28 Nov
2003
28 Nov
'03
5:59 p.m.
Hi!
sems version 1.0 is for ser0.8.11
the cvs version of sems is for ser0.8.12
klaus
-----Original Message-----
From: Mario Kolberg [mailto:mko@cs.stir.ac.uk]
Sent: Friday, November 28, 2003 5:50 PM
To: serusers
Subject: [Serusers] Re: voicemail config
Hi,
thanks for your help! I have now the two instances of ser working ok.
The vm ser instance also writes the message to the fifo and this is
picked up by Sems. However, Sems is complaining about the "wrong FIFO
interface version". Sems seems to pick up version 0.2. This
matches what
the ser proxy writes to the FIFO. I run ser 0.8.12 (stable
version) and
sems version 0.1.0 which appears to be the most recent.
How can I make ser and sems agree on the FIFO interface version?
Thanks,
Mario
--
The University of Stirling is a university established in Scotland by
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28 Nov
28 Nov
6:23 p.m.
New subject: [Serusers] Re: voicemail config
Hi,
I have just tried getting the CVS version of sems. But no luck!
I tried:
CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot
export CVSROOT
cvs co answer_machine
but then I get the message that the access to cvsroot/answer_machine is
rejected for anonymous users. Am I trying the right place?
Thanks,
Mario
Klaus Darilion wrote:
Hi!
sems version 1.0 is for ser0.8.11
the cvs version of sems is for ser0.8.12
klaus
>-----Original Message-----
>From: Mario Kolberg [mailto:mko@cs.stir.ac.uk]
>Sent: Friday, November 28, 2003 5:50 PM
>To: serusers
>Subject: [Serusers] Re: voicemail config
>
>
>Hi,
>
>thanks for your help! I have now the two instances of ser working ok.
>The vm ser instance also writes the message to the fifo and this is
>picked up by Sems. However, Sems is complaining about the "wrong FIFO
>interface version". Sems seems to pick up version 0.2. This
>matches what
>the ser proxy writes to the FIFO. I run ser 0.8.12 (stable
>version) and
>sems version 0.1.0 which appears to be the most recent.
>
>How can I make ser and sems agree on the FIFO interface version?
>
>Thanks,
>Mario
--
The University of Stirling is a university established in Scotland by
charter at Stirling, FK9 4LA. Privileged/Confidential Information may
be contained in this message. If you are not the addressee indicated
in this message (or responsible for delivery of the message to such
person), you may not disclose, copy or deliver this message to anyone
and any action taken or omitted to be taken in reliance on it, is
prohibited and may be unlawful. In such case, you should destroy this
message and kindly notify the sender by reply email. Please advise
immediately if you or your employer do not consent to Internet email
for messages of this kind.
6:29 p.m.
New subject: [Serusers] Re: voicemail config
On 28-11 17:23, Mario Kolberg wrote:
Hi,
I have just tried getting the CVS version of sems. But no luck!
I tried:
CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot
It should be CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/sems
Jan.
2 Dec
2 Dec
8:35 p.m.
New subject: [Serusers] Re: voicemail config
I am running two instances of SER on the same machine. One is the regular
config running on 5060 and the other is the voicemail config running on
5090. I also have a different fifo file for both of the configs. I dont see
any traffic on the port 5090 at all. I mean, the call is not forwarded to
the voicemail. Please help. Thank you.
8:33 p.m.
New subject: [Serusers] Re: voicemail config
On Tuesday 02 December 2003 20:35, Sesha B wrote:
I am running two instances of SER on the same machine.
One is the regular
config running on 5060 and the other is the voicemail config running on
5090. I also have a different fifo file for both of the configs. I dont see
any traffic on the port 5090 at all. I mean, the call is not forwarded to
the voicemail. Please help. Thank you.
Are you sure that you listen on the correct interface? Loopback interface in
this case, e.g. 'ngrep -d lo port 5090'.
Regards
Nils
9:27 p.m.
New subject: [Serusers] Re: voicemail config
Yeah, I'm listening on the loopback. I observed something interesting. When
I stop the first instance, the second instance (Which earlier used to go to
the voicemail directly), is also not working. It works only if I change the
port back to 5060 on the second instance of SER. I dont understand where I
am doing wrong. I'm attaching my ser.cfg and ser-vm.cfg. Please help. Thank
you.
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Tuesday, December 02, 2003 2:33 PM
To: sesha(a)iic.com; Jan Janak; Mario Kolberg
Cc: Klaus Darilion; serusers
Subject: Re: [Serusers] Re: voicemail config
On Tuesday 02 December 2003 20:35, Sesha B wrote:
I am running two instances of SER on the same machine.
