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Following is my debug and look like its not sending MWI NOTIFY message to phone<br><br>Reliably Transmitting (no NAT) to 172.30.245.143:5060:<br>OPTIONS sip:7623@172.30.245.143 SIP/2.0<br>Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK5bd640a3<br>Max-Forwards: 70<br>From: "asterisk" <sip:asterisk@172.30.1.46>;tag=as26352734<br>To: <sip:7623@172.30.245.143><br>Contact: <sip:asterisk@172.30.1.46:5060><br>Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060<br>CSeq: 102 OPTIONS<br>User-Agent: Asterisk PBX SVN-branch-1.8-r321926<br>Date: Wed, 08 Jun 2011 14:49:03 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Length: 0<br><br><br>---<br><br><--- SIP read from UDP:172.30.245.143:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK5bd640a3<br>From: "asterisk" <sip:asterisk@172.30.1.46>;tag=as26352734<br>To: <sip:7623@172.30.245.143>;tag=E777D3B9-F605D562<br>CSeq: 102 OPTIONS<br>Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060<br>Contact: <sip:7623@172.30.245.143><br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER<br>User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043<br>Content-Length: 0<br><br><-------------><br>--- (10 headers 0 lines) ---<br>Really destroying SIP dialog '44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060' Method: OPTIONS<br>Reliably Transmitting (no NAT) to 172.30.245.143:5060:<br>OPTIONS sip:7623@172.30.245.143 SIP/2.0<br>Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK18a12f37<br>Max-Forwards: 70<br>From: "asterisk" <sip:asterisk@172.30.1.46>;tag=as0c8778f4<br>To: <sip:7623@172.30.245.143><br>Contact: <sip:asterisk@172.30.1.46:5060><br>Call-ID: 50d5cc4d5510ae014c6641702faea18d@172.30.1.46:5060<br>CSeq: 102 OPTIONS<br>User-Agent: Asterisk PBX SVN-branch-1.8-r321926<br>Date: Wed, 08 Jun 2011 14:50:03 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Length: 0<br><br><br>---<br><br><--- SIP read from UDP:172.30.245.143:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK18a12f37<br>From: "asterisk" <sip:asterisk@172.30.1.46>;tag=as0c8778f4<br>To: <sip:7623@172.30.245.143>;tag=47557FCE-869CEA2F<br>CSeq: 102 OPTIONS<br>Call-ID: 50d5cc4d5510ae014c6641702faea18d@172.30.1.46:5060<br>Contact: <sip:7623@172.30.245.143><br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER<br>User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043<br>Content-Length: 0<br><br><-------------><br>--- (10 headers 0 lines) ---<br><br><br><hr id="stopSpelling">From: satish_lx@hotmail.com<br>To: asterisk-users@lists.digium.com<br>Date: Wed, 8 Jun 2011 14:43:57 +0000<br>Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI<br><br>
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Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 <br><br>May be i am missing some configuration option. Do you know any debug method to make it work ?<br><br>> From: EWieling@nyigc.com<br>> To: asterisk-users@lists.digium.com<br>> Date: Wed, 8 Jun 2011 10:34:16 -0400<br>> Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI<br>> <br>> All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball.<br>> <br>> > -----Original Message-----<br>> > From: asterisk-users-bounces@lists.digium.com<br>> > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of<br>> > satish patel<br>> > Sent: Wednesday, June 08, 2011 9:57 AM<br>> > To: asterisk-users<br>> > Subject: [asterisk-users] Asterisk 1.8 broken MWI<br>> ><br>> > Hi ALL,<br>> ><br>> > After upgrade 1.8 my MWI wasn't working I do have setting in<br>> > voicemail.conf. Do i need to do anything else to fix my MWI<br>> > on polycom 501 ? It was working with 1.2 asterisk.<br>> ><br>> > pollmailboxes=yes<br>> ><br>> ><br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>
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