Hi,
I need to know of any SIP cmpatible free video client for windows ...
Or if I have to use NetMeeting and put an H323/SIP gateway in the
middle of client and proxy ...
What I need is something like X-Lite soft phone ... with video suport
... For Windows ... and free ...
Thanks,
R.
Hi all,
Can anyone suggest a better logic for me. What I want is to account "Busy"
or "DND" as missed calls and forward the request to voicemail.
In my config (attached), I have a t_on_failure block which will relay the
request to voicemail system. The trouble is, once the request is relayed,
SER will not treat this as a missed call. Instead, SER will treat it as a
successful call with the duration of the voicemail. It will also account two
BYEs.
I have tried to send a 'redirect to voicemail' reply to UA in the failure
block. But than, the account reason becomes a "300" response, not "4xx"
response.
failure_route[2] {
xlog(L_NOTICE", "Callee busy or DND, forwarding to voicemail\n");
revert_uri();
lookup("aliases");
subst_uri('/(sip:)(.*)@(.*)/<\1voicemail-\2@\3>/i');
t_reply("300", "Redirect to voicemail");
}
Unless I can use acc_db_request() to do manual accounting or
sl_send_reply("300", ...) to send the redirect in the failure block, I don't
know to change the logic to get what I want. Any idea?
Zeus Ng
Hi I'm running trying out latest CVS checked out today with the following
configuration and am having the following error:
May 6 00:56:54 hostname /usr/local/sbin/ser[30030]: ERROR:acc:mod_init:
bind_db failed...did you load a database module?
May 6 00:56:54 hostname /usr/local/sbin/ser[30030]: init_mod(): Error
while initializing module acc
However I can connect to mysql just fine, from the command line. In
addition I just ran ser_mysql.sh create. Anyone know I
am getting this error? Attached is my ser.cfg with no modifications ( this
is stock from ser.cfg.m4)
regards,
Andy
Hi,
I've been experimenting with the following setup:
ipv6 ua --- ser (v6) --- msp gateway --- ser (v4) --- cisco 3640 pstn gateway
--- cisco ip-phone
Everything works fine with the IP phone but routing the call
to the PSTN gateway causes it to return a 400 Malformed/Missing Request
in response to the packet forwarded to it from ser(v4)
If i use the ip-phone to make a call via the pstn gateway
(i.e. involving only the V4 SER) everything is fine.
This makes me believe that the most likely cause of the crash
is that the PSTN gateway can not understand the IPv6 address
in the Via: or the Record-Route: header fields
If that is the case, where do people think would
be the best place to remove the 'offending' fields
and add them back when the reply comes back from the
PSTN gateway ? The SERv4 server or the MSP gateway ?
The last packet sent to the PSTN gateway is:
Session Initiation Protocol
Request-Line: INVITE sip:3007@128.16.67.2:5060 SIP/2.0
Message Header
Record-Route: <sip:3007@128.16.234.7;ftag=49F56B1E;lr=on>
Max-Forwards: 9
Record-Route: <sip:[2001:630:13:101:2E0:18FF:FE3E:ECB2];ftag=49F56B1E;l
r
=on>
Via: SIP/2.0/UDP 128.16.234.7;branch=z9hG4bK86ba.d7fafbd5.0
Via: SIP/2.0/UDP 128.16.67.73:5062
Via: SIP/2.0/UDP [2001:630:13:101:2e0:18ff:fe3e:ecb2];branch=0
Via: SIP/2.0/UDP [2001:630:13:101:204:23ff:fe0b:dc58]
CSeq: 6566 INVITE
To: <sip:3007@128.16.234.7>
Content-Type: application/sdp
From: "Manish Lad" <sip:1006@frostie.cs.ucl.ac.uk>;tag=49F56B1E
Call-ID: 1658003513
Subject: sip:1006@frostie.cs.ucl.ac.uk
Content-Length: 185
User-Agent: KPhone/3.11
Contact: "NAT Proxy" <sip:proxy@128.16.67.73:5062;realuri=0-ralOcJk1sRAM
6AlMTdrMoAvMTMvMTAmO6IlNDdnMrZbObZaMGIvZGMqODip-bFj-sBk-cQz9WRl>
Session Description Protocol
Session Description, version (v): 0
Owner/Creator, Session Id (o): username 0 0 IN IP4 128.16.67.73
Session Name (s): The Funky Flow
Connection Information (c): IN IP4 128.16.67.73
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 32974 RTP/AVP 0 97 3
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:97 iLBC/8000
The reply from the gateway:
Session Initiation Protocol
Status-Line: SIP/2.0 400 Bad Request - 'Malformed/Missing Record Route'
Message Header
Via: SIP/2.0/UDP 128.16.234.7;branch=z9hG4bK86ba.d7fafbd5.0,SIP/2.0/UDP
128.16.67.73:5062,SIP/2.0/UDP [2001:630:13:101:2e0:18ff:fe3e:ecb2];branch=0,SIP
/
2.0/UDP [2001:630:13:101:204:23ff:fe0b:dc58]
From: "Manish Lad" <sip:1006@frostie.cs.ucl.ac.uk>;tag=49F56B1E
To: <sip:3007@128.16.234.7>;tag=88098FAC-15AD
Call-ID: 1658003513
CSeq: 6566 INVITE
Content-Length: 0
Regards,
Lambros
Hi guys,
I was create a script to route some kinds of routes calls to PSTN ( like
that ). So after 59 sec. it is droped.