One is the regular
config running on 5060 and the other is the voicemail config running on
5090. I also have a different fifo file for both of the configs. I dont
see
any traffic on the port 5090 at all. I mean, the call
is not forwarded to
the voicemail. Please help. Thank you.
Are you sure that you listen on the correct interface? Loopback interface in
this case, e.g. 'ngrep -d lo port 5090'.
Regards
Nils
9:43 p.m.
New subject: [Serusers] Re: voicemail config
I guess your second Ser is not starting, because then path and name of the
fifo is the same for both Ser instances.
Nils
On Tuesday 02 December 2003 21:27, Sesha B wrote:
Yeah, I'm listening on the loopback. I observed
something interesting. When
I stop the first instance, the second instance (Which earlier used to go to
the voicemail directly), is also not working. It works only if I change the
port back to 5060 on the second instance of SER. I dont understand where I
am doing wrong. I'm attaching my ser.cfg and ser-vm.cfg. Please help. Thank
you.
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Tuesday, December 02, 2003 2:33 PM
To: sesha(a)iic.com; Jan Janak; Mario Kolberg
Cc: Klaus Darilion; serusers
Subject: Re: [Serusers] Re: voicemail config
On Tuesday 02 December 2003 20:35, Sesha B wrote:
I am running two instances of SER on the same
machine. One is the regular
config running on 5060 and the other is the voicemail config running on
5090. I also have a different fifo file for both of the configs. I dont
see
any traffic on the port 5090 at all. I mean, the
call is not forwarded to
the voicemail. Please help. Thank you.
Are you sure that you listen on the correct interface? Loopback interface
in this case, e.g. 'ngrep -d lo port 5090'.
Regards
Nils
9:56 p.m.
New subject: [Serusers] Re: voicemail config
Oops.. I have the second fifo as /tmp/vm_ser_fifo. I was just messing around
changing those while I was testing. So, it is not working with a different
fifo too.
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Tuesday, December 02, 2003 3:44 PM
To: sesha(a)iic.com
Cc: serusers
Subject: Re: [Serusers] Re: voicemail config
I guess your second Ser is not starting, because then path and name of the
fifo is the same for both Ser instances.
Nils
On Tuesday 02 December 2003 21:27, Sesha B wrote:
to
Yeah, I'm listening on the loopback. I observed
something interesting.
When
I stop the first instance, the second instance (Which
earlier used to go
to
the voicemail directly), is also not working. It works
only if I change
the
port back to 5060 on the second instance of SER. I
dont understand where I
am doing wrong. I'm attaching my ser.cfg and ser-vm.cfg. Please help.
Thank
you.
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Tuesday, December 02, 2003 2:33 PM
To: sesha(a)iic.com; Jan Janak; Mario Kolberg
Cc: Klaus Darilion; serusers
Subject: Re: [Serusers] Re: voicemail config
On Tuesday 02 December 2003 20:35, Sesha B wrote:
> I am running two instances of SER on the same machine. One is the
regular
config running
on 5060 and the other is the voicemail config running on
5090. I also have a different fifo file for both of the configs. I dont
see
> any traffic on the port 5090 at all. I mean, the call is not forwarded the voicemail.
Please help. Thank you.
Are you sure that you listen on the correct interface? Loopback interface
in this case, e.g. 'ngrep -d lo port 5090'.
Regards
Nils
9:54 p.m.
New subject: [Serusers] Re: voicemail config
On Tuesday 02 December 2003 21:56, Sesha B wrote:
Oops.. I have the second fifo as /tmp/vm_ser_fifo. I
was just messing
around changing those while I was testing. So, it is not working with a
different fifo too.
I dont know if it is messed too, but ser.cfg contains 'fork=no', which means
that this Ser instance will only listen on the first interface (normaly the
loopback interface).
Nils
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Tuesday, December 02, 2003 3:44 PM
To: sesha(a)iic.com
Cc: serusers
Subject: Re: [Serusers] Re: voicemail config
I guess your second Ser is not starting, because then path and name of the
fifo is the same for both Ser instances.
Nils
On Tuesday 02 December 2003 21:27, Sesha B wrote:
to
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Yeah, I'm listening on the loopback. I
observed something interesting.
When
I stop the first instance, the second instance
(Which earlier used to go
to
the voicemail directly), is also not working. It
works only if I change
the
port back to 5060 on the second instance of SER.
I dont understand where
I am doing wrong. I'm attaching my ser.cfg and ser-vm.cfg. Please help.
Thank
you.
