I have also had problems with getting the ACK back.
I don't completely understand your configuration, you
must allow for packets going both directions, right?
Here is my config :
route
{
# check to see if the message has been around too long
# probably means that it is looping
#
if (!mf_process_maxfwd_header("10"))
{
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
#
# make sure the length of the message isn't too long!
#
if (len_gt( max_len ))
{
sl_send_reply("513", "Wow -- Message too large");
break;
};
#
# do the loose-routing thing, this is important!
#
if(loose_route())
{
log(1,"doing top loose route");
t_relay();
break;
};
# this is where I was dropping the ACKS.
# I was simply dropping these, but they must be relayed
# because they can be ACKs
if(!(uri==myself))
{
if(!t_relay())
{
sl_reply_error();
break;
};
break;
};
This gets the ACKs through for me.
By the way, I have this configured with Cisco ATAs, version 2.16.
---greg
>
>I have the same problem and posed it to the group yesterday ([Serusers]
>Ignored 200 OK message.) So far the only workaround that I have found is to
>use the rules in my gateway to rewrite the dialed digits before sending them
>to the PSTN PRI, thus leaving the origianl URI intact for SIP
>communications.
>
>One person told me that this is a bug in the Cisco ATA, but it happens on my
>IPDialog phones also. It seems to me that the INVITE is being processed by
>the SER dial rules and is rewritten, but the ACK is not.
>
>Sean
>_______________________________________________
>
>Sean Robertson
>
>NETXUSA
>p. 800-289-6389
>f. 864-233-4344 "Ask me about Voice over IP."
>http://www.netxusa.com/
>
>----- Original Message -----
>From: "Alexander Mayrhofer" <axelm(a)nic.at>
>To: <serusers(a)lists.iptel.org>
>Sent: Friday, June 27, 2003 12:15 PM
>Subject: [Serusers] rewrite & ACK forwarding problem
>
>
>>
>> Hi,
>>
>> we're running SER together with a PSTN Gateway. Before a call get's
>> forwarded to the gateway, we are rewriting the request URI to make
>> rewriting on the GW as simple as possible:
>>
>> route {
>> ...
>> strip(3); # +43xxx -> xxx
>> prefix("0"); # xxx -> 0xxx
>> rewritehostport(xxx.xxx.xxx.xxx, 5060); # request to gateway
>> route(1);
>> break;
>> ...
>>
>> SIP call flow looks like (record route enabled):
>>
>> (1) phone -> SER
>> INVITE sip:*43699xxxxxxxx@nic.at43.at SIP/2.0
>>
>> (2) SER -> phone
>> SIP/2.0 100 trying -- your call is important to us
>>
>> (3) SER -> GW
>> INVITE sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0
>>
>> (4) GW -> SER
>> SIP/2.0 100 Trying
>>
>> (5) GW -> SER
>> SIP/2.0 183 Session Progress
>>
>> (6) SER -> phone
>> SIP/2.0 183 Session Progress
>>
>> (7) GW -> SER
>> SIP/2.0 180 Ringing
>>
>> (8) SER -> phone
>> SIP/2.0 180 Ringing
>>
>> (9) GW -> SER
>> SIP/2.0 200 OK
>> Contact: <sip:0699xxxxxxxx@xx.xx.xx.xx:5060>
>>
>> (10) SER -> phone
>> SIP/2.0 200 OK
>> Contact: <sip:0699xxxxxxx@xx.xx.xx.xx:5060>
>>
>> [ call established, we can talk, but ... ]
>>
>> (11) phone -> SER
>> ACK sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0
>>
>> --> Here starts the problem. That ACK (11) never gets forwarded to the
>> Gateway, so after a few seconds, the GW starts over at (9). Those three
>> packets (9-11) repeat a few times until GW runs into a timeout and drops
>> the call.
>>
>> I have the impression that SER can't match the packet to the previous
>> requests because of the rewritten URI. Is that correct?
>>
>> The only output at debug level 3 is:
>>
>> Warning: sl_send_reply: I won't send a reply for ACK!!
>>
>> Is that a routing goof somewhere in our scripts or is that a more
>> generic problem? Is the problem that the warning indicates somehow
>> related to the fact that the ACK is not being forwarded?
>>
>> Help appreciated.
>>
>> cheers
>>
>> axelm
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
At 07:39 PM 8/14/2003, Chad Brown wrote:
>Perfect,
>
>Let me ask 2 quick follow-on questions...
>
>1. Can I go back to http://www.iptel.org/ser/tarball/ser_8_11_stable.tgz
>to get the latest patched and STABLE builds?
That's the latest, most stable source, you need to compile it myself.
There will be a new complete distribution by end of this month -- we are now
waiting to run SER through the upcoming SIPIT to release it.
>2. What location / version of the modules should I use when running
>builds for this location? (Serweb, mysql, jabber, etc)
All SER modules are included there.
uptodate SERWEB is now in the http://www.iptel.org/ser/tarball/ directory too.
-jiri
Hi,
I have radius authentication working thanks to Jan's help but I am still not receiving any accounting messages to my radius server. I found a mention of setting radius_log_flag but ser 0.9.11 tells me it isn't found in the "acc" module. How can I set invite and goodbye messages (or any others for that matter) to send information to my radius server for accounting purposes. I
have read at leat 90% of the old messages but have yet to find an answer.
Thanks in advance,
Steve
Hai,
We Xten released a newer version (seradmin v .03e) of SERAdmin.
Xten is building this for the ser community and we would
very much like your feedback so that we can build you a better product
The Source code is available at
http://developer.berlios.de/projects/seradmin
SERAdmin is a GUI interface between SIP Express Router (SER) and a SER
administrator.SERAdmin has an intuitive look and feel.
