My service provider only supports g729 and I tried what you have mentioned here but same thing is happening here. Is there any why that I can see which codec my service provider is pushing when I'm receiving call on my asterisk server. When call comes comes to my server and then I type show g729 it shows 0/0 out of 15 lic.
<br><br>thanks<br>arun<br><br><div><span class="gmail_quote">On 2/21/07, <b class="gmail_sendername">Mike Lynchfield</b> &lt;<a href="mailto:theclubvoip@gmail.com">theclubvoip@gmail.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Well, could be the fact provider not pushing as g729 or someting else.<br><br>Can you set debug 999 and set verbose 999<br>then redump that ? you are missing the before the answer part also..<br><br>Also try G711 first then work your way to other codecs
<div><span class="e" id="q_110e07df1534bd74_1"><br><br><br>On 2/20/07, <b class="gmail_sendername">Rajeev Natarajan</b> &lt;<a href="mailto:twogigbox@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
twogigbox@gmail.com</a>&gt; wrote:<div><span class="gmail_quote"></span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Am working with Arun on this project - here&#39;s a longer description of the problem:<br><br>We&#39;ve been fighting with our service provider on this issue - we seem to be getting a BYE just after we receive an ACK. They claim that it is an asterisk issue! The service provider provides only IP based authentication for inbound.
<br><br>We have used username-password based authentication with the same setup with *no problems*&nbsp; whatsoever! <br><br>If we configure an Audiocodes MEdia gateway to receive the calls, there is no issue - so there&#39;s something that asterisk is doing? or asterisk-Provider gateway combo?
<br><br>In our efforts to mask IP, I have used PROVIDER-IP for the IP of my service provider (host) and AsteriskIP to indicate my asterisk server<br><br>sip.conf<br>[PROVIDER]<br>type=peer<br>disallow=all<br>allow=g729<br>


context=default<br>host=xxxx<br>fromuser=y.y.y.y<br>port=5060<br>insecure=very<br>canreinvite=no
<br>nat=yes<br>qualify=yes<br><br>CLI output:<br><br><span></span>&nbsp;&nbsp; -- Executing Answer(&quot;SIP/PROVIDER-IP-b7a076a8&quot;, &quot;&quot;) in new stack<br>We&#39;re at <a href="http://124.7.195.102" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">


124.7.195.102</a> port 47698<br>Adding codec 0x100 (g729) to SDP<br>Reliably Transmitting (NAT) to PROVIDER-IP:5060:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP
<br>From: &lt;sip:919444072925@PROVIDER-IP&gt;;tag=3380976385-794612<br>To: &lt;sip:8009422419@192.168.11.2:5060&gt;;tag=as52d36855<br>Call-ID: <a href="mailto:45018316-3380976385-794578@nextone-msw.mydomain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">

45018316-3380976385-794578@nextone-msw.mydomain.com
</a><br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: &lt;sip:8009422419@&#39;AsteriskIP&#39;&gt;<br>Content-Type: application/sdp<br>Content-Length: 183
<br><br>v=0<br>o=root 2172 2172 IN IP4 AsteriskIP<br>s=session<br>c=IN IP4 AsteriskIP<br>t=0 0<br>m=audio 47698 RTP/AVP 18<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=silenceSupp:off - - - -<br><br>---&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
<br>&nbsp;-- Executing Playback(&quot;SIP/PROVIDER-IP-b7a076a8&quot;, &quot;park&quot;) in new stack<br>&nbsp;&nbsp;&nbsp; -- Playing &#39;park&#39; (language &#39;en&#39;)<br>AstSQL*CLI&gt;<br>&lt;-- SIP read from PROVIDER-IP:5060:<br>ACK 
sip:8009422419@AsteriskIP
 SIP/2.0<br>Max-Forwards: 5<br>To: &lt;sip:8009422419@192.168.11.2:5060&gt;;tag=as52d36855<br>From: &lt;sip:919444072925@PROVIDER-IP&gt;;tag=3380976385-794612<br>Contact: &lt;sip:919444072925@PROVIDER-IP:5060&gt;<br>Call-ID: 
<a href="mailto:45018316-3380976385-794578@nextone-msw.mydomain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">45018316-3380976385-794578@nextone-msw.mydomain.com</a><br>CSeq: 1 ACK<br>Via: SIP/2.0/UDP 
<a href="http://221.135.102.100:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">221.135.102.100:5060</a>
;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422<br>Content-Length: 0<br><br><br>--- (9 headers 0 lines) ---<br>AstSQL*CLI&gt;<br>&lt;-- SIP read from PROVIDER-IP:5060:<br>BYE sip:8009422419@AsteriskIP SIP/2.0<br>Max-Forwards: 5
<br>To: &lt;sip:8009422419@192.168.11.2:5060&gt;;tag=as52d36855<br>From: &lt;sip:919444072925@PROVIDER-IP&gt;;tag=3380976385-794612<br>Contact: &lt;sip:919444072925@PROVIDER-IP:5060&gt;<br>Call-ID: <a href="mailto:45018316-3380976385-794578@nextone-msw.mydomain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">


