Here's the cisco AS5300 settings from our provider<div><br></div><div><div>codec preference 1 g729r8</div><div>codec preference 2 g729br8</div><div>codec preference 3 g723r53</div><div>codec preference 4 g723r63</div><div>
codec preference 5 g723ar53</div><div>codec preference 6 g723ar63</div><br><div class="gmail_quote">On Mon, Jan 9, 2012 at 5:18 PM, Roi Stork <span dir="ltr"><<a href="mailto:roi.stork@gmail.com">roi.stork@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>Hi Alex, here's the config and the sip debug output.</div><div><br></div><div>Guide:</div><div>xxx.xxx.xxx.xxx - the provider's cisco as5300 trunk ip add</div>
<div>yyy.yy.yy.yy - our asterisk 1.6.2.14 server</div>
<div><br></div><div>sip config:</div><div><br></div><div>type=peer</div><div>disallow=all</div><div>allow=g729</div><div>host=xxx.xxx.xxx.xxx</div><div>fromdomain=xxx.xxx.xxx.xxx</div><div>dtmfmode=rfc2833</div><div>nat=no</div>
<div>canreinvite=yes</div><div>context=from-trunk-sip-iaccess</div><div><br></div><div>sip debug:</div><div><div>v=0</div><div>o=root 249777024 249777024 IN IP4 yyy.yy.yy.yy</div><div>s=Asterisk PBX 1.6.2.14</div><div>c=IN IP4 yyy.yy.yy.yy</div>
<div>t=0 0</div><div>m=audio 13702 RTP/AVP 0 8 3 18 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:3 GSM/8000</div><div>a=rtpmap:18 G729/8000</div><div>a=fmtp:18 annexb=no</div><div>
a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=sendrecv</div><div><br></div><div>---</div><div><br></div><div><--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 ---></div><div>SIP/2.0 100 Trying</div><div>Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport</div>
<div>From: "6598715968" <sip:6598715968@yyy.yy.yy.yy>;tag=as6e218907</div><div>To: <sip:34546598715968@xxx.xxx.xxx.xxx></div><div>Date: Fri, 06 Jan 2012 04:51:39 GMT</div><div>Call-ID: 7e54da423b0e6e457475ab17694e5165@yyy.yy.yy.yy</div>
<div>Server: Cisco-SIPGateway/IOS-12.x</div><div>CSeq: 102 INVITE</div><div>Allow-Events: telephone-event</div><div>Content-Length: 0</div><div><br></div><div><br></div><div><-------------></div><div>--- (10 headers 0 lines) ---</div>
<div>Retransmitting #3 (no NAT) to zzz.zz.zz.zz:5060:</div><div>OPTIONS sip:zzz.zz.zz.zz SIP/2.0</div><div>Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport</div><div>Max-Forwards: 70</div><div>From: "Unknown" <sip:Unknown@yyy.yy.yy.yy>;tag=as5c8e3f97</div>
<div>To: <sip:zzz.zz.zz.zz></div><div>Contact: <sip:Unknown@yyy.yy.yy.yy></div><div>Call-ID: 7bdb028f789e3afa58b22db472d9dfb5@yyy.yy.yy.yy</div><div>CSeq: 102 OPTIONS</div><div>User-Agent: Asterisk PBX 1.6.2.14</div>
<div>Date: Fri, 06 Jan 2012 06:23:00 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</div><div>Supported: replaces, timer</div><div>Content-Length: 0</div><div><br></div><div><br></div>
<div>---</div><div><br></div><div><--- SIP read from UDP:<a href="http://69.90.209.57:5060" target="_blank">69.90.209.57:5060</a> ---></div><div><br></div><div><-------------></div><div>Retransmitting #4 (no NAT) to zzz.zz.zz.zz:5060:</div>
<div>OPTIONS sip:zzz.zz.zz.zz SIP/2.0</div><div>Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK78129860;rport</div><div>Max-Forwards: 70</div><div>From: "Unknown" <sip:Unknown@yyy.yy.yy.yy>;tag=as5c8e3f97</div>
<div>To: <sip:zzz.zz.zz.zz></div><div>Contact: <sip:Unknown@yyy.yy.yy.yy></div><div>Call-ID: 7bdb028f789e3afa58b22db472d9dfb5@yyy.yy.yy.yy</div><div>CSeq: 102 OPTIONS</div><div>User-Agent: Asterisk PBX 1.6.2.14</div>
<div>Date: Fri, 06 Jan 2012 06:23:00 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</div><div>Supported: replaces, timer</div><div>Content-Length: 0</div><div><br></div><div><br></div>
<div>---</div><div>Really destroying SIP dialog '7bdb028f789e3afa58b22db472d9dfb5@yyy.yy.yy.yy' Method: OPTIONS</div><div><br></div><div><--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 ---></div><div>SIP/2.0 183 Session Progress</div>
<div>Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport</div><div>From: "6598715968" <sip:6598715968@yyy.yy.yy.yy>;tag=as6e218907</div><div>To: <sip:34546598715968@xxx.xxx.xxx.xxx>;tag=B6534850-EC6</div>
