I will be on vacation from the 19th of December to the 31st of December.
For technical support, contact Alek or Jarod at techsupport(a)widernet.org.
See you next year!
-- Peter
Hello Darren,
Your python package seems to be broken. _socket is the low level
library from the socket module and is part of the python core. Or some
missing C libraries that _socket.so needs
Check with ldd /usr/lib/python2.2/lib-dynload/_socket.so that all
required libraries are there.
Regards,
Dan
----------
Hey All,
I've just installed the python version of the rtpproxy, but it won't
startup. I have followed all directions in the README file for install
to the letter. The system is a Sun Netra 1400 running Solaris 9
(sparc). I am using Python v2.2 with the Optik module installed.
The errors that I am receiving are:
Traceback (most recent call last):
File "rtpproxy.py", line 19, in ?
import rtphandler
File "rtphandler.py", line 8, in ?
File "/usr/local/lib/python2.2/asyncore.py", line 51, in ?
import socket
File "/usr/local/lib/python2.2/socket.py", line 41, in ?
from _socket import *
ImportError: No module named _socket
I get this same thing if I call the script from /etc/init.d/rtpproxy
start .. or just ./rtpproxy.py from the app directory.
Unfortunately I am rather unfamiliar with Python so I'm stumped at the
moment. I'm a perl guy... :)
Any assistance would be very much appreciated!
Thanks,
Darren Nay - dnay at libertyisp.com
Not sure about the loose routing.. here is my config file..
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
dns=yes
rev_dns=yes
check_via=no
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
#loadmodule "/usr/local/lib/ser/modules/textops.so"
#loadmodule "/usr/local/lib/ser/modules/vm.so"
#loadmodule "/usr/local/lib/ser/modules
#loadmodule "/usr/local/lib/ser/modules
#loadmodule "/usr/local/lib/ser/modules
#loadmodule "/usr/local/lib/ser/modules
#loadmodule "/usr/local/lib/ser/modules
#loadmodule "/usr/local/lib/ser/modules
# ----------------- setting module-specific parameters ---------------
modparam("auth_db", "db_url","mysql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("usrloc", "db_mode", 1)
# -- acc params --
modparam("acc", "log_level", 1)
# that is the flag for which we will account -- don't forget to
# set the same one :-)
modparam("acc", "db_flag", 1)
modparam("acc", "db_url","mysql://ser:heslo@localhost/ser")
modparam("acc", "db_missed_flag", 3)
modparam("tm", "fr_inv_timer", 20)
modparam("group", "db_url", "mysql://ser:heslo@localhost/ser")
# ------------------------- request routing logic -------------------
# main routing logic
alias="ion.dom"
alias="44.44.444.154"
route {
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483", "Alas Too Many Hops");
break;
};
setflag(1);
if (!(method=="REGISTER")) record_route();
if (loose_route()) {
t_relay();
break;
};
# if (!uri==myself) {
# t_relay();
# break;
# };
if (method == "REGISTER") {
if (!save("location")) {
sl_reply_error();
};
break;
};
# does the user wish redirection on no availability? (i.e., is he
# in the voicemail group?) -- determine it now and store it in
# flag 4, before we rewrite the flag using UsrLoc
if (is_user_in("Request-URI", "voicemail")) {
setflag(4);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
# handle user which was not found
route(4);
break;
};
# if user is on-line and is in voicemail group, enable redirection
if (method == "INVITE" && isflagset(4)) {
t_on_failure("1");
};
t_relay();
}
# ------------- handling of unavailable user ------------------
route[4] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" ||
met
hod == "BYE")) {
sl_send_reply("404", "Not Found");
break;
};
# not voicemail subscriber
if (!isflagset(4)) {
if (uri=~"^sip:1.*") {
setflag(1);
rewritehostport("33.33.333.157:5060");
t_relay_to_udp("33.33.333.157", "5060");
break;
};
if (uri=~"^sip:011.*") {
setflag(1);
rewritehostport("22.222.222.26:5060");
t_relay_to_udp("22.222.222.26", "5060");
break;
};
}
# forward to voicemail now
setflag(3);
rewritehostport("11.11.111.148:5060");
t_relay_to_udp("11.11.111.148", "5060");
}
# if forwarding downstream did not succeed, try voicemail running
# at bat.iptel.org:5060
failure_route[1] {
setflag(3);
revert_uri();
rewritehostport("11.11.111.148:5060");
append_branch();
t_relay_to_udp("11.11.111.148", "5060");
}
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Thursday, December 18, 2003 4:48 PM
To: Darren Sessions; serusers(a)lists.iptel.org
Subject: Re: [Serusers] RE: Serusers Digest, Vol 8, Issue 16
SER does not generate such an error, SEMS does. Possibly,
your BYEs are mistakenly passed to SEMS. They are supposed
to be processed by loose routing.