45018316-3380976385-794578@nextone-msw.mydomain.com</a><br>CSeq: 2 BYE<br>Via: SIP/2.0/UDP <a href="http://221.135.102.100:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">221.135.102.100:5060
</a>;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f<br>Content-Length: 0
<br><br><br>--- (9 headers 0 lines) ---<br>Sending to PROVIDER-IP : 5060 (NAT)<br>Transmitting (NAT) to PROVIDER-IP:5060:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP
<br>From: &lt;sip:919444072925@PROVIDER-IP&gt;;tag=3380976385-794612<br>To: &lt;sip:8009422419@192.168.11.2:5060&gt;;tag=as52d36855<br>Call-ID: <a href="mailto:45018316-3380976385-794578@nextone-msw.mydomain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">

45018316-3380976385-794578@nextone-msw.mydomain.com
</a><br>CSeq: 2 BYE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: &lt;sip:8009422419@AsteriskIP&gt;<br>Content-Length: 0<br><br>----------------------------------------------------------------------------------------------------------------------------------------------------------------------------
<br>The following is an ngrep of the traffic for an inbound call - &#39;U&#39; marks the begin of the packet grabbed. <br><br><br>U &lt;PROVIDER-IP&gt;:5060 -&gt; &lt;AsteriskIP&gt;:5060
<br>&nbsp; INVITE sip:800942xxxx@&lt;AsteriskIP&gt; SIP/2.0..Max-Forwards: 5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: &lt;sip:800942xxxx@192.168.11.2:5060&gt;..From: &lt;sip:&lt;PROVIDER-IP&gt;&gt;;tag=3380960452-790279..Co ntact: &lt;sip:&lt;PROVIDER-IP&gt;:5060&gt;..Remote-Party-Id: &lt;sip:&lt;PROVIDER-IP&gt;&gt;;party=calling;screen=no;privacy =off..Call-ID: 
44786336-3380960452-790235@nextone-msw.mydomain.com..CSeq: 1 INVITE..Via: SIP/2.0/UDP 221. 135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events: telephone-event..Content-T ype: application/sdp..Content-Length: 206....v=0..o=nextone-msw1 1774 4816 IN IP4 &lt;PROVIDER-IP&gt;..s=sip call..c=IN IP4 &lt;PROV-IP-2&gt;..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19 CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000..
<br><br><br>#<br>U &lt;AsteriskIP&gt;:5060 -&gt; &lt;PROVIDER-IP&gt;:5060<br>&nbsp; SIP/2.0 100 Trying..Via: SIP/2.0/UDP &lt;PROVIDER-IP&gt;:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4; received=&lt;PROVIDER-IP&gt;..From: &lt;sip:&lt;PROVIDER-IP&gt;&gt;;tag=3380960452-790279..To: &lt;
<a href="mailto:sip:800942xxxx@192.168" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:800942xxxx@192.168</a>. 11.2:5060&gt;..Call-ID: 44786336-3380960452-790235@nextone-msw.mydomain.com..CSeq: 1 INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: &lt;
sip:800942xxxx@AsteriskIP&gt;..Content-Length: 0.... <br><br><br>#<br>U &lt;AsteriskIP&gt;:5060 -&gt; &lt;PROVIDER-IP&gt;:5060<br>&nbsp; SIP/2.0 180 Ringing..Via: SIP/2.0/UDP &lt;PROVIDER-IP&gt;:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4 ;received=&lt;PROVIDER-IP&gt;..From: &lt;sip:&lt;PROVIDER-IP&gt;&gt;;tag=3380960452-790279..To: &lt;
<a href="mailto:sip:800942xxxx@192.168" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:800942xxxx@192.168</a> .11.2:5060&gt;;tag=as78bcde29..Call-ID: <a href="mailto:44786336-3380960452-790235@nextone-msw.mydomain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">

