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> <<a href="mailto:theclubvoip@gmail.com">theclubvoip@gmail.com</a>> 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> <<a href="mailto:twogigbox@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
twogigbox@gmail.com</a>> 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's a longer description of the problem:<br><br>We'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* whatsoever! <br><br>If we configure an Audiocodes MEdia gateway to receive the calls, there is no issue - so there'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> -- Executing Answer("SIP/PROVIDER-IP-b7a076a8", "") in new stack<br>We'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: <sip:919444072925@PROVIDER-IP>;tag=3380976385-794612<br>To: <sip:8009422419@192.168.11.2:5060>;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: <sip:8009422419@'AsteriskIP'><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>---
<br> -- Executing Playback("SIP/PROVIDER-IP-b7a076a8", "park") in new stack<br> -- Playing 'park' (language 'en')<br>AstSQL*CLI><br><-- SIP read from PROVIDER-IP:5060:<br>ACK
sip:8009422419@AsteriskIP
SIP/2.0<br>Max-Forwards: 5<br>To: <sip:8009422419@192.168.11.2:5060>;tag=as52d36855<br>From: <sip:919444072925@PROVIDER-IP>;tag=3380976385-794612<br>Contact: <sip:919444072925@PROVIDER-IP:5060><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><br><-- SIP read from PROVIDER-IP:5060:<br>BYE sip:8009422419@AsteriskIP SIP/2.0<br>Max-Forwards: 5
<br>To: <sip:8009422419@192.168.11.2:5060>;tag=as52d36855<br>From: <sip:919444072925@PROVIDER-IP>;tag=3380976385-794612<br>Contact: <sip:919444072925@PROVIDER-IP:5060><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: <sip:919444072925@PROVIDER-IP>;tag=3380976385-794612<br>To: <sip:8009422419@192.168.11.2:5060>;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: <sip:8009422419@AsteriskIP><br>Content-Length: 0<br><br>----------------------------------------------------------------------------------------------------------------------------------------------------------------------------
<br>The following is an ngrep of the traffic for an inbound call - 'U' marks the begin of the packet grabbed. <br><br><br>U <PROVIDER-IP>:5060 -> <AsteriskIP>:5060
<br> INVITE sip:800942xxxx@<AsteriskIP> SIP/2.0..Max-Forwards: 5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: <sip:800942xxxx@192.168.11.2:5060>..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..Co ntact: <sip:<PROVIDER-IP>:5060>..Remote-Party-Id: <sip:<PROVIDER-IP>>;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 <PROVIDER-IP>..s=sip call..c=IN IP4 <PROV-IP-2>..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 <AsteriskIP>:5060 -> <PROVIDER-IP>:5060<br> SIP/2.0 100 Trying..Via: SIP/2.0/UDP <PROVIDER-IP>:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4; received=<PROVIDER-IP>..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..To: <
<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>..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: <
sip:800942xxxx@AsteriskIP>..Content-Length: 0.... <br><br><br>#<br>U <AsteriskIP>:5060 -> <PROVIDER-IP>:5060<br> SIP/2.0 180 Ringing..Via: SIP/2.0/UDP <PROVIDER-IP>:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4 ;received=<PROVIDER-IP>..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..To: <
<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>;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: < sip:800942xxxx@<AsteriskIP>>..Content-Length: 0....
<br><br><br><br>#<br>U <AsteriskIP>:5060 -> <PROVIDER-IP>:5060<br> SIP/2.0 200 OK..Via: SIP/2.0/UDP <PROVIDER-IP>:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4;rece ived=<PROVIDER-IP>..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..To: <
<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>;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: <sip: 800942xxxx@<AsteriskIP>>..Content-Type: application/sdp..Content-Length: 182....v=0..o=root 2156 2156 IN IP4 <AsteriskIP>..s=session..c=IN IP4 <Asterisk>..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 <PROVIDER-IP>:5060 -> <AsteriskIP>:5060<br> ACK sip:800942xxxx@<AsteriskIP> SIP/2.0..Max-Forwards: 5..To: <sip:800942xxxx@192.168.11.2:5060>;tag=as7 8bcde29..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..Contact: <sip:<PROVIDER-IP>:5060>..Call-ID:
44786336-3380960452-790235@nextone-msw.mydomain.com..CSeq: 1 ACK..Via: SIP/2.0/UDP <PROVIDER-IP>:5060; branch=z9hG4bK74ac10cb8c5d89375bf77d4aaa15fcea..Content-Length: 0....<br><br><br>#<br>U <PROVIDER-IP>:5060 -> <AsteriskIP>:5060
<br> BYE sip:800942xxxx@<AsteriskIP> SIP/2.0..Max-Forwards: 5..To: <sip:800942xxxx@192.168.11.2:5060>;tag=as7 8bcde29..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..Contact: <sip:<PROVIDER-IP>:5060>..Call-ID:
44786336-3380960452-790235@nextone-msw.mydomain.com..CSeq: 2 BYE..Via: SIP/2.0/UDP <PROVIDER-IP>:5060; branch=z9hG4bK610e4f29ad9631a0065d4b23dc6c8c27..Content-Length: 0....<br><br><br>#<br>U <AsteriskIP>:5060 -> <PROVIDER-IP>:5060
<br> SIP/2.0 200 OK..Via: SIP/2.0/UDP <PROVIDER-IP>:5060;branch=z9hG4bK610e4f29ad9631a0065d4b23dc6c8c27;rece ived=<PROVIDER-IP>..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..To: <<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>;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: <sip:8009 422419@<AsteriskIP>>..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> <<a href="mailto:arunvoip@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
arunvoip@gmail.com</a>> 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'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> <<a href="mailto:g7ltt@g7ltt.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
g7ltt@g7ltt.com</a>> 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'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>> HI<br>><br>> I've configred an Incoming DID in my asterisk and when I call from<br>> outside I see call is coming to my Asterisk server and then from<br>> asterisk it rings on a particulat exten but when I pickup the call the
<br>> call get disconnect immediate and on the other end it keep trying<br>> (ringing).<br>><br>> here is my exten.conf:<br>><br>> exten => _80.,1,Answer<br>> exten => _80.,2,Dial(IAX2/2001)<br>
><br>> did starts with 80 and any call comes for my number they are sending<br>> to my asterisk IP.<br>><br>> thanks<br>><br>> _______________________________________________<br>> --Bandwidth and Colocation provided by
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