-jiri
At 08:33 PM 12/18/2003, Darren Sessions wrote:
>Situation:
>
>Endpoint is called Party (Sipura SPA2000)
>
>Calling Party is routed through from a TDM call and VoIP'd by Cisco 3640
>router then sent to SER.
>
>Endpoint receives call - everything works. When the Sipura hangs up the
>call, it sends a BYE to ser - but ser does not send the BYE to the router
to
>disconnect the TDM channel.
>
>All other situations result in a completed call with channels released on
>the router just fine.
>
>After further inspection - we noticed that when the Sipura hangs up and
send
>the BYE to SER, SER responds with this:
>
>
>SIP/2.0 481 Call/Transaction Does Not Exist
>Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f
>From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642
>To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3
>Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157
>CSeq: 101 BYE
>Contact: <sip:8644679887@44.44.444.148>
>Server: Sip EXpress router (0.8.12 (i386/linux))
>Content-Length: 0
>Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033
>req_src_ip=22.22.222.154 req_src_port=5060
>in_uri=sip:8644679887@44.44.444.148:5060
>out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0"
>
>
>Any ideas?
>
>Thanks,
>
> - Darren
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Hey All,
I've just installed the python version of the rtpproxy, but it won't startup. I have followed all directions in the README file for install to the letter. The system is a Sun Netra 1400 running Solaris 9 (sparc). I am using Python v2.2 with the Optik module installed.
The errors that I am receiving are:
Traceback (most recent call last):
File "rtpproxy.py", line 19, in ?
import rtphandler
File "rtphandler.py", line 8, in ?
File "/usr/local/lib/python2.2/asyncore.py", line 51, in ?
import socket
File "/usr/local/lib/python2.2/socket.py", line 41, in ?
from _socket import *
ImportError: No module named _socket
I get this same thing if I call the script from /etc/init.d/rtpproxy start .. or just ./rtpproxy.py from the app directory.
Unfortunately I am rather unfamiliar with Python so I'm stumped at the moment. I'm a perl guy... :)
Any assistance would be very much appreciated!
Thanks,
Darren Nay - dnay(a)libertyisp.com
Hello,
I am receiving calls from Vovida in SER. I want to forward the call
back to Vovida for a call forward application. I can do this using 2
Marshal Servers or by a stateless Redirect. Redirect doesn't work for
me b/c I want to keep track pf the call and if the called party doesn't
answer I want it to go to voicemail.
If I attempt to forward the call back to the original Vovida Marshal
Server, the Marshal Server detects that it is already in the via and
drops the packect sending a "482 Too Many Hops" message. Any
suggestions?
Call Flow
Doesn't work
GW --- Vovida MS A --Invite------> SER
Vovida MS A <------Invite-SER
Works
GW --- Vovida MS A --Invite------> SER
Vovida MS B <------Invite-SER
Works
GW --- Vovida MS A --Invite------> SER
Vovida MS B <------Redirect "302 Temporarily Moved" -SER
So, If there was a way to get SER to initiate a new invite to the call
forwarded destination, that is what I think would be the correct way. I
think this is how Asterisk does it from a call forward perspective.
Thanks in advance,
Mark
Sorry about that.
Quick question while I whip that up..
Does anyone have a configuration sample and/or knows of where one is in docs
about how to implement the nat helper module with a PSTN gateway for off-net
calls?
