Brad,
Great. I suspect my difficulty may be related to a NAT/PAT
configuration. The SIP/SDP negotiations go fine, but there could be
problems setting up the incoming RTP session. I wonder how SIP/SDP
decides on what port(s) to listen for the incoming RTP connection.
Gregg
On Sun, 2003-03-30 at 1
I'm not behind a NAT, but of course behind a firewall (duh). I was even
thinking to myself "this is very much like what happens with IAX when
there is a firewall issue". So having taken care of that, it works great
with the same sip.conf settings you have below, and both directions can
hear each ot
brad,
Just to make sure you understand the settings, not using the prefix
tells iconnect to use uncompressed codecs. Using sets iconnect into
compressed codec mode.
I am experience that same problem as you when I try to use the
uncompressed mode. I connect, but cannot hear the other par
I've tried these settings and I still find that I cannot hear the called
party. I've also tried what feels like every allow/disallow combination
with and without a prefix and I either get 488 errors, using one
format when the capability is another errors, or completed calls where I
can't h
>Is there a Record-Route header in the response that comes back from
>iconnect?
In the 480, not that I can see.
In the 408, I'm not sure as I didn't have SIP debugging enabled (and I don't
have anyone internationally to ring right now :-)).
-- Luke
--
Luke Howard | PADL Software Pty Ltd | www.
Mark,
I believe there is: Here is the exchange using sip debug.
Gregg
---
bigcat*CLI> sip debug
SIP Debugging Enabled
-- Executing Dial("Phone/phone0",
"SIP/[EMAIL PROTECTED]") in new stack
Interface is eth0
IP Address is 192.168.4.3
We're at 192.168.
Is there a Record-Route header in the response that comes back from
iconnect?
Mark
On Sun, 23 Mar 2003, Luke Howard wrote:
>
> >> Or maybe we should send an ACK to them -- I need to read the SIP RFC...
> >>
> >
> >Tried that, doesn't work.
> >
> >I should add that in my config I'm totally behind
Gregg,
>1) the prefix is not a toggle. It tells iconnects SIP gateway to
>use compressed codecs. The choices are gsm, g723.1, g729.
I figured as much. I'm sticking with G.711 as GSM sounds horrible (at least
with the snom phones) and the other codecs you mention are patent
encumbered.
>I
Luke,
here's some information I got back from iconnect:
1) the prefix is not a toggle. It tells iconnects SIP gateway to
use compressed codecs. The choices are gsm, g723.1, g729.
If you don't use , the gateway will tried to use PCMu/8000 (ulaw?)
or PCMa/8000 (alaw?).
I can get the gate
FWIW here's the patch I'm using to ignore 480s:
[EMAIL PROTECTED]/monk[16]% cvs diff -u channels/chan_sip.c
Index: channels/chan_sip.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.5
diff -u -r1.5 ch
>> Or maybe we should send an ACK to them -- I need to read the SIP RFC...
>>
>
>Tried that, doesn't work.
>
>I should add that in my config I'm totally behind NAT, both asterisk and
>an ATA186 that talks to it.
>
>So that may be confounding me in terms of what I'm seeing.
I do see the same pro
>I should add that in my config I'm totally behind NAT, both asterisk and
>an ATA186 that talks to it.
Hmm, both our SIP phones and Asterisk are on visible IPs.
-- Luke
--
Luke Howard | PADL Software Pty Ltd | www.padl.com
___
Asterisk-Users mailing
Luke Howard wrote:>>>I tried the grotesque hack of making
handle_response() ignore 480
errors, which *seems* to work. Hmm.
I tried that, and at least for me it has a number of subtle side effects:
Or maybe we should send an ACK to them -- I need to read the SIP RFC...
Tried that, doesn't work.
Moreover, if anyone has a packet trace of iConnectHere's SIP client
making a call (which presumably does work), then please send it
along... it would be interesting to see whether Asterisk, at fault
or not, can be made to work around this properly.
-- Luke
--
Luke Howard | PADL Software Pty Ltd
>> I tried the grotesque hack of making handle_response() ignore 480
>> errors, which *seems* to work. Hmm.
>>
>
>I tried that, and at least for me it has a number of subtle side effects:
Or maybe we should send an ACK to them -- I need to read the SIP RFC...
-- Luke
--
Luke Howard | PADL Soft
>> I tried the grotesque hack of making handle_response() ignore 480
>> errors, which *seems* to work. Hmm.
>>
>
>I tried that, and at least for me it has a number of subtle side effects:
>
>1. Calls all cut off after just a few minutes
Well, we could enable the hack only on outgoing calls (whic
Luke Howard wrote:>>I remember at some point getting 488 media errors if
I didn't enable
gsm.
As I mentioned, I'm getting 480 Temporarily not available, not
488 media errors.
I tried the grotesque hack of making handle_response() ignore 480
errors, which *seems* to work. Hmm.
I tried that, and a
>I remember at some point getting 488 media errors if I didn't enable
>gsm.
As I mentioned, I'm getting 480 Temporarily not available, not
488 media errors.
I tried the grotesque hack of making handle_response() ignore 480
errors, which *seems* to work. Hmm.
-- Luke
--
Luke Howard | PADL Softw
>GSM works but the voice quality is absolutely terrible. This is the
>case with or without the prefix. (Did anyone ever figure out
>whether is a toggle?)
One thing I didn't realise until reading the new documentation is that
the codec list is in order of preference. So, if there's an a
>I remember at some point getting 488 media errors if I didn't enable
>gsm.
GSM works but the voice quality is absolutely terrible. This is the
case with or without the prefix. (Did anyone ever figure out
whether is a toggle?)
>disallow=g723.1
>allow=gsm
>allow=ulaw
>allow=alaw
>allow
I remember at some point getting 488 media errors if I didn't enable
gsm.
Here are my sip.conf and extensions.conf entries. They work for calls
out to iconnect:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context=iconnect ;
Luke,
Try putting the prefix of before your phone number. It changes the
codec expected by iconnect.
Gregg
On Thu, 2003-03-20 at 18:25, Luke Howard wrote:
> I've found the same.
>
> If I make an outgoing call (snom 200 handset), I get about 5 seconds
> of audio and then it drops out (very
Just an FYI but I'm seeing the same thing using ata-186's
John
-
NetRom Internet Services973-208-1339 voice
[EMAIL PROTECTED] 973-208-0942 fax
http://www.netrom.com
I've found the same.
If I make an outgoing call (snom 200 handset), I get about 5 seconds
of audio and then it drops out (very occasionally it does work).
Incoming calls appear to work, though.
-- Executing Goto("SIP/515-Office-143b", "iconnecthere-ulaw|91800XXX|1") in new
stack
-- Got
, March 04, 2003 1:09 AM
Subject: Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
> I get these errors (480 "Temporarily...") when I try to use my
> iconnect account quickly after hanging up on a previous session.
> They have some sort of contention locking
I am going to have to find a fix for this problem or I'm going to have
to quit using iconnect.
About one call in 10 or so, iconnect's gateway gives me an error
(console output appended below).
So upon receiving the error, which as a 4XX error means, "Fatal,"
asterisk gives up and drops the cal
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