<div>Date: Fri, 06 Jan 2012 04:51:39 GMT</div><div>Call-ID: 7e54da423b0e6e457475ab17694e5165@yyy.yy.yy.yy</div><div>Server: Cisco-SIPGateway/IOS-12.x</div><div>CSeq: 102 INVITE</div><div>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</div>
<div>Allow-Events: telephone-event</div><div>Remote-Party-ID: "6598715968" </div><div><br></div><div><sip:1234#6598715968@xxx.xxx.xxx.xxx>;party=called;screen=no;privacy=off</div><div>Contact: <sip:34546598715968@xxx.xxx.xxx.xxx:5060></div>
<div>Content-Type: application/sdp</div><div>Content-Disposition: session;handling=required</div><div>Content-Length: 223</div><div><br></div><div>v=0</div><div>o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx</div>
<div>s=SIP Call</div><div>c=IN IP4 xxx.xxx.xxx.xxx</div><div>t=0 0</div><div>m=audio 18132 RTP/AVP 18</div><div>c=IN IP4 xxx.xxx.xxx.xxx</div><div>a=rtpmap:18 G729/8000</div><div>a=fmtp:18 annexb=no</div><div>a=ptime:20</div>
<div><br></div><div><-------------></div><div>--- (15 headers 10 lines) ---</div><div>Found RTP audio format 18</div><div>Found audio description format G729 for ID 18</div><div>Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 </div>
<div><br></div><div>(nothing)/text=0x0 (nothing), combined - 0x100 (g729)</div><div>Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 </div><div><br></div><div>(nothing)</div>
<div>Peer audio RTP is at port xxx.xxx.xxx.xxx:18132</div><div><br></div><div><--- SIP read from UDP:xxx.xxx.xxx.xxx:5060 ---></div><div>SIP/2.0 200 OK</div><div>Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK60c02567;rport</div>
<div>From: "6598715968" <sip:6598715968@yyy.yy.yy.yy>;tag=as6e218907</div><div>To: <sip:34546598715968@xxx.xxx.xxx.xxx>;tag=B6534850-EC6</div><div>Date: Fri, 06 Jan 2012 04:51:39 GMT</div><div>Call-ID: 7e54da423b0e6e457475ab17694e5165@yyy.yy.yy.yy</div>
<div>Server: Cisco-SIPGateway/IOS-12.x</div><div>CSeq: 102 INVITE</div><div>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</div><div>Allow-Events: telephone-event</div><div>
Contact: <sip:34546598715968@xxx.xxx.xxx.xxx:5060></div><div>Supported: replaces</div><div>Content-Type: application/sdp</div><div>Content-Disposition: session;handling=required</div><div>Content-Length: 223</div><div>
<br></div><div>v=0</div><div>o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx</div><div>s=SIP Call</div><div>c=IN IP4 xxx.xxx.xxx.xxx</div><div>t=0 0</div><div>m=audio 18132 RTP/AVP 18</div><div>c=IN IP4 xxx.xxx.xxx.xxx</div>
<div>a=rtpmap:18 G729/8000</div><div>a=fmtp:18 annexb=no</div><div>a=ptime:20</div><div><br></div><div><-------------></div><div>--- (15 headers 10 lines) ---</div><div>list_route: hop: <sip:34546598715968@xxx.xxx.xxx.xxx:5060></div>
<div>set_destination: Parsing <sip:34546598715968@xxx.xxx.xxx.xxx:5060> for address/port to send to</div><div>set_destination: set destination to xxx.xxx.xxx.xxx, port 5060</div><div>Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:</div>
<div>ACK sip:34546598715968@xxx.xxx.xxx.xxx:5060 SIP/2.0</div><div>Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK17854b94;rport</div><div>Max-Forwards: 70</div><div>From: "6598715968" <sip:6598715968@yyy.yy.yy.yy>;tag=as6e218907</div>
<div>To: <sip:34546598715968@xxx.xxx.xxx.xxx>;tag=B6534850-EC6</div><div>Contact: <sip:6598715968@yyy.yy.yy.yy></div><div>Call-ID: 7e54da423b0e6e457475ab17694e5165@yyy.yy.yy.yy</div><div>CSeq: 102 ACK</div><div>
User-Agent: Asterisk PBX 1.6.2.14</div><div>Content-Length: 0</div><div><br></div><div><br></div><div>---</div><div> > Channel SIP/xxx.xxx.xxx.xxx-00003693 was answered.</div><div> -- Executing [6591394459@a2billing-callback:1] DeadAGI("SIP/xxx.xxx.xxx.xxx-00003693", </div>
<div><br></div><div>"a2billing.php,1,callback") in new stack</div><div> -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php</div><div> -- AGI Script Executing Application: (DIAL) Options: </div>
<div><br></div><div>(SIP/xxx.xxx.xxx.xxx/34546591394459,60,HRrL(370239000:61000:30000))</div><div> -- Limit Data for this call:</div><div> > timelimit = 370239000</div><div> > play_warning = 61000</div>
<div> > play_to_caller = yes</div><div> > play_to_callee = no</div><div> > warning_freq = 30000</div><div> > start_sound =</div><div> > warning_sound = timeleft</div><div>