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Tuesday, December 02, 2003 2:33 PM
To: sesha(a)iic.com; Jan Janak; Mario Kolberg
Cc: Klaus Darilion; serusers
Subject: Re: [Serusers] Re: voicemail config
On Tuesday 02 December 2003 20:35, Sesha B wrote:
> I am running two instances of SER on the same machine. One is the
regular
config
running on 5060 and the other is the voicemail config running on
5090. I also have a different fifo file for both of the configs. I dont
see
> any traffic on the port 5090 at all. I mean, the call is not forwarded the
voicemail. Please help. Thank you.
Are you sure that you listen on the correct interface? Loopback interface
in this case, e.g. 'ngrep -d lo port 5090'.
Regards
Nils 
10:37 p.m.
New subject: [Serusers] What's the best solutions ? RTP Proxy or STUN Server ?
Hi
I'm looking for the best solution for Private IPs accessing my Public SER
server
I was testing RTP Proxy, but I don't like all the traffic going in & out of
my server
I was looking for a STUN server, but I only find the one at Vovida, and I
don't like it (I don't like products without documentation)
Any sugestion ?
I'm testing X-Lite, Cisco ATA & GrandStream
Pablo Murillo
10:38 p.m.
New subject: [Serusers] What's the best solutions ? RTP Proxy or STUN Server ?
at best both, STUN does not help with all kinds of NATs. Use STUN
if you can, RTP proxy otherwise. -jiri
At 10:37 PM 12/2/2003, Pablo Murillo wrote:
Hi
I'm looking for the best solution for Private IPs accessing my Public SER
server
I was testing RTP Proxy, but I don't like all the traffic going in & out of
my server
I was looking for a STUN server, but I only find the one at Vovida, and I
don't like it (I don't like products without documentation)
Any sugestion ?
I'm testing X-Lite, Cisco ATA & GrandStream
Pablo Murillo
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
10:48 p.m.
New subject: [Serusers] What's the best solutions ? RTP Proxy or STUN Server ?
Jiri Kuthan wrote:
at best both, STUN does not help with all kinds of
NATs. Use STUN
if you can, RTP proxy otherwise. -jiri
You can't choose between the fruit basket and an apple...
STUN checks the NAT situation from a client point of view. STUN does not
do magic to let your calls through, it gives the UA an idea on how the
world looks and then the UA makes the choice on how to get traffic going
or give up and eat up all the fruits in the basket.
If you have problems with your NAT, the RTP proxy might be part of the solution to
get calls going in combination with SER module Nathelper. The whole idea is
to take care of the media stream when the two SIP clients can't handle the
media stream across the NAT's. Yes, it shouldn't be necessary.
Other solutions involve symmetric RTP, see
http://www.voip-info.org/wiki-RTP+Symmetric
More info on STUN:
http://www.voip-info.org/tiki-index.php?page=Stun
Vovida.org have an Open Source STUN server.
Pointer here:
http://www.voip-info.org/tiki-index.php?page=Open%20Source%20Voip%20software
Disclaimer: This is a messy area. I could be all wrong... :-)
/Olle
10:56 p.m.
New subject: [Serusers] What's the best solutions ? RTP Proxy or STUN Server ?
At 10:48 PM 12/2/2003, Olle E. Johansson wrote:
Jiri Kuthan wrote:
I'm not good in agronomy so I can't compete in statements about fruits,
apples and other healthy food.
However as for NATs, the strategy "use STUN by default, rtp relay as last
resort"
works pretty well. STUN actually does the magic to let your calls through
for many NATs.
-jiri
at best both, STUN does not help with all kinds of
NATs. Use STUN
if you can, RTP proxy otherwise. -jiri
You can't choose between the fruit basket and an apple... STUN checks the NAT situation from a client point of
view. STUN does not
do magic to let your calls through, it gives the UA an idea on how the
world looks and then the UA makes the choice on how to get traffic going
or give up and eat up all the fruits in the basket.
If you have problems with your NAT, the RTP proxy might be part of the solution to
get calls going in combination with SER module Nathelper. The whole idea is
to take care of the media stream when the two SIP clients can't handle the
media stream across the NAT's. Yes, it shouldn't be necessary.
Other solutions involve symmetric RTP, see
http://www.voip-info.org/wiki-RTP+Symmetric
More info on STUN:
http://www.voip-info.org/tiki-index.php?page=Stun
Vovida.org have an Open Source STUN server.
Pointer here:
http://www.voip-info.org/tiki-index.php?page=Open%20Source%20Voip%20software
Disclaimer: This is a messy area. I could be all wrong... :-)
/Olle
--
Jiri Kuthan http://iptel.org/~jiri/
11:31 p.m.
