it lookups the location
table and call all registered 222 phones. In
addational, call some other numbers 223 and 224. I
don't know how to do it in cpled. The closest I can
get is this script for 222.
Can someone show me what's the proper way of doing
this?
Thanks,
Richard
========================
<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE cpl PUBLIC '-//IETF//DTD RFCxxxx CPL
1.0//EN' 'cpl.dtd'>
<cpl>
<incoming>
<location url="sip:223@example.org">
<proxy>
<default>
<lookup source="registration" />
</default>
</proxy>
</location>
</incoming>
</cpl>
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