Hi,Rafael<div><br></div><div>I found some difference in INVITE message when the switch is long name, see inline mark in red.</div><div>I guess the root cause is the server <a href="http://msssasu1.mydomain.com.py/" target="_blank">MSSSASU1.MYDOMAiN.COM.PY</a> doesn't support tcp.</div>
<div><br></div><div>Best Regards</div><div><br></div><div>Ben<br><br><div class="gmail_quote">2012/8/27 Rafael Visser <span dir="ltr"><<a href="mailto:rafael_visser@hotmail.com" target="_blank">rafael_visser@hotmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr">
Hi Gurus..<br>I use asterisk for just for ivr.<br>My issue is that when the switch changes it's host name from MSSASU1.MYDOMAIN to <a href="http://MSSSASU1.MYDOMAiN.COM" target="_blank">MSSSASU1.MYDOMAiN.COM</a> or <a href="http://MSSSASU1.MYDOMAiN.COM.PY" target="_blank">MSSSASU1.MYDOMAiN.COM.PY</a> the call is rejected with "No matching peer" and the "handle_request_invite: Sending fake auth rejection for device x". It doesn't match it's own default context. <br>
<br>Also, it has somethig to do with the numbers of digits of the dialed number. Few digits works ok, 14 to more works wrong.<br>Do you know what am i missing?<br>Thanks in advance.<br><br><br><br><br><br><br><br><br><br>
Debug with long hostname (B is considered as an '*')<br>================================<br><--- SIP read from TCP:<a href="http://10.146.9.70:6240" target="_blank">10.146.9.70:6240</a> ---><br>INVITE <a href="mailto:sip%3AB56510123456789012345@SISIVR03.MYDOMAIN.COM.PY" target="_blank">sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY</a>;user=phone SIP/2.0<br>
From: <<a href="mailto:sip%3A971200152@MSSASU1.MYDOMAIN.COM.PY" target="_blank">sip:971200152@MSSASU1.MYDOMAIN.COM.PY</a>;user=phone>;tag=3016589695<br>To: <<a href="mailto:sip%3AB56510123456789012345@SISIVR03.MYDOMAIN.COM.PY" target="_blank">sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY</a>;user=phone><br>
Max-Forwards: 70<br><font color="#ffffff"><span style="background-color:rgb(255,0,0)">Via: SIP/2.0/TCP</span> </font>MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096 <span style="background-color:rgb(255,0,0)"><font color="#ffffff">[Ben] --here indicate the SIP use TCP as transport protocol, but the Contact header field is UDP protocol, they are mismatch, you can check the same INVITE message when the switch is short name,[ they are matched.</font></span></div>
<div dir="ltr">Call-ID: <a href="mailto:9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY" target="_blank">9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY</a><br>CSeq: 7313 INVITE<br>P-Asserted-Identity: <<a href="mailto:sip%3A971200152@MSSASU1.MYDOMAIN.COM.PY" target="_blank">sip:971200152@MSSASU1.MYDOMAIN.COM.PY</a>;user=phone><br>
Accept: application/sdp<br>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE<br>P-Charging-Vector: icid-value=6510081000-0826-16155907;icid-generated-at=<a href="http://MSSASU1.MYDOMAIN.COM.PY" target="_blank">MSSASU1.MYDOMAIN.COM.PY</a>;orig-ioi=<a href="http://MSSASU1.MYDOMAIN.COM.PY" target="_blank">MSSASU1.MYDOMAIN.COM.PY</a><br>
Supported: 100rel<br>Content-Type: application/sdp<br><font style="background-color:rgb(255,0,0)" color="#ffffff">Contact: <sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP></font><br>Content-Length: 414<br><br>v=0<br>
o=- 7530078 7530078 IN IP4 <a href="http://MSSASU1.MYDOMAIN.COM.PY" target="_blank">MSSASU1.MYDOMAIN.COM.PY</a><br>s=-<br>t=0 0<br>a=sendrecv<br>m=audio 13802 RTP/AVP 8 96 18 97<br>c=IN IP4 10.143.1.67<br>b=RR:0<br>b=RS:0<br>
a=rtpmap:8 PCMA/8000<br>a=rtpmap:96 AMR/8000<br>a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=yes<br>a=rtpmap:97 telephone-event/8000<br>
a=fmtp:97 0-15<br>a=maxptime:40<br><-------------><br>--- (15 headers 17 lines) ---<br>Sending to <a href="http://10.146.9.70:5060" target="_blank">10.146.9.70:5060</a> (no NAT)<br>Using INVITE request as basis request - <a href="mailto:9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY" target="_blank">9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY</a><br>
################<br>No matching peer for '971200152' from '<a href="http://10.146.9.70:6240" target="_blank">10.146.9.70:6240</a>'<br>[Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite: Sending fake auth rej<br>
ection for device <<a href="mailto:sip%3A971200152@MSSASU1.MYDOMAIN.COM.PY" target="_blank">sip:971200152@MSSASU1.MYDOMAIN.COM.PY</a>;user=phone>;tag=3016589695<br>#################<br><--- Reliably Transmitting (no NAT) to <a href="http://10.146.9.70:5060" target="_blank">10.146.9.70:5060</a> ---><br>
SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096;received=10.146.9.70<br>From: <<a href="mailto:sip%3A971200152@MSSASU1.MYDOMAIN.COM.PY" target="_blank">sip:971200152@MSSASU1.MYDOMAIN.COM.PY</a>;user=phone>;tag=3016589695<br>
To: <<a href="mailto:sip%3AB56510123456789012345@SISIVR03.MYDOMAIN.COM.PY" target="_blank">sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY</a>;user=phone>;tag=as4cfd0d54<br>Call-ID: <a href="mailto:9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY" target="_blank">9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY</a><br>
CSeq: 7313 INVITE<br>Server: Asterisk PBX 1.8.7.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35ff0feb"<br>
Content-Length: 0<br><br><br><br><br>Short hostname on switch<br>===============<br>Connected to Asterisk 1.8.7.0 currently running on fdosis-ims1 (pid = 25430)<br>fdosis-ims1*CLI> core set verbose 1<br>Verbosity was 0 and is now 1<br>
<br><--- SIP read from UDP:<a href="http://10.146.9.70:5060" target="_blank">10.146.9.70:5060</a> ---><br>INVITE <a href="mailto:sip%3AB56510123456789012345@SISIVR03.MYDOMAIN.COM.PY" target="_blank">sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY</a>;user=phone SIP/2.0<br>
From: <sip:971200152@MSSASU1.MYDOMAIN;user=phone>;tag=0046120455<br>To: <<a href="mailto:sip%3AB56510123456789012345@SISIVR03.MYDOMAIN.COM.PY" target="_blank">sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY</a>;user=phone><br>
Max-Forwards: 70<br>Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982<br>Call-ID: qDaQ1240646182201-AAAAAKDE-@MSSASU1.MYDOMAIN<br>CSeq: 14481 INVITE<br>P-Asserted-Identity: <sip:971200152@MSSASU1.MYDOMAIN;user=phone><br>
Accept: application/sdp<br>llow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE<br>P-Charging-Vector: icid-value=A6D6B81000-0826-16440101;icid-generated-at=MSSASU1.MYDOMAIN;orig-ioi=<a href="http://MSSASU1.MYDOMAIN.COM.PY" target="_blank">MSSASU1.MYDOMAIN.COM.PY</a><br>
Supported: 100rel<br>Content-Type: application/sdp<br>Contact: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP><br>Content-Length: 407<br><br>v=0<br>o=- 8986991 8986991 IN IP4 MSSASU1.MYDOMAIN<br>s=-<br>t=0 0<br>a=sendrecv<br>
m=audio 30838 RTP/AVP 8 96 18 97<br>c=IN IP4 10.143.1.68<br>b=RR:0<br>b=RS:0<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:96 AMR/8000<br>a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0<br>
a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=yes<br>a=rtpmap:97 telephone-event/8000<br>a=fmtp:97 0-15<br>a=maxptime:40<br><-------------><br>--- (15 headers 17 lines) ---<br>Sending to <a href="http://10.146.9.70:5060" target="_blank">10.146.9.70:5060</a> (no NAT)<br>
Using INVITE request as basis request - qDaQ1240646182201-AAAAAKDE-@MSSASU1.MYDOMAIN<br>Found peer 'sip.ericsson' for '971200152' from <a href="http://10.146.9.70:5060" target="_blank">10.146.9.70:5060</a><br>
Found RTP audio format 8<br>Found RTP audio format 96<br>Found RTP audio format 18<br>Found RTP audio format 97<br>Found audio description format PCMA for ID 8<br>Found unknown media description format AMR for ID 96<br>Found audio description format G729 for ID 18<br>
Found audio description format telephone-event for ID 97<br>Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)<br>Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)<br>
Peer audio RTP is at port <a href="http://10.143.1.68:30838" target="_blank">10.143.1.68:30838</a><br>Looking for B56510123456789012345 in incoming-sip-ericsson (domain <a href="http://SISIVR03.MYDOMAIN.COM.PY" target="_blank">SISIVR03.MYDOMAIN.COM.PY</a>)<br>
list_route: hop: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP><br><br><--- Transmitting (no NAT) to <a href="http://10.146.9.70:5060" target="_blank">10.146.9.70:5060</a> ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982;received=10.146.9.70<br>
From: <sip:971200152@MSSASU1.MYDOMAIN;user=phone>;tag=0046120455<br>To: <<a href="mailto:sip%3AB56510123456789012345@SISIVR03.MYDOMAIN.COM.PY" target="_blank">sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY</a>;user=phone><br>
Call-ID: qDaQ1240646182201-AAAAAKDE-@MSSASU1.MYDOMAIN<br>CSeq: 14481 INVITE<br>Server: Asterisk PBX 1.8.7.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>
Contact: <<a href="http://sip:B56510123456789012345@10.146.9.132:5060" target="_blank">sip:B56510123456789012345@10.146.9.132:5060</a>><br>Content-Length: 0<br><br><br> </div></div>
<br>--<br>
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