> end_sound =</div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5</div><div>Audio is at yyy.yy.yy.yy port 14212</div><div>Adding codec 0x100 (g729) to SDP</div><div>Adding codec 0x4 (ulaw) to SDP</div>
<div>Adding codec 0x8 (alaw) to SDP</div><div>Adding codec 0x2 (gsm) to SDP</div><div>Adding non-codec 0x1 (telephone-event) to SDP</div><div>Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060:</div><div>INVITE sip:34546591394459@xxx.xxx.xxx.xxx SIP/2.0</div>
<div>Via: SIP/2.0/UDP yyy.yy.yy.yy:5060;branch=z9hG4bK4ea95f20;rport</div><div>Max-Forwards: 70</div><div>From: "6598715968" <sip:6598715968@yyy.yy.yy.yy>;tag=as492477b7</div><div>To: <sip:34546591394459@xxx.xxx.xxx.xxx></div>
<div>Contact: <sip:6598715968@yyy.yy.yy.yy></div><div>Call-ID: 4d866149766030b331fee79f62bc2030@yyy.yy.yy.yy</div><div>CSeq: 102 INVITE</div><div>User-Agent: Asterisk PBX 1.6.2.14</div><div>Date: Fri, 06 Jan 2012 06:23:10 GMT</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</div><div>Supported: replaces, timer</div><div>Content-Type: application/sdp</div><div>Content-Length: 331</div><div><br></div><div>v=0</div><div>
o=root 1686167830 1686167830 IN IP4 yyy.yy.yy.yy</div><div>s=Asterisk PBX 1.6.2.14</div><div>c=IN IP4 yyy.yy.yy.yy</div><div>t=0 0</div><div>m=audio 14212 RTP/AVP 18 0 8 3 101</div><div>a=rtpmap:18 G729/8000</div><div>a=fmtp:18 annexb=no</div>
<div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:3 GSM/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=sendrecv</div><div><br></div><div>
<br></div><div><br></div><div>To: <sip:34546598715968@xxx.xxx.xxx.xxx>;tag=B6534850-EC6</div><div>Date: Fri, 06 Jan 2012 04:51:39 GMT</div><div>Call-ID: 7e54da423b0e6e457475ab17694e5165@yyy.yy.yy.yy</div><div>Server: Cisco-SIPGateway/IOS-12.x</div>
<div>CSeq: 102 INVITE</div><div>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</div><div>Allow-Events: telephone-event</div><div>Remote-Party-ID: "6598715968" </div>
<div><br></div><div><sip:1234#6598715968@xxx.xxx.xxx.xxx>;party=called;screen=no;privacy=off</div><div>Contact: <sip:34546598715968@xxx.xxx.xxx.xxx:5060></div><div>Content-Type: application/sdp</div><div>Content-Disposition: session;handling=required</div>
<div>Content-Length: 223</div><div><br></div><div>v=0</div><div>o=CiscoSystemsSIP-GW-UserAgent 6911 3862 IN IP4 xxx.xxx.xxx.xxx</div><div>s=SIP Call</div><div>c=IN IP4 xxx.xxx.xxx.xxx</div><div>t=0 0</div><div>m=audio 18132 RTP/AVP 18</div>
<div>c=IN IP4 xxx.xxx.xxx.xxx</div><div>a=rtpmap:18 G729/8000</div><div>a=fmtp:18 annexb=no</div><div>a=ptime:20</div></div><div class="HOEnZb"><div class="h5"><div><br></div><div><br></div><br><div class="gmail_quote">On Mon, Jan 9, 2012 at 4:33 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">You are hereby encouraged to post your AS5300 IOS config, sip.conf peer declaration, and packet capture. Those three things would aid greatly in diagnosis, especially the capture.<br>
<br>
--<br>
This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness.<br>
<br>
Alex Balashov - Principal<br>
Evariste Systems LLC<br>
260 Peachtree Street NW<br>
Suite 2200<br>
Atlanta, GA 30303<br>
Tel: <a href="tel:%2B1-678-954-0670" value="+16789540670" target="_blank">+1-678-954-0670</a><br>
Fax: <a href="tel:%2B1-404-961-1892" value="+14049611892" target="_blank">+1-404-961-1892</a><br>
Web: <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a><br>
<div><div><br>
On Jan 9, 2012, at 3:20 AM, Roi Stork <<a href="mailto:roi.stork@gmail.com" target="_blank">roi.stork@gmail.com</a>> wrote:<br>
<br>
> Hi,<br>
><br>
> We have a problem connecting to a Cisco AS5300 trunk.<br>
><br>
> We set the sip peer to allow only g729. The call attempt is able to connect, but when answered, no audio is heard or transmitted.<br>
><br>
> Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium.<br>
><br>
> We do not have this problem on our other providers using asterisk and other non-cisco systems.<br>
> Anyone else having this same problem?<br>
</div></div><span><font color="#888888">> --<br>
> _____________________________________________________________________<br>
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</font></span></blockquote></div><br>
</div></div></blockquote></div><br></div>