Thanks,
- Darren
-----Original Message-----
From: Jan Janak [mailto:jan@iptel.org]
Sent: Thursday, December 18, 2003 5:23 PM
To: Darren Sessions
Cc: Andres; serusers(a)lists.iptel.org
Subject: Re: [Serusers] Unformatted Exerpts from for BYE problem
This is unreadable, please send us proper dumps -- you can use ngrep to
make them. Also please attach your config file.
Jan.
On 18-12 15:31, Darren Sessions wrote:
> SIP/2.0 200 OK
>
> To: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642
>
> From: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3
>
> Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157
>
> CSeq: 101 INVITE
>
> Via: SIP/2.0/UDP 63.86.212.154;branch=z9hG4bKb2f4.035.0
>
> Via: SIP/2.0/UDP 63.86.212.157:5060
>
> Record-Route: <sip:8646783182@63.86.212.154;ftag=322BCA98-25A3;lr>
>
> Timestamp: 1071766157
>
> Contact: John Walter <sip:8646783182@63.86.212.155:5060>
>
> Server: Sipura/SPA2000-1.0.20
>
> Content-Length: 150
>
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
>
> Supported: 100rel
>
> Content-Type: application/sdp
>
> v=0
>
> o=- 88599 88599 IN IP4 192.168.0.101
>
> s=-
>
> c=IN IP4 63.86.212.155
>
> t=0 0
>
> m=audio 16404 RTP/AVP 0
>
> a=rtpmap:0 PCMU/8000
>
> a=ptime:20
>
> a=sendrecv
>
>
?Úá?ÉF�ó��ó�����������???�
>
> ^?N�E?"8@�?Á?VÔs?¨�eÄÄ?BEACK
> sip:8646783182@63.86.212.155:5060 SIP/2.0
>
> Record-Route: <sip:8646783182@63.86.212.154;ftag=322BCA98-25A3;lr>
>
> Via: SIP/2.0/UDP 63.86.212.154;branch=0
>
> Via: SIP/2.0/UDP 63.86.212.157:5060
>
> From: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3
>
> To: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642
>
> Date: Thu, 18 Dec 2003 16:49:17 GMT
>
> Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157
>
> Max-Forwards: 5
>
> Content-Length: 0
>
> CSeq: 101 ACK
>
> ?¨�e?VÔsÄÄ?á}BYE sip:8644679887@63.86.212.157:5060 SIP/2.0
>
> Via: SIP/2.0/UDP 63.86.212.155:5060;branch=z9hG4bK-b76fe29f
>
> From: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642
>
> To: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3
>
> Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157
>
> CSeq: 101 BYE
>
> Max-Forwards: 70
>
> Route: <sip:8646783182@63.86.212.154;ftag=322BCA98-25A3;lr>
>
> User-Agent: Sipura/SPA2000-1.0.20
>
> Content-Length: 0
>
> ?Úá?�?��?�����������???�
>
> ^?N�E":@�?%?VÔs?¨�eÄÄk??SIP/2.0 481 Call/Transaction Does
> Not Exist
>
> Via: SIP/2.0/UDP 63.86.212.155:5060;branch=z9hG4bK-b76fe29f
>
> From: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642
>
> To: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3
>
> Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157
>
> CSeq: 101 BYE
>
> Contact: <sip:8644679887@63.86.212.148>
>
> Server: Sip EXpress router (0.8.12 (i386/linux))
>
> Content-Length: 0
>
> Warning: 392 63.86.212.148:5060 "Noisy feedback tells: pid=20033
> req_src_ip=63.86.212.154 req_src_port=5060
> in_uri=sip:8644679887@63.86.212.148:5060
> out_uri=sip:8644679887@63.86.212.148:5060 via_cnt==0"
>
>
> -----Original Message-----
> From: Andres [ mailto:andres@telesip.net <mailto:andres@telesip.net> ]
> Sent: Thursday, December 18, 2003 3:25 PM
> To: Darren Sessions; serusers(a)lists.iptel.org
> Subject: Re: [Serusers] BYE Message Problem
>
>
> On Thursday 18 December 2003 15:11, Darren Sessions wrote:
> > Sipura Firmware is 1.0.20
> >
> > We are already in talks with Sipura about this issue as we found a bug
for
> > them yesterday. It appears though based on the Etheral traces and ngrep
> > that the Sipura is sending the BYE messages - but SER is not forwarding
> > them along.
