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Hello All,
If I wanted to see the debug messages in /var/log/messages, then
how high do I need to set the debug level in ser.cfg.
I have "L_DBG" in LOG statement and for some reason I am not seeing them in
the /var/log/messages.
my current setting in the ser.cfg file is debug=4
thanks,
Jignesh Gandhi
Software Engineer II
Jignesh.Gandhi(a)glenayre.com
(770)283 1706
Hi All,
I recently started working for a voip company and I am put
into testing of ser and asterisk.
The set up is as follows.
UA1-------NAT------------SER----------------Asterisk
|
UA2
All the messages(Invites,Oks, ACKs) from the UA1 are passed to the SER,
the SER forwards them to the Asterisk, then the asterisk hands it back
to the SER and the SER forwards it to UA2. Same works vice a versa
too.(I have no clue why the set up is this way!!)
I have NAThelper/RTP proxy running.
Both UA1 and UA2 are registered to the SER
UA1 can make a call to UA2 and both can hear each other.
But UA2 cannot make a call to UA1. Here is what happens.
UA2 sends invite to SER.The SER changes the headers and 'c' and 'm'
field in SDP and forwards it to Asterisk. The Asterisk acting as B2BUA,
again changes Call ID and required fields and hands it back to SER.The
SER then forwards the invite to UA1.UA1 sends an OK back to SER which
the SER gives it to Asterisk.As soon as Asterisk receives OK, it sends
an ACK back to SER for UA1, but the SER never forwards this ACK back to
UA1.The Asterisk then sends OK to SER for UA2 which SER forwards
correctly.But UA1 is still waiting for the ACK...hence the signalling
gets looped.(ringing can be heard).
The ACK which the Asterisk sends back contains the field Route<sip:
617......@private IP of UA1>
The only places the Asterisk can get this private IP is from the
first OK which SER sends Asterisk.And in this OK two fields contain this
address, the contact field in header and 'o' in SDP. So my questions
are:
1) What is the route field for.Does it tell the SER where to forward the
request?
2) Where does the asterisk pick up the Route field from?(contact header
or 'o' field)
3) If this is the problem that what changes do I need to make?
I hope I have not confused you.Would really appreciate any help.
Thanks,
Hitesh Jain.
Hey All,
We experienced a problem yesterday with SER and now are attempting to track
down the reason that it happened in order to "hopefully" find a way to
prevent it from happening again.
We have been running SER successfully for the past 5+ months without much
incident until yesterday morning. SER suddenly stopped responding to TCP
requests.
Once we restarted it then it came back.
During the period that SER had stopped responding we received the following
errors repeatedly in syslog.
Nov 30 13:00:37 ss1 /usr/local/sbin/ser[19457]: ERROR:
tcp_blocking_connect: timeout (10)
Nov 30 13:00:48 ss1 /usr/local/sbin/ser[19459]: ERROR:
tcp_blocking_connect: timeout (10)
Any idea what would cause this specific error? Or what would cause SER to
suddenly behave this way?
Thanks for the help!
Darren Nay
VOIP Network Development
Ionosphere, Inc.
dnay(a)ionosphere.net <mailto:dnay@ionosphere.net>
Hi,
Is there anyone out there who has used the SER Application Agent
recently? I don't notice any particularly obvious links to it from the
main iptel page, but I see it on http://www.iptel.org/aa/.
The reason I'm asking is that it seems to implement some B2BUA
functionality (e.g. click2dial) based on SER. We're looking at using
SER as our basic Proxy, but we may want to evolve into B2BUA
functoinaltiy like click2dial or other kinds of 3rd-party call control.
I'm just wondering if anyone out there has had any (successful or
unsuccessful) experience trying to do this.
Thanks
Alex Rootham
Hi all,
Its me again ... I got the RTP proxy thing working
... SER connects to RTPProxy ... then its stuck there
... doesnt proceed. the console log i get is like
this:
Listening on
192.168.1.21 [192.168.1.21]:5060
Aliases: machinename.localdomain:5060 machinename:*
WARNING: no fork mode
stateless - initializing
Maxfwd module- initializing
0(8908) mod_init(): Database connection opened
successfuly
textops - initializing
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
2(0) INFO: fifo process starting: 8912
1(8910) rtpp_test: RTP proxy found, support for it
enabled
2(8912) rtpp_test: RTP proxy found, support for it
enabled
2(8912) SER: open_uac_fifo: fifo server up at
/tmp/ser_fifo...
0(8908) rtpp_test: RTP proxy found, support for it
enabled
1(8910) 2(8912) INFO: signal 15 received
Thanks in advance. Any pointer would be appreciated.
- Tareq
__________________________________________________
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RTPProxy runs on a unix port and must be running prior to running SER.
Make sure you have the latest
version of RTPProxy. I use portaone's version and it works great.. If
you are certain both of the above are verified, you can run "rtpproxy
-f" to keep it in foreground and it may pass some errors. Also, I've
never used SER/RTPPRoxy when itself was *behind* a NAT. Not sure how
this will react. Good luck!
Matt
-----Original Message-----
From: Tareq Siraj [mailto:to_tareq@yahoo.com]
Sent: Thursday, December 02, 2004 7:24 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] SER+ASTERISK+RTP Proxy
Hi All,
I am trying to setup a pre-paid billing system and
from the search results, i see most ppl are refering
to this combination. Well, i got the latest versions
and compiled ok. I edit the asterisk conf files for my
gateway and ser.conf to use SER. Now, i run asterisk
and rtp proxy ... but when i run ser, it gives me the
following message.
