17 Feb
2004
17 Feb
'04
2:36 p.m.
Hey Guys,
I am having issues with One way Audio for outgoing phone calls from my SIP phones. It
works fine for Incoming, but outgoing audio is not working. Also, Outgoing works fine if
I put my SIP phone on an Internet IP address, but if it's NAT'd then I get the 1
way audio on outgoing calls.
I am using the default nathelper module config, but I have hacked it a bit and maybe my
changes are causing the problem? I had to use t_relay_to_tcp for our PSTN gateway and so
I had to change the routing around a little.
Prior to today I was using SerMediaProxy (AG Projects), but I switched to PortaOnes
rtpproxy just to make sure that there wasn't a version incompatibility problem.
I have also attached my current ser.cfg file.
Anyone have any suggestions? Or should I change the routing in my ser.cfg file? or would
that make a difference?
Any help would be greatly appreciated. Thanks!
Darren Nay - dnay(a)libertyisp.com
17 Feb
17 Feb
2:41 p.m.
As I said in my previoud message, most likely that isflagset(6) test in
onreply_route is causing problems. Remove it completely and try again.
-Maxim
Darren Nay wrote:
Hey Guys,
I am having issues with One way Audio for outgoing phone calls from my
SIP phones. It works fine for Incoming, but outgoing audio is not
working. Also, Outgoing works fine if I put my SIP phone on an Internet
IP address, but if it's NAT'd then I get the 1 way audio on outgoing calls.
I am using the default nathelper module config, but I have hacked it a
bit and maybe my changes are causing the problem? I had to use
t_relay_to_tcp for our PSTN gateway and so I had to change the routing
around a little.
Prior to today I was using SerMediaProxy (AG Projects), but I switched
to PortaOnes rtpproxy just to make sure that there wasn't a version
incompatibility problem.
I have also attached my current ser.cfg file.
Anyone have any suggestions? Or should I change the routing in my
ser.cfg file? or would that make a difference?
Any help would be greatly appreciated. Thanks!
Darren Nay - dnay(a)libertyisp.com <mailto:dnay@libertyisp.com>
------------------------------------------------------------------------
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
3:09 p.m.
I removed it and I still have the same problem. My onreply_route looks like
this now.
onreply_route[1] {
if (status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
I also tried this..
onreply_route[1] {
fix_nated_contact();
force_rtp_proxy();
}
I must admit that I'm new to all this SIP routing. :(
One more thing of note is that we are getting the following errors in the
syslog.
Feb 17 12:53:15 lvl3 /usr/local/sbin/ser[19711]: ERROR: send_rtpp_command:
can't read reply from a RTP proxy
Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: extract_body:
message body has lenght zero
Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: force_rtp_proxy:
can't extract body from the message
Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: on_reply processing
failed
I just noticed this on my last call attempt.
Any more suggestions. I've tried all kinds of combinations .. I'm stumped..
Thanks!
Darren
----- Original Message -----
From: "Maxim Sobolev" <sobomax(a)portaone.com>
To: "Darren Nay" <dnay(a)libertyisp.com>
Cc: <serusers(a)lists.iptel.org>
Sent: Tuesday, February 17, 2004 2:41 PM
Subject: Re: [Serusers] RTP Proxy Help - One way Audio
As I said in my previoud message, most likely that
isflagset(6) test in
onreply_route is causing problems. Remove it completely and try again.
-Maxim
Darren Nay wrote:
> Hey Guys,
>
> I am having issues with One way Audio for outgoing phone calls from my
> SIP phones. It works fine for Incoming, but outgoing audio is not
> working. Also, Outgoing works fine if I put my SIP phone on an Internet
> IP address, but if it's NAT'd then I get the 1 way audio on outgoing
calls.
I am using the default nathelper module config, but I have hacked it a
bit and maybe my changes are causing the problem? I had to use
t_relay_to_tcp for our PSTN gateway and so I had to change the routing
around a little.
Prior to today I was using SerMediaProxy (AG Projects), but I switched
to PortaOnes rtpproxy just to make sure that there wasn't a version
incompatibility problem.
I have also attached my current ser.cfg file.
Anyone have any suggestions? Or should I change the routing in my
ser.cfg file? or would that make a difference?
Any help would be greatly appreciated. Thanks!
Darren Nay - dnay(a)libertyisp.com <mailto:dnay@libertyisp.com>
------------------------------------------------------------------------
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
7:14 p.m.
Hi,
That is not answer.. just advise.
Install ngrep.. then execute ngrep SIP -q and watch
it a best tool to troubleshoot ser and understand sip
sorry for not answering.
On Tue, 2004-02-17 at 12:09, Darren Nay wrote:
I removed it and I still have the same problem. My
onreply_route looks like
this now.
onreply_route[1] {
if (status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
I also tried this..
onreply_route[1] {
fix_nated_contahttp://freshmeat.net/redir/screen/9322/url_tgz/screen-4.0.2.…;
force_rtp_proxy();
}
I must admit that I'm new to all this SIP routing. :(
One more thing of note is that we are getting the following errors in the
syslog.
Feb 17 12:53:15 lvl3 /usr/local/sbin/ser[19711]: ERROR: send_rtpp_command:
can't read reply from a RTP proxy
Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: extract_body:
message body has lenght zero
Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: force_rtp_proxy:
can't extract body from the message
Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: on_reply processing
failed
I just noticed this on my last call attempt.