SERAdmin provides control over many SER tasks such as:
Start, Stop, Pause, Re-start, Monitor SER, Add User, Change Password
and EmailId Delete User, Add Alias , Edit Alias etc.
With this we can also use FIFO commands,Access controls,User Location.
The objective of the SERAdminv03e is to support the ser-0.8.11 version also. This latest SER Administration Application handles both ser-0.8.10 and ser-0.8.11.
What is new?.
The following commands available in this release
------------------------------------------------
Ping(uri) - Pinging a URI
Cisco_restart(uri) - Restart a Cisco Phone
arg - Arguments of SER
pwd - Present Working Directory
t_uac_dlg - Initiate a Transaction
you can download the components from
http://developer.berlios.de/projects/seradmin
For any feedback & help pls. contact xten_india(a)yahoo.com
Regards,
Team,
Xten_india
---------------------------------
Do you Yahoo!?
Yahoo! SiteBuilder - Free, easy-to-use web site design software
Hello All !
I'm new to SER, and I'd like to install rtpproxy.
I've found the program to compile
(https://demo.portaone.com/~sobomax/PortaSIP/rtpproxy/),
but I don't know what to do with it once it is compiled,
and I don't know what to change in SER to integrate the proxy.
Thank you all for your help !
Mathieu
Steven,
I remember having the same problem. In my case, the problem seemed to be in the
loose_route processing. I don't know what it has to do with the accounting
module, but commenting out this part made the trick in my case.
Jaime
"Steven R. Bunin" <steve(a)solaas.com> on 26/09/2003 17:11:41
To: serusers(a)lists.iptel.org
cc: (bcc: Jaime GIL/EN/HTLUK)
Subject: [Serusers] Re: Serusers Digest, Vol 5, Issue 63
Hi all,
I have successfully gotten radius authentication working and I started getting
Radius Start records for accounting but I am not sure what I am doing wrong in
regards to getting radius Stop records.
Below is the area I believe has the most affect on Radius Acccounting from my
Log File. Any suggestions would be appreciated and if it would help to see the
full Config file I will send it as well.
record_route();
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!radius_www_authorize("")) {
www_challenge("", "0");
break;
};
save("location");
break;
};
if (method =="INVITE")
{
log(1,"INVITE\n");
setflag(1);
};
if (method=="MESSAGE") {
log(1,"MESSAGE\n");
setflag(1);
};
if (method=="BYE"){
log (1, "BYE or CANCEL\n");
setflag(1);
};
if (method=="CANCEL"){
log (1, "BYE or CANCEL\n");
setflag(1);
};
Thanks in advance,
Steve
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
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Hi,
When trying to send a SIP message through the fifo server, it comes back with an
error:
"400 fifo_uac: next hop address expected"
What could be causing this and how could we monitor what gets in and out of the
fifo?
Thanks!
Jaime
*******************************************************************************
Important.
Confidentiality: This communication is intended for the above-named person and
may be confidential and/or legally privileged. Any opinions expressed in this
communication are not necessarily those of the company. If it has come to you
in error you must take no action based on it, nor must you copy or show it to
anyone; please delete/destroy and inform the sender immediately.
Monitoring/Viruses
Orange may monitor all incoming and outgoing emails in line with current
legislation. Although we have taken steps to ensure that this email and
attachments are free from any virus, we advise that in keeping with good
computing practice the recipient should ensure they are actually virus free.
Orange PCS Limited is a subsidiary of Orange SA and is registered in England No
2178917, with its address at St James Court, Great Park Road, Almondsbury Park,
Bradley Stoke, Bristol BS32 4QJ.
*******************************************************************************
>> I did place this portion inside the myself check
>>and it still tries to transfer to vm after the time expires.
>I'm puzzled -- did not you want to transfer to vm after the time
expires?
I will try to make this clearer. I am behind an ATA with a SIP proxy of
209.242.10.153. If I call someone else registered on my domain and they
are not available, I want to go to voice mail. If I call 1-800-555-1212
from my phone, I do not want my sip proxy to reroute the call to
voicemail after 10 seconds if no one answers(or ever for that matter).
Right now if I dial 18005551212 from my handset, I see the destination
as sip:18005551212@209.242.10.153 on the server which matches to myself
and ser tries to send it to voicemail.
Someone calling into the network is not a problem. They will never hit
our server unless the destination is local.
>>This is the part that I really need help with! When the call timer
>>fails, the call goes to the route[1]. How do I get it into voice mail
>>from that point?
>See bellow, I think that should work.
This is what I had originally, and I get the following syslog.
Sep 10 16:36:36 voip2 ser: parse error (127,37-38): Command cannot be
used in the block
Sep 10 16:36:36 voip2 ser: ERROR: bad config file (1 errors)
Sep 10 16:36:36 voip2 ser: ser startup failed
Is says that vm is not valid in the block. According the admin guide,
only certain commands can be used within a failure block. I assume that
is the problem here. If not, please let me know as this is exactly what
I want to do.
>THE SAME STUFF LIKE ABOVE, YOU DON'T WANT TO t_relay ANYTHING
>if(!vm("/tmp/am_fifo","voicemail")){
> t_reply("500", "SEMS
>error");
> };
> break;
I am looking for the following feature
A list of Directory Services is displayed on the serweb, (for example, Hotel List, Emergency List, Airlines List ect).
A registered user will subscribe to the intended list (Eg a hotel will subsribe to the hotel list).
A normal user will be able to view the selected list. The seleted list show only online members at that time. Off-line members will only be shown when they are login in.
Can this feature be done?
Thanks