Someone can help me ?
Thank you all for attention and kindness in advance.
Jadylson Bomfim
if (uri=~"sip:0[0-9]*@.*") {
route(6);
break;
};
route[6] {
setflag(1);
if (uri=~"^sip:0[1-9]*@") {
strip(1);
prefix("01155");
rewritehostport("ip address:5060");
acc_rad_request("Saindo Para PSTN no Brasil");
if (method=="INVITE" && search("User-Agent: snom")) {
replace("100rel, ", "");
};
append_hf("P-hint: GATEWAY\r\n");
# use UDP to guarantee well-known sender port (TCP ephemeral)
t_relay_to_udp("ip address","5060");
} else {
if (uri=~"^sip:00[0-9][0-9]*@") { # ... forward to gateways then;
#Cheque se inicia com 00
if (uri=~"^sip:00") {
# strip the leading "1"
strip(2);
prefix("011");
# if you have passed through all the checks, let your call go to GW!
rewritehostport("ip address:5060");
# snom conditioner
if (method=="INVITE" && search("User-Agent: snom")) {
replace("100rel, ", "");
};
append_hf("P-hint: GATEWAY\r\n");
# use UDP to guarantee well-known sender port (TCP ephemeral)
t_relay_to_udp("ip address","5060");
};
} else {
if (uri=~"^sip:001*@") { # ... forward to gateways then;
#Cheque se inicia com 00
if (uri=~"^sip:00") {
# strip the leading "1"
strip(2);
# if you have passed through all the checks, let your call go to GW!
rewritehostport("ip address:5060");
# snom conditioner
if (method=="INVITE" && search("User-Agent: snom")) {
replace("100rel, ", "");
};
append_hf("P-hint: GATEWAY\r\n");
# use UDP to guarantee well-known sender port (TCP ephemeral)
t_relay_to_udp("ip address","5060");
};
};
};
};
}
Jadylson da Rocha Passos Bomfim
Redevox Telecom
Uberlandia +55 34 3234-7813
São Paulo +55 11 5055-6888
Móvel +55 34 9103-6854
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Version: 6.0.643 / Virus Database: 411 - Release Date: 25/3/2004
Hi all, i am new in this list.
In my ser-0.8.12-0 (rpm installed) system i have problems with aliases, when create an alias binding between any numerical alias and my alphanumeric user with serctl i note that the Expires field is filled by an negative number like -3221225512. Whe i do a serctl alias show [numeric alias] i receive a "404 No registered contacts found" and the line "Binding '4517','sip:ecolombo@arcotel.net' has expired" in my messages file.
The alias is never dumped to the mysql database and if i call to the alias a 404 Not Found is received.
-------------------------------------------------
Paste of system commands and results:
-------------------------------------------------
[root@myser log]# serctl alias show
Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y
===Domain list===
---Domain---
name : 'aliases'
size : 512
table: 0x422b42b8
d_ll {
n : 0
first: (nil)
last : (nil)
}
---/Domain---
---Domain---
name : 'location'
size : 512
table: 0x422b2258
d_ll {
n : 0
first: (nil)
last : (nil)
}
---/Domain---
===/Domain list===
[root@myser log]# serctl alias add 4517 sip:ecolombo@arcotel.net
sip:ecolombo@arcotel.net
200 Added to table
('4517','sip:ecolombo@arcotel.net') to 'aliases'
[root@myser log]# serctl alias show
Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y
===Domain list===
---Domain---
name : 'aliases'
size : 512
table: 0x422b42b8
d_ll {
n : 1
first: 0x422b62c0
last : 0x422b62c0
}
...Record(0x422b62c0)...
domain: 'aliases'
aor : '4517'
~~~Contact(0x422b6300)~~~
domain : 'aliases'
aor : '4517'
Contact: 'sip:ecolombo@arcotel.net'
Expires: -3221225478
q : 1.00
Call-ID: 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
replic : 0
State : CS_NEW
Flags : 0
next : (nil)
prev : (nil)
~~~/Contact~~~~
.../Record...