New subject: [Serusers] What's the best solutions ? RTP Proxy or STUN Server ?
Jiri Kuthan wrote:
At 10:48 PM 12/2/2003, Olle E. Johansson wrote:
I didn't involve tofu :-)
Jiri Kuthan wrote:
I'm not good in agronomy so I can't compete in statements about fruits,
apples and other healthy food. at best both, STUN does not help with all kinds of
NATs. Use STUN
if you can, RTP proxy otherwise. -jiri
You can't choose between the fruit basket and an apple...
However as for NATs, the strategy "use STUN by default, rtp relay as last
resort"
works pretty well. STUN actually does the magic to let your calls through
for many NATs.
Your right, but for documentation I should say "use STUN
discovery to make the
client behave correct for your network. If it doesn't work out, use a RTP
relay".
Or have I missed something in regards to STUN? STUN, TURN, COMEDIA - I wish
we didn't have NAT's... IPv6 - a glory new world.
/O ;-)
-jiri
--
*** Olle E. Johansson, oej(a)edvina.net
Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net
Runbovägen 10, 192 48 Sollentuna, Sweden
Phone: +46 8 594 78 810, Fax: +46 8 594 78 820
STUN checks the NAT situation from a client point
of view. STUN does not
do magic to let your calls through, it gives the UA an idea on how the
world looks and then the UA makes the choice on how to get traffic going
or give up and eat up all the fruits in the basket.
If you have problems with your NAT, the RTP proxy might be part of the solution to
get calls going in combination with SER module Nathelper. The whole idea is
to take care of the media stream when the two SIP clients can't handle the
media stream across the NAT's. Yes, it shouldn't be necessary.
Other solutions involve symmetric RTP, see
http://www.voip-info.org/wiki-RTP+Symmetric
More info on STUN:
http://www.voip-info.org/tiki-index.php?page=Stun
Vovida.org have an Open Source STUN server.
Pointer here:
http://www.voip-info.org/tiki-index.php?page=Open%20Source%20Voip%20software
Disclaimer: This is a messy area. I could be all wrong... :-)
/Olle
--
Jiri Kuthan http://iptel.org/~jiri/
3 Dec
3 Dec
4:35 p.m.
New subject: [Serusers] LINUX gateway with RTP Proxy & STUN Server
Hi
Thanks for the answer on the other subject
Now, a new question
I think that I have the worst scenario
I have an Intranet with a RH 6.2 working as a Gateway/Firewall with a 512Kb
ADSL with 2 switchs 3Com & 1MB w/Cisco 575 LRE with 5 IPs & 1 swithc 3COM
I don't mix both connection
If I connect a device on the Cisco 575 with one of the public IPs,
obviously, everything works
If I connect a device to the switch (Gateway/Firewall), the problems begin
I doing the test with a soft phone (X-Ten Lite) with a GrandStream HandyTone
ATA286 & CiscoATA
With the X-Ten I can login, and I can make calls and "send" audio, but I
can't receive calls or audio when I make the call
With the Cisco ATA all works if I change the IP everytime I connect to ADSL
and use UREDIR to redirect ports to the internal IP (not a good solution)
With the GrandStream I can't login :(
I allways receive both IPs on SER "check_via_address(xxx.xxx.xxx.xxx,
192.168.0.76, 0)", and SER reject the registration
-----------------------------------------------------
if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) {
# allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if ((method=="REGISTER" || ! search("^Record-Route:"))
&& !( src_ip==192.168.0.0/16 || src_ip==10.0.0.0/8 ||
src_ip==172.16.0.0/12 )) {
log("LOG: Someone trying to register from private IP again\n");
sl_send_reply("479", "No se permiten conexiones desde IP
privadas" );
break;
};
};
-----------------------------------------------------
Now the question:
What I need to get a "clear" connection to SER with public IP from my
Intranet ?
Pablo Murillo
5:30 p.m.
New subject: [Serusers] LINUX gateway with RTP Proxy & STUN Server
I'm answering to myself :)
Now the question:
What I need to get a "clear" connection to SER with public IP from my
Intranet ?
With rtpproxy & NAT, X-Ten works better
I added to make things simple & for test:
-----------------
if (method=="INVITE") {
force_rtp_proxy();
t_on_reply("1");
}
/* NAT */
onreply_route[1] {
if (status=~"[12][0-9][0-9]")
force_rtp_proxy();
}
-----------------
This solved audio problem placing call from intranet, BUT:
- it doesn't recognize hang-up
- I can't pleace calls from ATA with it's own IP to x-ten in the intranet
what am I forgetting ?