> Can you post all the messages here(INVITE...to ...BYE)? Otherwise it is
> hard
> to guess where the problem might be. I'm sure Jiri or Jan can spot the
> error
> right away.
>
> >
> > We are using Ser 8.12 on a Sun Ultra 60 - and as I said before, all
other
> > instances work flawlessly.
> >
> > Thanks,
> >
> > - Darren
> >
> > -----Original Message-----
> > From: Andres [ mailto:andres@telesip.net <mailto:andres@telesip.net> ]
> > Sent: Thursday, December 18, 2003 2:58 PM
> > To: Darren Sessions; serusers(a)lists.iptel.org
> > Subject: Re: [Serusers] BYE Message Problem
> >
> > On Thursday 18 December 2003 14:50, Darren Sessions wrote:
> > > We also have a problem with BYE's not being sent from an on-net call
> > > (sipura to sipura).
> >
> > What firmware version are you using? We have been testing the SPA2000
for
> > about 3 weeks now and have not seen this issue.
> >
> > > -----Original Message-----
> > > From: Darren Sessions
> > > Sent: Thursday, December 18, 2003 2:32 PM
> > > To: 'serusers(a)lists.iptel.org'
> > > Subject: RE: Serusers Digest, Vol 8, Issue 16
> > > Importance: High
> > >
> > >
> > > Situation:
> > >
> > > Endpoint is called Party (Sipura SPA2000)
> > >
> > > Calling Party is routed through from a TDM call and VoIP'd by Cisco
3640
> > > router then sent to SER.
> > >
> > > Endpoint receives call - everything works. When the Sipura hangs up
the
> > > call, it sends a BYE to ser - but ser does not send the BYE to the
> router
> > > to disconnect the TDM channel.
> > >
> > > All other situations result in a completed call with channels released
> on
> > > the router just fine.
> > >
> > > After further inspection - we noticed that when the Sipura hangs up
and
> > > send the BYE to SER, SER responds with this:
> > >
> > >
> > > SIP/2.0 481 Call/Transaction Does Not Exist
> > > Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f
> > > From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642
> > > To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3
> > > Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157
> > > CSeq: 101 BYE
> > > Contact: <sip:8644679887@44.44.444.148>
> > > Server: Sip EXpress router (0.8.12 (i386/linux))
> > > Content-Length: 0
> > > Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033
> > > req_src_ip=22.22.222.154 req_src_port=5060
> > > in_uri=sip:8644679887@44.44.444.148:5060
> > > out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0"
> > >
> > >
> > > Any ideas?
> > >
> > > Thanks,
> > >
> > > - Darren
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> <http://lists.iptel.org/mailman/listinfo/serusers>
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
http://www.iptel.org/ser/doc/seruser/seruser.html#AEN852
regards,
Klaus
> -----Original Message-----
> From: listas iPfone [mailto:listas@ipfone.com.br]
> Sent: Friday, December 19, 2003 11:50 AM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] 400 Table 'aliases' not found in memory
>
>
> Hi!
>
> Hen i use the serctl alias add i get that message:
>
> 400 Table 'aliases' not found in memory, use save("aliases") or
> lookup("aliases") in the configuration script first
>
> This happens when i use the confirmation link in the user confirmation
> email..
>
> What is the problem? How can i use save("aliases") or
> lookup("aliases") in
> the configuration script ?
>
> I have that table im my database.
>
> Thanks
>
> Miklos
>
>
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
Hi All!
I´m trying to run ser as my sip server and asterisk as my pbx in the same machine with acess to pstn.
My goal is to have external users registering in ser ( with my dyndns - sipserver.com.br) and redirect the calls to my estensions in asterisk.
How to do it?
When i run ser, asterisk don´t find my phones anymore.
I have two nic in this machine 192.168.0.31/37:
[root@localhost root]# ser
Listening on
127.0.0.1 [127.0.0.1]:5060
192.168.0.37 [192.168.0.37]:5060
192.168.0.31 [192.168.0.31]:5060
Aliases: localhost:5060 localhost.localdomain:5060
i think that i need to make ser listen to one ip and asterisk to other.