---/Domain---
---Domain---
name : 'location'
size : 512
table: 0x422b2258
d_ll {
n : 0
first: (nil)
last : (nil)
}
---/Domain---
===/Domain list===
[root@myser log]# serctl alias show 4517
404 No registered contacts found
[root@myser log]# serctl alias show
Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y
===Domain list===
---Domain---
name : 'aliases'
size : 512
table: 0x422b42b8
d_ll {
n : 1
first: 0x422b62c0
last : 0x422b62c0
}
...Record(0x422b62c0)...
domain: 'aliases'
aor : '4517'
~~~Contact(0x422b6300)~~~
domain : 'aliases'
aor : '4517'
Contact: 'sip:ecolombo@arcotel.net'
Expires: -3221225512
q : 1.00
Call-ID: 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
replic : 0
State : CS_NEW
Flags : 0
next : (nil)
prev : (nil)
~~~/Contact~~~~
.../Record...
---/Domain---
---Domain---
name : 'location'
size : 512
table: 0x422b2258
d_ll {
n : 0
first: (nil)
last : (nil)
}
---/Domain---
===/Domain list===
[root@myser log]# tail -2 /var/log/messages
May 6 11:52:56 billing ser: ser startup succeeded
May 6 11:53:56 billing /usr/sbin/ser[30873]: Binding '4517','sip:ecolombo@arcotel.net' has expired
Thanks
Ezequiel Colombo
ARCOTEL S.A.
Hi there,
I have got a problem with ser pa module (presence module).
When I include it into my ser.cfg ( loadmodule
"/usr/local/lib/ser/modules/pa.so" ), I cannot start ser correctly.
Here is my syslog:
May 6 13:23:05 sip ser: Listening on
<<<<<<<<<<<<<<<SNIP>>>>>>>>>>>>>>>>>
May 6 13:23:06 sip ser: ser startup succeeded
This should mean that ser is running.
But when i call "service ser status", it says:
ser dead but subsys locked
Has anyone a reason for this ?
I am planning to include presence display in serweb...
Oliver
hello
Logout and Faq don't display in user_management page contrary admin page
so users can't logout !!
I looked in page.php without succes to solve this problem
I use serweb 2004
Harry
Hi all,
in order to achieve reliability I am currently trying to setup
two SIP proxy responsible for a single DNS SRV domain.
My User Agents do support a DNS fail-over
mechanism and they are able to communicate with
the secondary Proxy Server when the dialog with the Primary Proxy fails.
At this time I am setting up two proxy servers (primary and secondary) both responsible
for the same DNS SRV domain.
Each Proxy do connect to its local mysql database (located on the same host
of the SER) and they are both configured to exchange REGISTERs using the
t_replicate() function.
Now I have to setup the mysql DB replication and I have some questions:
1) Is the described way of setting up a primary and a secondary Proxy server right ?
2) Which kind of DB replication should I setup ?
3) Do I have to exclude the USRLOC table from the replication or not ?
(the USRLOC replication should be guaranteed by the REGISTERs exchange)
Thank you in advance,
Andrea Bondavalli
I.NET S.p.A. - Research & Development
mailto:a.bondavalli@inet.it
Tel. +39.02.328637232 fax. +39.02.328637702
Hi All,
I'm having a problem with mysql accounting, I tested a call to my
mobile, after 20 secs I ended the call. But when I looked at mysql,
there was no BYE, but I'm pretty sure that I got a BYE because I was
looking at "ngrep 5060" from start up to end of the call.
------------------------------------------------------------------------
| sip_method | time |
+----------------------------------+
| ACK | 2004-05-06 16:40:51 |
| INVITE | 2004-05-06 16:40:51 |
+----------------------------------+
What could be wrong? Why is it that the BYE was not inserted to the
database?
TIA
Ronald