Listening on
192.168.1.21 [192.168.1.21]:5060
Aliases: machinename.localdomain:5060 machinename:*
WARNING: no fork mode
stateless - initializing
Maxfwd module- initializing
0(6768) mod_init(): Database connection opened
successfuly
textops - initializing
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
2(0) INFO: fifo process starting: 6772
1(6770) ERROR: send_rtpp_command: can't read reply
from a RTP proxy
1(6770) WARNING: rtpp_test: can't get version of the
RTP proxy
1(6770) WARNING: rtpp_test: support for RTP proxyhas
been disabled temporarily
2(6772) ERROR: send_rtpp_command: can't read reply
from a RTP proxy
2(6772) WARNING: rtpp_test: can't get version of the
RTP proxy
2(6772) WARNING: rtpp_test: support for RTP proxyhas
been disabled temporarily
2(6772) SER: open_uac_fifo: fifo server up at
/tmp/ser_fifo...
0(6768) ERROR: send_rtpp_command: can't read reply
from a RTP proxy
0(6768) WARNING: rtpp_test: can't get version of the
RTP proxy
0(6768) WARNING: rtpp_test: support for RTP proxyhas
been disabled temporarily
Any idea why this is happening? I am a total newbie to
this system. Any help would be appreciated. Thanks.
- tareq
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_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi All,
I am trying to setup a pre-paid billing system and
from the search results, i see most ppl are refering
to this combination. Well, i got the latest versions
and compiled ok. I edit the asterisk conf files for my
gateway and ser.conf to use SER. Now, i run asterisk
and rtp proxy ... but when i run ser, it gives me the
following message.
Listening on
192.168.1.21 [192.168.1.21]:5060
Aliases: machinename.localdomain:5060 machinename:*
WARNING: no fork mode
stateless - initializing
Maxfwd module- initializing
0(6768) mod_init(): Database connection opened
successfuly
textops - initializing
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
2(0) INFO: fifo process starting: 6772
1(6770) ERROR: send_rtpp_command: can't read reply
from a RTP proxy
1(6770) WARNING: rtpp_test: can't get version of the
RTP proxy
1(6770) WARNING: rtpp_test: support for RTP proxyhas
been disabled temporarily
2(6772) ERROR: send_rtpp_command: can't read reply
from a RTP proxy
2(6772) WARNING: rtpp_test: can't get version of the
RTP proxy
2(6772) WARNING: rtpp_test: support for RTP proxyhas
been disabled temporarily
2(6772) SER: open_uac_fifo: fifo server up at
/tmp/ser_fifo...
0(6768) ERROR: send_rtpp_command: can't read reply
from a RTP proxy
0(6768) WARNING: rtpp_test: can't get version of the
RTP proxy
0(6768) WARNING: rtpp_test: support for RTP proxyhas
been disabled temporarily
Any idea why this is happening? I am a total newbie to
this system. Any help would be appreciated. Thanks.
- tareq
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Ron:
The failure_route will be activated when a transaction fails. This is
different
than a route because it is triggered by the failure of a previously executed
command. The reason I asked about the word "automatically" is that a
failure_route is activated after an initial attempt, in this case a
t_relay command,
fails. It will not be activated when an invite for a user is first
received by
the proxy. An example follows:
#
# Redirect user to vm if not availabile?
#
if (is_user_in("Request-URI", "voicemail")) {
t_on_failure("6");
setflag(6);
log(1, "[SER]: Flag for VM redirect successful. \n");
};
failure_route[6] {
xlog("L_INFO", "\n[SER]: START FAILURE BLOCK #6 Unavailable user:
Time: [%Tf] Method: <%rm> From uri <%fu> To < %tu> IP source
address <%is>
R-uri: <%ru> Contact Header: <%ct> \n\n");
revert_uri();
rewritehostport("voicemailserver.myco.com:5070");
append_branch();
t_relay_to_udp("voicemailserver.myco.com", "5070");
break;
}
-Steve
Ron Ramos wrote:
>No idea what a failure_route do. But that sounds like what I need.
>Do you have any sample configuration for that. Thank you
>
>-----Original Message-----
>From: Steve Blair [mailto:blairs@isc.upenn.edu]
>Sent: Thursday, December 02, 2004 4:44 AM
>To: Klaus Darilion
>Cc: ron(a)silverbackasp.com; serusers(a)lists.iptel.org
>Subject: Re: [Serusers] two gateways
>
>
>
> That is what I do too but I am not sure if this fits with Ron's
>definition of "automatically". Does it Ron?
>
>Klaus Darilion wrote:
>
>
>
>>use a failure_route
>>
>>klaus
>>
>>Ronald Ramos wrote:
>>
>>
>>
>>>hi,
>>>
>>>How can i configure my sip proxy to automatically know that the PSTN
>>>gateway
>>>is down and transfer all calls to another gateway?
>>>
>>>thank you
>>>
>>>regards,
>>>ron
>>>
>>>_______________________________________________
>>>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
>>
>>
--
ISC Network Engineering
The University of Pennsylvania
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Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
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sip:blairs@upenn.edu