44786336-3380960452-790235@nextone-msw.mydomain.com</a>
..CSeq: 1 INVITE. .User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: &lt; sip:800942xxxx@&lt;AsteriskIP&gt;&gt;..Content-Length: 0....
<br><br><br><br>#<br>U &lt;AsteriskIP&gt;:5060 -&gt; &lt;PROVIDER-IP&gt;:5060<br>&nbsp; SIP/2.0 200 OK..Via: SIP/2.0/UDP &lt;PROVIDER-IP&gt;:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4;rece ived=&lt;PROVIDER-IP&gt;..From: &lt;sip:&lt;PROVIDER-IP&gt;&gt;;tag=3380960452-790279..To: &lt;
<a href="mailto:sip:800942xxxx@192.168.11.2" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:800942xxxx@192.168.11.2</a> :5060&gt;;tag=as78bcde29..Call-ID: <a href="mailto:44786336-3380960452-790235@nextone-msw.mydomain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">

44786336-3380960452-790235@nextone-msw.mydomain.com</a>
..CSeq: 1 INVITE..User -Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: &lt;sip: 800942xxxx@&lt;AsteriskIP&gt;&gt;..Content-Type: application/sdp..Content-Length: 182....v=0..o=root 2156 2156 IN IP4 &lt;AsteriskIP&gt;..s=session..c=IN IP4 &lt;Asterisk&gt;..t=0 0..m=audio 5676 RTP/AVP 18..a=rtpmap:18 G729/80 00..a=fmtp:18 annexb=no..a=silenceSupp:off - - - -..
<br><br><br><br>#<br>U &lt;PROVIDER-IP&gt;:5060 -&gt; &lt;AsteriskIP&gt;:5060<br>&nbsp; ACK sip:800942xxxx@&lt;AsteriskIP&gt; SIP/2.0..Max-Forwards: 5..To: &lt;sip:800942xxxx@192.168.11.2:5060&gt;;tag=as7 8bcde29..From: &lt;sip:&lt;PROVIDER-IP&gt;&gt;;tag=3380960452-790279..Contact: &lt;sip:&lt;PROVIDER-IP&gt;:5060&gt;..Call-ID: 
44786336-3380960452-790235@nextone-msw.mydomain.com..CSeq: 1 ACK..Via: SIP/2.0/UDP &lt;PROVIDER-IP&gt;:5060; branch=z9hG4bK74ac10cb8c5d89375bf77d4aaa15fcea..Content-Length: 0....<br><br><br>#<br>U &lt;PROVIDER-IP&gt;:5060 -&gt; &lt;AsteriskIP&gt;:5060
<br>&nbsp; BYE sip:800942xxxx@&lt;AsteriskIP&gt; SIP/2.0..Max-Forwards: 5..To: &lt;sip:800942xxxx@192.168.11.2:5060&gt;;tag=as7 8bcde29..From: &lt;sip:&lt;PROVIDER-IP&gt;&gt;;tag=3380960452-790279..Contact: &lt;sip:&lt;PROVIDER-IP&gt;:5060&gt;..Call-ID: 
44786336-3380960452-790235@nextone-msw.mydomain.com..CSeq: 2 BYE..Via: SIP/2.0/UDP &lt;PROVIDER-IP&gt;:5060; branch=z9hG4bK610e4f29ad9631a0065d4b23dc6c8c27..Content-Length: 0....<br><br><br>#<br>U &lt;AsteriskIP&gt;:5060 -&gt; &lt;PROVIDER-IP&gt;:5060
<br>&nbsp; SIP/2.0 200 OK..Via: SIP/2.0/UDP &lt;PROVIDER-IP&gt;:5060;branch=z9hG4bK610e4f29ad9631a0065d4b23dc6c8c27;rece ived=&lt;PROVIDER-IP&gt;..From: &lt;sip:&lt;PROVIDER-IP&gt;&gt;;tag=3380960452-790279..To: &lt;<a href="mailto:sip:800942xxxx@192.168.11.2" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">



