<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
While the call is progressing<br>
<br>
sip show channels<br>
Peer User/ANR Call ID Seq (Tx/Rx) Format
Hold Last Message<br>
82.101.62.XX 0475769XXX 14151-EX-29 00101/703757593 0x4
(ulaw) No Rx: ACK<br>
82.101.62.XX 0475769XXX 6ec6f62d57d 00103/00000 0x0 (nothing)
No<br>
<br>
Codec=Ulaw, still no "ringing"<br>
<br>
Fons van der Beek schreef:
<blockquote cite="mid:47BFE733.3000505@84-it.com" type="cite">
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
I guess we are back to the fundamental problem: "no asterisk generated
sounds on the external call"<br>
<br>
After implementing the described test for indications.conf<br>
The CLI outputted:<br>
-- Executing [s@default:1] Answer("SIP/0475769XXX-095a8488", "") in
new stack<br>
-- Executing [s@default:2] PlayTones("SIP/0475769XXX-095a8488",
"ring") in new stack<br>
-- Executing [s@default:3] Wait("SIP/0475769XXX-095a8488", "30") in
new stack<br>
<br>
This looks OK, but there is no sound to be heard on the other end.<br>
<br>
Sip show peers for the other end shows:<br>
* Name : sip.xs4all.nl<br>
Secret : <Set><br>
MD5Secret : <Not set><br>
Context : default<br>
Subscr.Cont. : default<br>
Language : en<br>
AMA flags : Unknown<br>
Transfer mode: open<br>
CallingPres : Presentation Allowed, Not Screened<br>
FromUser : 0475769XXX<br>
FromDomain : sip.xs4all.nl<br>
Callgroup :<br>
Pickupgroup :<br>
Mailbox :<br>
VM Extension : asterisk<br>
LastMsgsSent : 32767/65535<br>
Call limit : 0<br>
Dynamic : No<br>
Callerid : "" <><br>
MaxCallBR : 384 kbps<br>
Expire : -1<br>
Insecure : port,invite<br>
Nat : RFC3581<br>
ACL : No<br>
T38 pt UDPTL : No<br>
CanReinvite : No<br>
PromiscRedir : No<br>
User=Phone : No<br>
Video Support: Yes<br>
Trust RPID : No<br>
Send RPID : No<br>
Subscriptions: Yes<br>
Overlap dial : No<br>
DTMFmode : auto<br>
LastMsg : 0<br>
ToHost : sip.xs4all.nl<br>
Addr->IP : 82.101.XX.XX Port 5060<br>
Defaddr->IP : 0.0.0.0 Port 0<br>
Def. Username: 0475769XXX<br>
SIP Options : (none)<br>
Codecs : 0x104 (ulaw|g729)<br>
Codec Order : (ulaw:20,g729:20)<br>
Auto-Framing: No<br>
Status : Unmonitored<br>
Useragent :<br>
Reg. Contact :<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
Trevor Peirce schreef:
<blockquote cite="mid:47BFE1F2.1020700@digitalcon.ca" type="cite">
<pre wrap="">Fons van der Beek wrote:
</pre>
<blockquote type="cite">
<pre wrap="">I've overwritten the indications.conf with the one from the
sourcecode, stil no luck
Perhaps somebody knows what the correct value for indications.conf is
when using the dutch xs4all as sip carrier??
</pre>
</blockquote>
<pre wrap=""><!---->
A simple way for you to test your indications.conf as far as the ringing
goes is something like this:
exten => s,1,Answer
exten => s,n,PlayTones(ring)
exten => s,n,Wait(30)
exten => s,n,Hangup
That should pick up the line and then play your locale's ring tone for
30 seconds before hanging up. If you hear ringing then indications.conf
is fine, otherwise you have confirmed that there is a problem somewhere.
This will have nothing to do with your carrier as the sounds are
generated by asterisk itself as audio (as opposed to any kind of
carrier-specific signaling).
Trevor
Real CNAM data for incoming Caller ID @ <a moz-do-not-send="true"
class="moz-txt-link-abbreviated" href="http://www.cnam.info">www.cnam.info</a>
_______________________________________________
-- Bandwidth and Colocation Provided by <a moz-do-not-send="true"
class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a> --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<br>
<pre wrap="">
<hr size="4" width="90%">
_______________________________________________
-- Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a> --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></pre>
</blockquote>
<br>
</body>
</html>