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@portaone.com To: "Darren Nay" dnay@libertyisp.com Cc: serusers@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@libertyisp.com mailto:dnay@libertyisp.com

---

---

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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@portaone.com To: "Darren Nay" dnay@libertyisp.com Cc: serusers@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@libertyisp.com mailto:dnay@libertyisp.com

---

---

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

---

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

-- Jiri Kuthan http://iptel.org/~jiri/