Any more suggestions. I've tried all kinds of combinations .. I'm stumped..
Thanks!
Darren
----- Original Message -----
From: "Maxim Sobolev" <sobomax(a)portaone.com>
To: "Darren Nay" <dnay(a)libertyisp.com>
Cc: <serusers(a)lists.iptel.org>
Sent: Tuesday, February 17, 2004 2:41 PM
Subject: Re: [Serusers] RTP Proxy Help - One way Audio
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
As I said in my previoud message, most likely
that isflagset(6) test in
onreply_route is causing problems. Remove it completely and try again.
-Maxim
Darren Nay wrote:
> Hey Guys,
>
> I am having issues with One way Audio for outgoing phone calls from my
> SIP phones. It works fine for Incoming, but outgoing audio is not
> working. Also, Outgoing works fine if I put my SIP phone on an Internet
> IP address, but if it's NAT'd then I get the 1 way audio on outgoing
calls.
I am using the default nathelper module config, but I have hacked it a
bit and maybe my changes are causing the problem? I had to use
t_relay_to_tcp for our PSTN gateway and so I had to change the routing
around a little.
Prior to today I was using SerMediaProxy (AG Projects), but I switched
to PortaOnes rtpproxy just to make sure that there wasn't a version
incompatibility problem.
I have also attached my current ser.cfg file.
Anyone have any suggestions? Or should I change the routing in my
ser.cfg file? or would that make a difference?
Any help would be greatly appreciated. Thanks!
Darren Nay - dnay(a)libertyisp.com <mailto:dnay@libertyisp.com>
------------------------------------------------------------------------
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
7:16 p.m.
And of course put:
log(1, "blablabla"); and all if, else, etc in config....
On Tue, 2004-02-17 at 12:09, Darren Nay wrote:
I removed it and I still have the same problem. My
onreply_route looks like
this now.
onreply_route[1] {
if (status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
I also tried this..
onreply_route[1] {
fix_nated_contact();
force_rtp_proxy();
}
I must admit that I'm new to all this SIP routing. :(
One more thing of note is that we are getting the following errors in the
syslog.
Feb 17 12:53:15 lvl3 /usr/local/sbin/ser[19711]: ERROR: send_rtpp_command:
can't read reply from a RTP proxy
Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: extract_body:
message body has lenght zero
Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: force_rtp_proxy:
can't extract body from the message
Feb 17 12:53:29 lvl3 /usr/local/sbin/ser[19716]: ERROR: on_reply processing
failed
I just noticed this on my last call attempt.
Any more suggestions. I've tried all kinds of combinations .. I'm stumped..
Thanks!
Darren
----- Original Message -----
From: "Maxim Sobolev" <sobomax(a)portaone.com>
To: "Darren Nay" <dnay(a)libertyisp.com>
Cc: <serusers(a)lists.iptel.org>
Sent: Tuesday, February 17, 2004 2:41 PM
Subject: Re: [Serusers] RTP Proxy Help - One way Audio
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
As I said in my previoud message, most likely
that isflagset(6) test in
onreply_route is causing problems. Remove it completely and try again.
-Maxim
Darren Nay wrote:
> Hey Guys,
>
> I am having issues with One way Audio for outgoing phone calls from my
> SIP phones. It works fine for Incoming, but outgoing audio is not
> working. Also, Outgoing works fine if I put my SIP phone on an Internet
> IP address, but if it's NAT'd then I get the 1 way audio on outgoing
calls.
I am using the default nathelper module config, but I have hacked it a
bit and maybe my changes are causing the problem? I had to use
t_relay_to_tcp for our PSTN gateway and so I had to change the routing
around a little.
Prior to today I was using SerMediaProxy (AG Projects), but I switched
to PortaOnes rtpproxy just to make sure that there wasn't a version
incompatibility problem.
I have also attached my current ser.cfg file.
Anyone have any suggestions? Or should I change the routing in my
ser.cfg file? or would that make a difference?
Any help would be greatly appreciated. Thanks!
Darren Nay - dnay(a)libertyisp.com <mailto:dnay@libertyisp.com>
------------------------------------------------------------------------
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
7:52 p.m.
New subject: [Serusers] ENUM variable TLD?
Normally I do enum lookups inside my
domain, however, I want to be able to do
an enum lookup at other TLD. Is there a
way to do that? Maybe enum_query("e164.other.com")??
---greg
8:25 p.m.
New subject: [Serusers] ENUM variable TLD?
on development branch there is an updated enum module which
allows you to use private enum trees the way you are asking.
If you need it, there should be no conflicts if you take
devel version of enum and put it in your 8.12 source tree.
-jiri
At 01:52 AM 2/18/2004, Greg Fausak wrote:
Normally I do enum lookups inside my
domain, however, I want to be able to do
an enum lookup at other TLD. Is there a
way to do that? Maybe enum_query("e164.other.com")??
---greg
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
7434
days inactive
7435
days old
6 comments
5 participants
participants (5)
-
Darren Nay -
Greg Fausak -
Jiri Kuthan -
Maxim Sobolev -
Vitaly Nikolaev