sip:800942xxxx@192.168.11.2</a> :5060&gt;;tag=as78bcde29..Call-ID: 44786336-3380960452-790235@nextone-msw.mydomain.com..CSeq: 2 BYE..User-Ag ent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: &lt;sip:8009 422419@&lt;AsteriskIP&gt;&gt;..Content-Length: 0....
<br><br>--------------------------------------------------------------------------------------------------------------------------------<br><br>Any help appreciated<br>Thanks!<br><span>Rajeev</span><div><span>
<br><br><div><span class="gmail_quote">
On 2/20/07, <b class="gmail_sendername">Arun Kumar</b> &lt;<a href="mailto:arunvoip@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
arunvoip@gmail.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Instead of forwarding to IAX softphone if I&#39;ll play some music same thing is happening in this case also.
<div><span><br><br><div><span class="gmail_quote">On 2/20/07, <b class="gmail_sendername">Mark Phillips</b> &lt;<a href="mailto:g7ltt@g7ltt.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">

g7ltt@g7ltt.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Without seeing your config files my guess would be that this is
<br>something to do with a bad codec negotiation.<br><br>I&#39;d bet that your IAX phone is using ulaw and your DID provider is using<br>something else like G729.<br><br>Mark<br><br>On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote:
<br>&gt; HI<br>&gt;<br>&gt; I&#39;ve configred an Incoming DID in my asterisk and when I call from<br>&gt; outside I see call is coming to my Asterisk server and then from<br>&gt; asterisk it rings on a particulat exten but when I pickup the call the
<br>&gt; call get disconnect immediate and on the other end it keep trying<br>&gt; (ringing).<br>&gt;<br>&gt; here is my exten.conf:<br>&gt;<br>&gt; exten =&gt; _80.,1,Answer<br>&gt; exten =&gt; _80.,2,Dial(IAX2/2001)<br>





&gt;<br>&gt; did starts with 80 and any call comes for my number they are sending<br>&gt; to my asterisk IP.<br>&gt;<br>&gt; thanks<br>&gt;<br>&gt; _______________________________________________<br>&gt; --Bandwidth and Colocation provided by 
<a href="http://Easynews.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Easynews.com</a> --<br>&gt;<br>&gt; asterisk-users mailing list<br>&gt; To UNSUBSCRIBE or update options visit:<br>&gt;&nbsp;&nbsp;&nbsp;&nbsp;
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://lists.digium.com/mailman/listinfo/asterisk-users
</a><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Easynews.com</a> --
<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:
<br>&nbsp;&nbsp; <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote>




</div><br>
</span></div><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Easynews.com
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br></blockquote></div><br>
</span></div><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Easynews.com
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br></blockquote></div><br><br clear="all"><br></span></div><span class="sg">-- <br>Mike<br>Sales Manager<br><a href="http://www.voicemeup.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
http://www.voicemeup.com</a><br>Making it happen
<br>1.877.807.VOIP (8647)<br>1.514.312.7030
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