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
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Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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
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.
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.
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
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@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
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@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
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@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:
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@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
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@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:
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@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
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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
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@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Jiri Kuthan http://iptel.org/~jiri/
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
At 10:48 PM 12/2/2003, Olle E. Johansson wrote:
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...
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
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/
Jiri Kuthan wrote:
At 10:48 PM 12/2/2003, Olle E. Johansson wrote:
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...
I'm not good in agronomy so I can't compete in statements about fruits, apples and other healthy food.
I didn't involve tofu :-)
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
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/
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): .*@(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
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
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@iic.com Cc: serusers Subject: Re: [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@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:
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@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
- 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
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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?
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@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?
-
Jan Janak -
Jiri Kuthan -
Klaus Darilion -
Mario Kolberg -
Nils Ohlmeier -
Olle E. Johansson -
Pablo Murillo -
Sesha B