If that is rigth, how can i make ser listen only on 192.168.0.31:5060?
Thanks for all
Miklos
I have tried SER 0.8.11 and SER 0.8.12 with serweb and both have the
same problem. When I reply the registration e-mail or try to log in with
the admin username in serweb's login page this message appears:
Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied
in /var/www/html/serweb/html/functions.php on line 181
sorry -- cannot open write fifo
I changed the user rights in ser_fifo from root to myself but it didn't
help.
So what is the problem.
Teppo
I agree that features are not the only thing to look at for decision-making.
The purpose for this table is to have a compiled list of SIP proxy specific
features that are already available in SER (maybe it can be added to the SER
admin manual), and have an idea of what may need development if required.
>>* Private number (permanent Caller-ID blocking):
>A way I do maintain privacy is I use a stupid user id
>(You certainly know how much identity is dislocsed if you
>send from mr_goerge_w_bush(a)hotmail.com....)
Sure, but this may not be possible if some sort of authentication is
required. Isn't this better handle on the SIP proxy server?
>>* Caller blocking:
>Yes -- we block requests from source causing annoying traffic.
Is caller blocking system wide or configurable per user?
>>* Call Transfer:
>phone feature
Don't the SIP proxy needs to support the SIP REDIR too?
>>* Support for call time limit:
>call-limitation is part of separate software
Is there software already available or are you just saying that this will
need to be developed as separate software?
>>* Redundant server configuration:
>available as separate software
Again is there some software developed for SER or are you thinking about
clustering software and/or making usage of DNS SRV resource records which
will provide some kind of redundancy?
Thanks, your input was appreciated.
-Cesar
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Wednesday, December 17, 2003 3:42 PM
To: Hernandez, Cesar; 'serusers(a)lists.iptel.org'
Subject: Re: [Serusers] SER supported feature list
At 09:19 PM 12/17/2003, Hernandez, Cesar wrote:
>Hello,
>
>I'm compiling a table of features supported in SER, such table will be
helpful to compare SER capabilities with other SIP proxy servers.
Keep in mind that's a hard job. Some of the features may be
end-device feature and some may have different mimics than
you would expect. Also, it is not only specific telephony
features which matter: importantly, there are things such
as operational characteristics (performance, scalability)
and ability to introduce new features and integrate with web.
>* Private number (permanent Caller-ID blocking):
A way I do maintain privacy is I use a stupid user id
(You certainly know how much identity is dislocsed if you
send from mr_goerge_w_bush(a)hotmail.com....)
>* Caller-ID blocking per call (like Bell *67):
see above, a phone may maintain multiple identities, one of
them less informational than the other one.
>* Caller blocking:
Yes -- we block requests from source causing annoying traffic.
>* Call Waiting (support more than one call):
phone feature (see for example www.iptel.org/tt/ for such a phone)
>* International routing dial plan:
>* Least-cost routing:
all trivial config options
>* URI/Phone manipulation: Yes
>* Find-me parallel calling: No
find-me is built-in (known under the SIP term "parallel forking")
>* Find-me consecutive calling: No
unavailable now
>* Call Transfer:
phone feature
>* Conference Calls (3-way calling): Yes
>* NAT STUN: Yes
just for sake of clarity: we have a STUN impelmentation,
but it is separate from SER
>* Billing Flat-file CDR: Yes
>* Authentication: Yes
>* Authorization: Yes
>* RADIUS authentication/Authorization: Yes
>* RADIUS accounting: Yes
>* Support for call time limit:
call-limitation is part of separate software
>* ENUM support : Yes
>* IM&P (Instant Messaging and presence): Yes (2G/SMS, SIMPLE/XMPP Jabber)
>* WEB Provisioning Interfaces: Yes
>* Programming interface to remotely provision from an OSS system
(add/remove/modify user/settings):
provisioning available as command-line or web tools
>* Programming interface to remotely modify system settings from an OSS
system (routing rules, gateways, etc.):
available as server configuration options
>* Redundant server configuration:
available as separate softwarea
>* Call return (like Bell *69):
>* Call screening:
see missed calls and click-to-dial in SERweb
>* MWI (SIP NOTIFY):
not now
-jiri