Thanks
Pablo Murillo
10:10 p.m.
New subject: [Serusers] Re: voicemail config
I have observed that, if I run the voicemail config seperately with port
5060, the call is directly going to the voicemail. But if I change the port
to any other, it is not working!! I believe this might be the reason I am
not able to transfer to the voicemail when I run two instances of SER? Also,
in the sems.conf, the fifo file is /tmp/ser_fifo. Now, if I run 2 instances,
shouldnt we add two lines in the sems.conf?
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Tuesday, December 02, 2003 3:55 PM
To: sesha(a)iic.com
Cc: serusers
Subject: Re: [Serusers] Re: voicemail config
On Tuesday 02 December 2003 21:56, Sesha B wrote:
interface
Oops.. I have the second fifo as /tmp/vm_ser_fifo. I
was just messing
around changing those while I was testing. So, it is not working with a
different fifo too.
I dont know if it is messed too, but ser.cfg contains 'fork=no', which means
that this Ser instance will only listen on the first interface (normaly the
loopback interface).
Nils
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Tuesday, December 02, 2003 3:44 PM
To: sesha(a)iic.com
Cc: serusers
Subject: Re: [Serusers] Re: voicemail config
I guess your second Ser is not starting, because then path and name of the
fifo is the same for both Ser instances.
Nils
On Tuesday 02 December 2003 21:27, Sesha B wrote:
on
Yeah, I'm listening on the loopback. I
observed something interesting.
When
I stop the first instance, the second instance
(Which earlier used to go
to
the voicemail directly), is also not working. It
works only if I change
the
port back to 5060 on the second instance of SER.
I dont understand where
I am doing wrong. I'm attaching my ser.cfg and ser-vm.cfg. Please help.
Thank
you.
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Tuesday, December 02, 2003 2:33 PM
To: sesha(a)iic.com; Jan Janak; Mario Kolberg
Cc: Klaus Darilion; serusers
Subject: Re: [Serusers] Re: voicemail config
On Tuesday 02 December 2003 20:35, Sesha B wrote:
> I am running two instances of SER on the same machine. One is the
regular
> > config running on 5060 and the other is the voicemail config running > > 5090. I also have a different fifo file for
both of the configs. I
dont
see
> any traffic on the port 5090 at all. I mean, the call is not forwarded
to
> > the voicemail. Please help. Thank you.
>
> Are you sure that you listen on the correct interface? Loopback in this case,
e.g. 'ngrep -d lo port 5090'.
Regards
Nils
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
10:14 p.m.
New subject: [Serusers] Re: voicemail config
I hope you are aware that the lines 130-148 of your ser.cfg which you sent to
the list are never executed for any request, because every request is first
relayed by the t_relay() in line 127.
Nils
On Wednesday 03 December 2003 22:10, Sesha B wrote:
I have observed that, if I run the voicemail config
seperately with port
5060, the call is directly going to the voicemail. But if I change the port
to any other, it is not working!! I believe this might be the reason I am
not able to transfer to the voicemail when I run two instances of SER?
Also, in the sems.conf, the fifo file is /tmp/ser_fifo. Now, if I run 2
instances, shouldnt we add two lines in the sems.conf?
4 Dec
4 Dec
9:39 p.m.
New subject: [Serusers] Re: voicemail config
I'm getting a busy tone after it dials for some time on the SIP phone. In
the debug messages, i see 408 request timeout and 404 not found errors. I
have the SIP phone in the grp table in the database. I am wondering if I'd
need another instance of mysql? Or is there any other problem? Please help!!
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@iptel.org]
Sent: Wednesday, December 03, 2003 4:15 PM
To: sesha(a)iic.com
Cc: serusers
Subject: Re: [Serusers] Re: voicemail config
I hope you are aware that the lines 130-148 of your ser.cfg which you sent
to
the list are never executed for any request, because every request is first
relayed by the t_relay() in line 127.
Nils
On Wednesday 03 December 2003 22:10, Sesha B wrote:
I have observed that, if I run the voicemail config
seperately with port
5060, the call is directly going to the voicemail. But if I change the
port
to any other, it is not working!! I believe this might
be the reason I am
not able to transfer to the voicemail when I run two instances of SER?
Also, in the sems.conf, the fifo file is /tmp/ser_fifo. Now, if I run 2
instances, shouldnt we add two lines in the sems.conf?
7443
days inactive
7449
days old
18 comments
8 participants
participants (8)
-
Jan Janak -
Jiri Kuthan -
Klaus Darilion -
Mario Kolberg -
Nils Ohlmeier -
Olle E. Johansson -
Pablo Murillo -
Sesha B